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Orgasmatron 5.2: The Secure Swiss Army Knife for Asterisk

Monday, March 15, 2010 / 18 Comments on Orgasmatron 5.2: The Secure Swiss Army Knife for Asterisk

It’s been an exciting couple of weeks watching the overwhelmingly positive response to our release of Orgasmatron 5.1. With this version, we introduced a new Asterisk® security model that took into account the ever-increasing security risks posed by exposing web and telephony servers to direct Internet access. The bottom line is this. If your telecom requirements still can be accomplished by placing a server securely behind a $35 hardware-based Internet firewall with no Internet exposure, then it makes absolutely no sense to dangle such a tempting target in front of the world’s most nefarious creeps.

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

Our experience suggests that the only trade off with this new approach is the inability to receive anonymous SIP calls… a small price to pay considering the potential financial and computer risks involved. You still can place outbound VoIP calls as well as placing and receiving calls using any of the phone numbers registered on your new PBX in a Flash server. And, thanks to Google Voice, SIPgate, and IPkall, all inbound calls are free, and all outbound calls to numbers in the U.S. and Canada are free as well.

If a SIP URI and your own Freenum/ISN number are simply features you can’t live without, sign up for a voip.ms IAX account, and you’ll get a SIP URI for free. Inbound SIP URI and Freenum/ISN calls will set you back $1 for every 1,000 minutes billed in 6 second increments.

Or you can sign up for a free IP Freedom CallCentric account and configure a new SIP trunk in FreePBX by following these directions. Once configured, your new server SIP URI will be 1777xxxxxxx@in.callcentric.com where xxxxxxx is your assigned 7-digit CallCentric number.

Keep in mind that a new security vulnerability has been found with either Asterisk or FreePBX almost monthly. The chart below tells you why. With virtually limitless attack surfaces because of the number of interrelated components in CentOS, Asterisk, and FreePBX comes enormous and recurring potential for remote compromise of these systems. Rather than play this cat-and-mouse security game with the underworld, the Orgasmatron design changes the paradigm. It lets you use any (secure or insecure) version of Asterisk and FreePBX without worrying about any outside attacks. Do passwords on your new server matter? Not really… unless there is someone inside your firewall that you don’t trust. 🙄 Are we going to secure them anyway? Absolutely. But instead of the constant worry over new security vulnerabilities, Orgasmatron 5.2 lets you enjoy exploring the world of Asterisk and VoIP telephony with an incredibly rich feature set that you won’t find anywhere else, period! We’ll resist making any other device analogies, but the idea here is to protect the good guy (you!) while keeping the bad guys out. No penetration. No worries. Simple as that.

In our former life working for a living, we actually procured and managed multimillion dollar PBXs as part of our "other duties as assigned." Without qualification, we can tell you that the feature set that Orgasmatron 5.2 brings to the table for free runs circles around anything you could buy (then or now) in the commercial marketplace. And, at one time or another, we purchased every Nortel feature good money could buy. There’s one other difference. Orgasmatron 5.2 runs swimmingly on a $200 Atom-based PC that you can purchase at any Best Buy as well as hundreds of other stores including Amazon, NewEgg, and Buy.com. We paid more than $200 to provision an additional extension on our Nortel switch! You, of course, can add as many extensions as you like. De nada.

So, why a new version of Orgasmatron in only a few weeks? Well, it’s not security-related. In fact, there is nothing wrong with continuing on with Orgasmatron 5.1. Unfortunately, it relied exclusively upon SIPgate to make free Google Voice calls in the U.S. and Canada. And SIPgate required an invite using an SMS message from a U.S.-based cellphone. That pretty well knocked out all of our friends living outside the United States. Today’s version fixes that by letting anyone sign up for a free IPkall phone number in Washington state. All you need is a valid email address. The setup process is a bit more complex because IPkall doesn’t support registered connections to their servers. But we’ll walk you through the additional steps and, once completed, your server will be just as secure as the SIPgate approach we set up with Orgasmatron 5.1. And few, if any, Linux skills are required to set up or manage Orgasmatron 5.2. As we’ve noted previously, if you can handle slice and bake cookies, you’ve got the necessary skillset! Be aware this is about a one-hour project, and you need to track through the article carefully, or the entire house of cards comes down.

New Asterisk Security Model. Orgasmatron 5.2 maintains our design goal of running an absolutely secure Asterisk PBX from behind a hardware-based firewall with either NO INBOUND PORTS exposed to the Internet with SIPgate or an IP-address-restricted IAX port for IPkall. Don’t defeat this security mechanism by exposing additional ports on your PBX in a Flash server to Internet access. And choose your NAT-based firewall/router carefully. All of these devices are not created equally. Not only do some perform better than others, but certain models are notoriously bad at handling NAT-based routing tasks, a critical requirement in the Asterisk VoIP environment. In almost every case of problems with one-way audio, the real culprit can be traced back to a crappy router. For $35, you really can’t go wrong with the dLink WBR-2310. If you want traffic shaping functionality as well, take a look at dLink’s Gaming Router, our personal favorite.

As long as your router, Google Voice, SIPgate, and IPkall passwords are secure, you can sleep like a baby. We use an intermediate SIP provider for Google Voice to set up free outbound Google Voice calls in the U.S. and Canada because Google Voice actually places two calls to connect you to your destination. First, you get a call back. And then the party you’re calling is connected. The SIPgate or IPkall trunk is used by Google Voice to call you back so the inbound call is always free. We handle the interconnection magic with Asterisk transparently so your calls appear to be processed as if you were using a standard telephone to dial out. Just refrain from using extension 75 in Asterisk for personal conferencing!

The choice is yours. You can use SIPgate with no incoming ports exposed to your server from the Internet. Or you can use IPkall and map UDP port 4569 (IAX2) on your hardware-based firewall to the internal IP address of your new PBX in a Flash server. Even with the IPkall setup, we’ve locked down IPtables (our Linux firewall) to restrict IAX access to several specific IP addresses so your server remains absolutely secure. We’ve also included support for FonicaTec’s IAX offering for those that want a backup IAX provider. We’ll have much more to say about IPtables in coming weeks.

If you’ve already installed Orgasmatron 5.1 and it’s working for you, do you need to upgrade? NO. With the exception of the new IAX support for IPkall, the code in Orgasmatron 5.2 is identical.

We, of course, continue to recommend that you sign up with Vitelity so you have an alternate communications vehicle in the event of a problem with your free service. Vitelity also can provide 911 emergency service for your home or home office. You can save a little money while supporting the PBX in a Flash project by using the links at the end of this article.

Swiss Army Knife Inventory. There’s no need for a Swiss Army Knife if you don’t know what all the blades are for. So, for those that are wondering what’s included in the Orgasmatron 5.2 build, here’s a feature list of the components you get in addition to the base PBX in a Flash build with CentOS 5.4, Asterisk 1.4, FreePBX 2.6, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Please note that A2Billing, Cepstral TTS, Hamachi VPN, and Mondo Backups are optional and may be installed using the scripts that are provided.

Prerequisites. Here’s what you’ll need to get started:

  • Broadband Internet connection
  • Rock-solid NAT router/firewall. Recommend: $35 dLink WBR-2310
  • $200 PC on which to run PBX in a Flash or a Proxmox Virtual Machine
  • Free Google Voice account (HINT: Under $2 on eBay)
  • Free SIPgateOne residential account (Use cell to get SMS invite) OR
  • Free IPkall IAX account

Learn First. Install Second. Even though the installation process is now a No-Brainer, you are well-advised to do some reading before you begin. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some time learning where the minefields are in today’s VoIP world. Start by reading our Primer on Asterisk Security. Then read our PBX in a Flash and VPN in a Flash knols. If you’re still not asleep, there’s loads of additional documentation on the PBX in a Flash documentation web site.

Today’s Drill. The installation process is straight-forward, but a little different than the Orgasmo 5.1 scenario because of the need to accommodate IPkall. Just don’t skip any steps. In a nutshell, here are the 6 Steps to Free Calling and an incredibly versatile, preconfigured Asterisk PBX:

1. Install the latest version of PBX in a Flash
2. Run the Orgasmatron 5.2 Installer
3. Configure a softphone or SIP telephone
4. Configure Providers for Orgasmatron 5.2
5. Enter your Google Voice and SIPgate/IPkall credentials
6. Change existing passwords to secure your system

Installing PBX in a Flash. Here’s a quick tutorial to get PBX in a Flash installed. We recommend you install the latest PIAF 1.6 beta on a new Atom-based PC. This beta is virtually identical to version 1.4 except it uses CentOS 5.4 instead of CentOS 5.2. This means it works better with newer hardware including Atom-based computers and newer network cards. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS operating system. Once installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities. We use the identical payload for versions 1.3, 1.4, 1.5, and 1.6 of PBX in a Flash. The beta label simply means we haven’t had time to sufficiently test CentOS. But this is not a Microsoft-style beta so fear not!

Download the 32-bit, PIAF 1.6 version from SourceForge, Vitelity, Cybernetic Networks, or AdHoc Electronics. The MD5 checksum for the file is e8a3fc96702d8aa9ecbd2a8afb934d36. Burn the ISO to a CD. Then boot from the installation CD and type ksalt to begin.

WARNING: This install will completely erase, repartition, and reformat ALL disks on your system! 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’. 🙂

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 A option. Have a 10-minute cup of coffee. After installation is complete, the machine will reboot a second time. Log in as root with your new password and execute the following commands:

update-scripts
update-fixes

When prompted, change the ARI password to something really obscure. You’re never going to use it! 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.

NOTE: So long as your system is safely sitting behind a hardware-based firewall, we do NOT recommend running update-source on the Orgasmatron builds because of parking lot issues in the latest releases of Asterisk.

Running the Orgasmatron 5.2 Installer. Log into your server as root and issue the following commands to run the Orgasmatron 5.2 installer:

cd /root
wget http://pbxinaflash.net/orgasmo52.x
chmod +x orgasmo52.x
./orgasmo52.x

Have another 15-minute cup of coffee. It’s a great time to consider a modest donation to the Nerd Vittles project. You’ll find a link at the top of the page. When the installer finishes, READ THE SCREEN!

Now run passwd-master1. Set your FreePBX passwords to something very secure but different from your Linux root password.

Next, type status2 and press Enter. Write down the IP address of your new server.

If you’re using IPkall, now’s the time to log in to your hardware-based firewall/router and map UDP port 45693 to the private IP address that you just wrote down. This tells your firewall to pass all IAX2 traffic from the Internet directly to your new server. Don’t worry. We have severely restricted which IP addresses can actually send IAX data through the PBX in a Flash IPtables firewall which is an integral part of this build. And, remember, no hardware firewall adjustments are necessary if you’re using SIPgate instead of IPkall.

For good measure, we recommend you reboot your server at this point. The command to type is simple: reboot4

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, and you’ll find lots of recommendations on Nerd Vittles. 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 82812661 as the password for extension 701 and the actual IP address of your PBX in a Flash server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

Don’t Forget! After you change your extension passwords later in this tutorial, you will need to update the password entry in X-Lite, or you will no longer be able to place calls! In fact, you will get locked out of your server for 90 minutes after three failed password attempts. So put this on a sticky note so you don’t forget, or you’ll regret it in about 15 minutes.

Either a free SIPgate One residential phone number or an IPkall number is a key component in today’s project. And there’s really no reason you can’t use both if they’re available in your location. Do NOT use special characters in your provider passwords, or nothing will work! Continue reading whichever section below applies to you.

Configuring SIPgate. If you live in the U.S. and have a cellphone, we’d recommend the SIPgate option since no adjustment of your hardware-based firewall is required. Otherwise, skip to the IPkall setup below. Step #1 is to request a SIPgate invite at this link. You’ll need to enter your U.S. cellphone number to receive the SMS message with your invitation code. Don’t worry. You can erase your cellphone number from your account once it is set up. Once you receive the invite code, enter it and choose the option to set up a residential account. Next, choose a phone number and write it down. The area code really doesn’t matter because Google Voice is the only one that will be calling this number after we get things set up. For now, leave your cellphone number in place so that you can receive your confirmation call from Google Voice in the next step. After that, you’ll want to revisit SIPgate and remove all parallel calling numbers. Finally, click on the Settings link and write down your SIP ID and SIP Password. You’ll need these in a few minutes to configure PBX in a Flash. Now place a call to your new SIPgate number and make certain that your cellphone rings before proceeding.

Configuring IPkall. If you’ve opted to use IPkall, here’s the drill. First, you’ll need to register for a free IPkall number. This is actually a two-step process. Set it up as a SIP connection when you first register. Then we’ll change it to IAX once your new phone number is provided. So your initial IPkall request should look like this:

We recommend area code 425 for your requested number because IPkall appears to have lots of them. If they don’t have an available number, your request apparently goes in the bit bucket. You’ll know because IPkall typically turns these requests around in a few minutes. Don’t worry about the mothership entry. We’ll change it shortly. The other issue here is your public IP address. If you have a dedicated IP address, no worries. Just plug in the IP address for SIP Proxy. If it’s dynamic, then you’ll need to set up a fully-qualified domain name (FQDN) with a provider such as dyndns.com. Once you’ve got it set up, enter your credentials in the Dynamic DNS tab of your hardware-based firewall to assure that your dynamic IP address is always synchronized with your FQDN. Then enter the FQDN for your SIP Proxy address in the IPkall form. Be sure to make up a VERY secure password. Now send it off and wait for the return email with your new phone number.

When you receive your new phone number, you’ll need to revisit the IPkall site and log in with your phone number and the password you chose above. Make the changes shown below using your actual IPkall phone number instead of 4259876543:

It’s worth stressing that these settings are extremely important so check your work carefully. Be sure the IAX option is selected. Be sure there are no typos in your two phone number entries. And be sure your FQDN or public IP address is correct. Then save your new settings.

We’re going to be making some entries in FreePBX which is the web-GUI that manages PBX in a Flash. For now, we simply need to enter your new IPkall phone number so that incoming calls to your IPkall number will actually ring on your softphone. Later, we’ll make some further adjustments once we get Google Voice humming along.

Using a web browser from your desktop, log in to FreePBX 2.6 at the following link substituting your server’s private IP address for ipaddress: http://ipaddress/admin. You’ll be prompted for a user name (maint) and password (the one you just created with passwd-master).

When FreePBX loads, choose Setup, Trunks, ipkall (iax). In the USER Context field, enter your 10-digit IPkall phone number. Click Submit Changes, Apply Configuration Changes, Continue with Reload to save your settings.

TIP: Be aware that IPkall cancels an assigned phone number after 30 consecutive days of inactivity. If you will be using your number infrequently, it’s a good idea to schedule a Weekly Reminder to call the number with a prerecorded message. This will assure that your number stays functional.

Now let’s test your new phone number. Call your IPkall number from a cellphone or some other phone. Your softphone should ring. Answer the call, and be sure you have voice in both directions! Do not proceed without success here, or the rest of the adventure is a waste of your time.

Configuring Google Voice. Google Voice still is by invitation only so the first thing you’ll need is an invite. If you’re in a hurry, then stroll over to eBay where you’ll find lots of them for under $2. Once you have your invite in hand, click on the email link to set up your account. After you’ve chosen a telephone number, plug in your new SIPgate or IPkall number as the destination for your Google Voice calls and choose Office as the Phone Type. Trust us.

Google then will place a call to your number and ask you to enter a confirmation code that’s been provided. When your cellphone (SIPgate) or softphone (IPkall) rings, answer it and punch in the number. Wait for confirmation. Then hang up.

As we mentioned earlier, there’s no reason you can’t set up both SIPgate and IPkall forwarding numbers in Google Voice. Just repeat the drill with the other provider’s number if you wish to activate both numbers for use with Google Voice. They’re not both going to ring simultaneously as you will see in a minute.

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.

Finally, place a test call to your new Google Voice number and be sure your cellphone or softphone rings. Don’t move forward until you’ve been able to successfully place a call to your phone by dialing your Google Voice number. Once this is working, revisit SIPgate and remove all parallel calling numbers including your cell number.

Adding Your Credentials to PBX in a Flash. We’re ready to insert your Google Voice credentials and SIPgate/IPkall number into PBX in a Flash. You’ll need four pieces of information: your 10-digit Google Voice phone number, your Google Voice account name (which is the email address you used to set up your GV account), your GV password (no spaces!), and your 11-digit SIPgate or IPkall RingBack DID (beginning with a 1). Don’t get the 10-digit GV number mixed up with the 11-digit SIPgate/IPkall RingBack DID, or nothing will work. 🙂

Log back into your server as root and issue the following command: ./configure-gv. Check your entries carefully. If you make a typo in entering any of your data, press Ctrl-C to cancel the script and then run it again!!

Configuring FreePBX. Now shift back to your Desktop and, using a web browser, log in to FreePBX 2.6 at the following link substituting your actual IP address for ipaddress: http://ipaddress/admin. You’ll be prompted for a user name (maint) and password (the one you just created with passwd-master). Depending upon which intermediate provider you’re using, do the following:

SIPgate Setup. When FreePBX loads, choose Setup, Trunks, sipgate. In Peer Details, replace both instances of sipID with your actual SipGate SIP ID. In Peer Details, replace sipPassword with your actual SipGate SIP Password. In Register String, replace sipID with your SipGate SIP ID, replace sipPassword with your SipGate SIP Password, and replace 3333333333 with your 10-digit SipGate Phone Number. When finished, the Register String should look something like the following:

7004484f0:B8TTW3@sipgate.com/4155201234

Click Submit, Apply Configuration Changes, Continue with Reload to save your changes.

SIPgate and IPkall Setup. While still in FreePBX with your browser, click Setup, Inbound Routes, gv-ringback. In DID Number, replace 3333333333 with your 10-digit SIPGate or IPkall Phone Number. In CallerID Number, replace 7777777777 with your 10-digit Google Voice Number.

Click Submit, Apply Configuration Changes, Continue with Reload to save your changes.

Securing FreePBX. You’re almost done. While still in FreePBX, choose each of the 16 preconfigured extensions on your new server and change the extension AND voicemail passwords. Here’s the drill: Setup, Extensions, 501, Submit. After changing secret and Voicemail Password, repeat with the next extension number instead of 501. Then Apply Config Changes, Continue when you’ve finished with all of them.

Now change the default DISA password: Setup, DISA, DISAmain, PIN, Submit Changes, Apply Config Changes, Continue.

Don’t forget to adjust your X-Lite password to match the password entry you made for extension 701!

Orgasmatron Test Flight. The proof is in the pudding as they say. So let’s try two simple tests. First, from another phone, call your Google Voice number. Your softphone should begin ringing shortly. Answer the call and make sure you can send and receive voice on both phones. Hang up. Now let’s place an outbound call. Using the softphone, dial your cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. If everything is working, congratulations!

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. Save the file and restart Asterisk with the command: amportal restart.

Choosing a VoIP Provider. For this week, we’ll point you to some things to play with on your new server. Then, in the subsequent articles below, we’ll cover in detail how to customize every application that’s been loaded. Nothing beats free when it comes to long distance calls. But nothing lasts forever. So we’d 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 VoIP world is new territory for some of you. Unlike the Ma Bell days, there’s really no reason not to have multiple VoIP providers especially for outbound calls. Depending upon where you are calling, calls may be cheaper using different providers for calls to different locations. So we recommend having at least two providers. Visit the PBX in a Flash Forum to get some ideas on choosing alternative providers.

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

  • D-E-M-O – Nerd Vittles Orgasmatron 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 – PBX in a Flash Support Conference Bridge
  • 882*1061 – VoIP Users Conference every Friday at Noon (EST)


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

Homework. Your homework for this week is to do some exploring. FreePBX is a treasure trove of functionality, and the Orgasmatron build adds a bunch of additional options. See if you can find all of them. For starters, you’ll want to activate CallerID Lookups in FreePBX. Choose Setup, CID Superfecta, Default and enter the maint password you created with passwd-master. Then choose Tools, Module Administration, CallerID Lookup, Enable, Process and Save the Settings. Then edit each of the Inbound Routes and choose CallerID Superfecta as the CID Lookup Source. Save your changes. Finally, choose Setup, CallerID Lookup Sources, CallerID Superfecta and be sure your maint password created with passwd-master is correct here, too. If not, update it. For additional tips, visit the forums.

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. s3cmd.faq tells you how to quickly activate the Amazon S3 Cloud Computing service. And, if you’ve heeded our advice and purchased a PogoPlug, you can link to your home-grown cloud. 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 also perfect for off-site backups!

Also check out Tweet2Dial which lets you use Twitter to make Google Voice calls, send free SMS messages, and manage your new Asterisk server. 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. Finally, try out the included Stealth AutoAttendant by dialing your own number and pressing 0 while the greeting is played. This will reroute your call to the demo applications option in the IVR.

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 for Orgasmatron 5.1 and 5.2. 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.

Coming Attractions. In our next episode, we’ll walk you through the process of adding a second, third, fourth, and fifth Google Voice line to your server so that you’ll never run out of free calling on your server. 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.

 

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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. passwd-master is the PIAF utility for setting a master password for FreePBX access with the maint user account. [↩]
  2. status is the PIAF utility program that displays the current status of most major applications running on your server. [↩]
  3. Mapping a port on your firewall to a private IP address unblocks certain Internet packets and allows them to pass through your firewall directly to an IP device "inside" your firewall for further processing. [↩]
  4. reboot is the Linux command for restarting your server. It’s functionally equivalent to shutdown -r now. [↩]

It’s Orgasmatron 5.1: The Ultimate Asterisk Kitchen Sink

Thursday, February 25, 2010 / 28 Comments on It’s Orgasmatron 5.1: The Ultimate Asterisk Kitchen Sink

For those that want a turnkey Asterisk® VoIP PBX with every bell and whistle, today is your very lucky day. This tutorial will walk you through every step. In less than an hour, you'll have your very own, fully functional Asterisk PBX. No Linux skills are required for this setup. There's no charge for any outbound call made to any number in the U.S. or Canada. And inbound calls are free as well.

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

New Asterisk Security Model. Orgasmatron 5.1 has an all-new design which is intended to let you run an absolutely secure Asterisk PBX in your home from behind a secure firewall with NO INBOUND PORTS exposed to the Internet. So long as your router, Google Voice, and SIPgate passwords are secure, you can sleep like a baby. Today's Magic uses SIPgate as an intermediate SIP provider for Google Voice to set up free outbound Google Voice calls in the U.S. and Canada. Remember that Google Voice actually places two calls to connect you to your destination. First, you get a call back. And then the party you're calling is connected. The SIPgate trunk is used by Google Voice to call you back so the inbound SIPgate call is free. We handle all of the interconnection magic with Asterisk transparently so your calls appear to be processed as if you were using a standard telephone to dial out. Just remember not to use extension 75 in Asterisk for your personal conferences!

Because we register your SIP connection with SIPgate permanently, there is no need to open the SIP or IAX Internet ports on your router. In short, your SIP connection with SIPgate works just as if you were using a browser behind a firewall. The return port will automatically be mapped by your NAT-based router. Hence, no security worries! We, of course, do recommend that you sign up with Vitelity so you have an alternate communications vehicle in the event of a problem with your free service. Vitelity also can provide 911 emergency service for your home or home office. You can save a little money while supporting the PBX in a Flash project by using the links at the end of this article.

Kitchen Sink Inventory. No kitchen is complete without an inventory. So, for those that are wondering what's included in the Orgasmatron 5.1 build, here's a feature list of the components you get in addition to the base PBX in a Flash build with Asterisk 1.4, FreePBX 2.6, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. A2Billing, Cepstral, Hamachi VPN, and Mondo Backups are optional and may be installed using the scripts that are provided.

Prerequisites. Here's what you'll need to get started:

  • Broadband Internet connection
  • Rock-solid NAT router/firewall. Recommend: $35 dLink WBR-2310
  • $200 PC on which to run PBX in a Flash or a Proxmox Virtual Machine
  • Free Google Voice account (HINT: Under $2 on eBay)
  • Free SIPgateOne residential account (Use cell to get SMS invite)

Learn First. Install Second. Even though the installation process is now a No-Brainer, you are well-advised to do some reading before you begin. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some precautions to protect your phone bill. Start by reading our Primer on Asterisk Security. Then read our PBX in a Flash and VPN in a Flash knols. If you're still not asleep, there's loads of additional documentation on the PBX in a Flash documentation web site.

Today's Drill. The installation process is straight-forward. Just don't skip any steps. In a nutshell, here are the 6 Steps to Free Calling and an incredibly versatile, preconfigured Asterisk PBX:

1. Configure SIPgate and Google Voice for Orgasmatron 5.1
2. Install the latest version of PBX in a Flash
3. Run the Orgasmatron 5.1 Installer
4. Enter your Google Voice and SIPgate credentials
5. Change existing passwords to secure your system
6. Configure a softphone or SIP telephone

Configuring SIPgate. A free SIPgate One residential phone number is a key component in today's project. This allows you to receive free incoming calls on your SIPgate number. Step #1 is to request an invite at this link. You'll need to enter your U.S. cellphone number to receive the SMS message with your invitation code. Don't worry. You can erase your cellphone number from your account once it is set up. Once you receive the invite code, enter it and choose the option to set up a residential account. Next, choose a phone number and write it down. The area code really doesn't matter because Google Voice is the only one that will be calling this number after we get things set up. For now, leave your cellphone number in place so that you can receive your confirmation call from Google Voice in the next step. After that, you'll want to revisit SIPgate and remove all parallel calling numbers. Finally, click on the Settings link and write down your SIP ID and SIP Password. You'll need these in a few minutes to configure PBX in a Flash. Now place a call to your new SIPgate number and make certain that your cellphone rings before proceeding.

Configuring Google Voice. Google Voice still is by invitation only so the first thing you'll need is an invite. If you're in a hurry, then stroll over to eBay where you'll find lots of them for under $2. Once you have your invite in hand, click on the email link to set up your account. After you've chosen a telephone number, plug in your new SIPgate number as the destination for your Google Voice calls and choose Office as the Phone Type. Trust us.

Google then will place a call to your SIPgate number and ask you to enter a confirmation code that's been provided. When your cellphone rings, answer it and punch in the number. Wait for confirmation. Then hang up.

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.

Now place a test call to your new Google Voice number and be sure your cellphone rings. Don't move forward until you've been able to successfully place a call to your cellphone by dialing your Google Voice number. Once this is working, revisit SIPgate and remove all parallel calling numbers including your cell number.

Installing PBX in a Flash. Now for the fun part. Here's a quick tutorial to get PBX in a Flash installed. We recommend you install the latest PIAF 1.6 beta which is virtually identical to version 1.4 except it uses CentOS 5.4 instead of CentOS 5.2. This means it works better with newer hardware including Atom-based computers and newer network cards. Download the 32-bit, PIAF 1.6 version from here, here, or here. The MD5 checksum for the file is e8a3fc96702d8aa9ecbd2a8afb934d36. Burn the ISO to a CD. Then boot your system from the installation CD and type ksalt to begin.

WARNING: This install will completely erase, repartition, and reformat ALL disks on your system! 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'. 🙂

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 A option. Have a 10-minute cup of coffee. After installation is complete, the machine will reboot a second time. Log in as root with your new password and execute the following commands:

update-scripts
update-fixes

When prompted, change the ARI password to something really obscure. You're never going to use it! 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.

Running the Orgasmatron 5.1 Installer. Log into your server as root and issue the following commands to run the Orgasmatron 5.1 installer:

cd /root
wget http://pbxinaflash.net/orgasmo51.x
chmod +x orgasmo51.x
./orgasmo51.x

Have another 15-minute cup of coffee. It's a great time to consider a modest donation to the Nerd Vittles project. You'll find a link at the top of the page. When the installer finishes, READ THE SCREEN!

Adding Your Credentials to PBX in a Flash. Now we're ready to insert your Google Voice credentials and SIPgate number into PBX in a Flash. You'll need four pieces of information: your 10-digit Google Voice phone number, your Google Voice account name (which is the email address you used to set up your GV account), your GV password (no spaces!), and your 11-digit SIPgate RingBack DID (beginning with a 1). Don't get the 10-digit GV number mixed up with the 11-digit SIPgate RingBack DID, or nothing will work. 🙂

While logged into your server as root, issue the following command: ./configure-gv. Check your entries carefully. If you make a typo in entering any of your data, press Ctrl-C to cancel the script and then run it again!!

Next, run passwd-master and set your FreePBX passwords to something equally secure but different from your Linux root password.

Finally, type status and press Enter. Write down the IP address of your new server. You'll need it in the next step.

Configuring FreePBX. Using a web browser, log in to FreePBX 2.6 at the following link substituting your actual IP address for ipaddress: http://ipaddress/admin. You'll be prompted for a user name (maint) and password (the one you just created with passwd-master).

When FreePBX loads, choose Setup, Trunks, sipgate. In Peer Details, replace both instances of sipID with your actual SipGate SIP ID. In Peer Details, replace sipPassword with your actual SipGate SIP Password. In Register String, replace sipID with your SipGate SIP ID, replace sipPassword with your SipGate SIP Password, and replace 3333333333 with your 10-digit SipGate Phone Number. When finished, the Register String should look something like the following:

7004484f0:B8TTW3@sipgate.com/4155201234

Click Submit Changes, Apply Configuration Changes, Continue with Reload to save your settings.

Now click Setup, Inbound Routes, gv-ringback. In DID Number, replace 3333333333 with your 10-digit SIPGate Phone Number. In CallerID Number, replace 7777777777 with your 10-digit Google Voice Number.

Click Submit, Apply Configuration Changes, Continue with Reload to save your changes.

Securing FreePBX. You're almost done. While still in FreePBX, choose each of the 16 preconfigured extensions on your new server and change the extension AND voicemail passwords. Here's the drill: Setup, Extensions, 501, Submit. After changing secret and Voicemail Password, repeat with the next extension number instead of 501. Then Apply Config Changes, Continue when you've finished with all of them.

Now change the default DISA password: Setup, DISA, DISAmain, PIN, Submit Changes, Apply Config Changes, Continue.

Whew! We recommend you reboot your server at this juncture just to be sure everything gets initialized correctly. Then all we need is a phone and you're all set.

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, and you'll find lots of recommendations on Nerd Vittles. 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 PBX in a Flash server instead of 192.168.0.251. Click OK when finished.

Orgasmatron Test Flight. The proof is in the pudding as they say. So let's try two simple tests. First, from another phone, call your Google Voice number. Your softphone should begin ringing shortly. Answer the call and make sure you can send and receive voice on both phones. Hang up. Now let's place an outbound call. Using the softphone, dial your cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. If everything is working, congratulations!

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. Save the file and restart Asterisk with the command: amportal restart.

Choosing a VoIP Provider. For this week, we'll point you to some things to play with on your new server. Then, in the subsequent articles below, we'll cover in detail how to customize every application that's been loaded. Nothing beats free when it comes to long distance calls. But nothing lasts forever. So we'd 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 VoIP world is new territory for some of you. Unlike the Ma Bell days, there's really no reason not to have multiple VoIP providers especially for outbound calls. Depending upon where you are calling, calls may be cheaper using different providers for calls to different locations. So we recommend having at least two providers. Visit the PBX in a Flash Forum to get some ideas on choosing alternative providers.

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

  • D-E-M-O - Nerd Vittles Orgasmatron 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 - PBX in a Flash Support Conference Bridge
  • 882*1061 - VoIP Users Conference every Friday at Noon (EST)


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

Homework. Your homework for this week is to do some exploring. FreePBX is a treasure trove of functionality, and the Orgasmatron build adds a bunch of additional options. See if you can find all of them. For starters, you'll want to activate CID Superfecta in FreePBX. For tips, start here in the forums. Then log into your server as root and look through the scripts added in the /root/nv folder. You'll find all sorts of goodies to keep you busy. And, be sure to check out Tweet2Dial which lets you use Twitter to make Google Voice calls, send free SMS messages, and manage your new Asterisk server. Finally, don't forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And be sure to 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!

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 for Orgasmatron 5.1. 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.

Upgrading Previous Orgasmatron V Installs. The question we hear over and over is "How do I upgrade from an existing Orgasmatron V install or from an existing Asterisk system?" The short answer is you can't. But there is some good news. For those with existing Orgasmatron V installs, we think we can fix your system so that it makes calls reliably. First, be sure your sipgate and gv-incoming settings match what is shown above in this article. Second, be sure you have configured a sipgate trunk with your proper sipgate credentials. Finally, log into your server as root and issue the following commands:
cd /root
wget http://pygooglevoice.googlecode.com/files/pygooglevoice-0.5.tar.gz
tar zxvf pygooglevoice-0.5*
cd pygooglevoice-0.5
python setup.py install
cd /etc/asterisk
sed -i 's|\${RINGBACK}|\${RINGBACK} 3|' extensions_custom.conf
asterisk -rx "dialplan reload"

Early Adopter WARNING. Current downloads are bug-free as best we can tell. But, for those that installed Orgasmatron 5.1 before 2:20 PM (EST) on Saturday, 2/27/2010, a couple of issues have arisen that need to be addressed. Please visit the following link to Orgasmatron 5.1 patches and apply those applicable to your particular situation. Without these patches, a security vulnerability may exist if you expose your server to web access from the Internet and a number of dialplan errors will cause unexpected behavior. It takes less than a minute to apply all of the patches! I'm reminded of the old Wild West adage: "You can always tell the pioneers by the arrows in their back."

Originally published: February 25, 2010


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

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

 

Special Thanks to Our Generous Sponsors


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

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

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

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

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

Some Recent Nerd Vittles Articles of Interest...

Introducing the Orgasmatron V, Google Voice Edition

Monday, August 17, 2009 / 48 Comments on Introducing the Orgasmatron V, Google Voice Edition

It's been an interesting couple of weeks watching many of our readers flock to Google Voice in order to make free calls in the U.S. and Canada. The only problem with our Google Voice solution was the skill set required to get everything humming along as it should. For those new to the Asterisk® world, it only made sense to create a special installer that would build an Instant PBX.

Check Out the Latest! The Incredible PBX

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

In putting this together, we couldn't help noticing the dilemma posed on the new FreePBX web site: "Looking for Phone Service? We can't quite give you the phone service for the same price (free) as the PBX..." Well, maybe they can't, but we certainly can thanks to our friends at Google Voice. So today we're pleased to introduce the first Truly Freeâ„¢ Asterisk PBX. If you've mastered slice-and-bake cookies, you'll have no trouble with today's recipe.

Welcome to the Orgasmatron V Installer, the wonderscript that lets you create a turnkey Asterisk system with free U.S. and Canada calling through Google Voice in less than 15 minutes! When you're finished you'll have a PBX in a Flash system with every bell and whistle on the planet. Not only is the PBX absolutely free but so are all of your outbound and incoming calls throughout the United States and Canada. All you'll need is an Internet connection and any garden variety PC that's less than 3 years old. Or you can splurge and buy yourself a new Atom-based PC or NetBook and have a state-of-the-art PBX that may last you close to a decade. While you'll still need to change a few passwords and plug in some phones, the Orgasmatron V build reduces the Asterisk learning curve to almost zero. Out of the box, email works. Faxing works. ENUM works. And free calling in the U.S. and Canada works. Just plug in your Google Voice credentials, and you can start placing calls to every phone in the U.S. and Canada for free in just a few minutes.

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

Getting Started. Even though the installation process is now a No-Brainer, you are well-advised to do some reading before you begin. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some precautions to protect your phone bill. Start by reading our Primer on Asterisk Security. Then read our PBX in a Flash and VPN in a Flash knols. If you're still not asleep, there's loads of additional documentation on the PBX in a Flash documentation web site.

Prerequisites. You obviously still need a free Google Voice account. If you don't have one, you can request an invite here. At last report, it's only taking a few days from application to invite which is really great news. Don't use a space in your Google Voice password! Once you have a Google Voice account and phone number (Google has reserved several million of them so... not to worry!), then you'll need a DID that provides unlimited, free incoming calls. We'll use it as your Google Voice RingBack DID and will explain all of this after we get your PBX up and running. We'd recommend a free IPkall or SIPgate DID, but we'll get to that.

Installation. Here's a quick tutorial to get you going. First, install the 32-bit, Asterisk 1.4 version of PBX in a Flash. Boot your system from the installation CD and type ksalt to begin. As your machine reboots, remove the CD and choose option A to load the most stable payload. When the install completes, reboot your system once again and login as root with the password you chose when you built your system. Now issue the following commands to bring your system current and protect your system passwords: update-scripts, update-fixes, passwd-master. You now have a PBX in a Flash base install. On a stand-alone machine, it takes about 30 minutes. On a virtual machine, it takes about half that time.

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

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

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

If you'd prefer to pick and choose the apps to install, use this fully-interactive installer instead:

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

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

http://192.168.0.123/admin/

You're NOT done yet!

These next three steps are important. They get all of the FreePBX modules installed and then restore the FreePBX backup set that's at the heart of the Orgasmatron build. Just follow along here. If you're using the new OpenVZ template for PBX in a Flash, start at step 3 and then complete step 1 and 2. Otherwise...

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

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

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

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

Now let's change the default DISA password: Setup, DISA, DISAmain, PIN, Submit Changes, Apply Config Changes, Continue. Whew! Your system now is relatively secure. Follow the steps in the tutorials we recommended, and you're ready to experiment. Plug in a couple of SIP phones or softphones and configure them using the available extensions (701-715) together with the secrets for those extensions. Place a test call between the extensions to make sure you have a working PBX. Now we're ready to add the pieces so that people from outside your system can call you and so that you can call them as well.

Setting Up An IPkall RingBack DID. Step #1 is obtaining a free DID which will be used to handle RingBack calls from Google Voice. If you're new to Google Voice, here's a quick primer. Whenever you place an outbound call through Google Voice, GV actually places two calls. It returns your call to a number you designate as your RingBack number, and then GV places the call to the destination number you've chosen. We will transparently merge the two calls together behind the scenes so the caller will think it's a "normal" long distance call. But, before Google Voice calling will work with Asterisk, you'll need another DID (in addition to your new Google Voice number) to transparently handle these RingBack calls into Asterisk.

Shown above is the IPkall request form to sign up for a free DID. Make your form look like the one above but change 3 pieces of information: (1) the SIP Proxy which is the public IP address of your Asterisk server or its fully-qualified domain name, (2) a working Email Address which will be used to confirm your request for a free DID, and (3) a password to protect your DID at IPkall. Leave the other entries the way they're shown, especially the SIP Phone Number, gv-ringback, which is preconfigured to route incoming SIP calls on your new PBX to any phones connected to extensions 701-715. Once you have confirmed your request by email, you will be assigned a phone number. Assuming you've already connected a phone to your new PBX on one of the above extensions, it should ring when you call your new IPkall number. Don't proceed until you get this working because it must be functional before you can complete the set up of your Google Voice account.

Setting Up A SIPgate RingBack DID. If you elect to use a SIPgate DID, the process is a bit more complicated. Once you've registered for a free DID on their site, you'll get an email with your credentials. You then will need to create a new trunk using FreePBX with the following entries replacing SIP-ID and SIP-Password with your actual credentials. Use sipgate for the Trunk Name and fill in the following in the Outgoing Settings section of the form:

type=peer
username=SIP-ID
fromuser=SIP-ID
secret=SIP-Password
context=from-trunk
host=sipgate.com
fromdomain=sipgate.com
insecure=very
caninvite=no
canreinvite=no
nat=no
disallow=all
allow=ulaw&alaw

Leave the Incoming Settings blank, and enter the following Registration String using your actual credentials:

SIP-ID:SIP-Password@sipgate.com/10-digitSIPgateNumber

Save your entries and then create an Inbound Route called sipgate. Enter your 10-digit SIPgate number in the DID Number field and choose Ring Group: 700 as the Destination for the inbound calls to this number. Reload your Asterisk dialplan when prompted to do so. Connect a phone to an extension on your PBX and be sure the phone rings when you call your new SIPgate DID number before proceeding.

Google Voice Setup. Once you get your RingBack DID set up on your Asterisk system, we need to configure your new Google Voice account. Log into your GV account and click Settings, Phones, Add Another Phone. Add the area code and phone number of your RingBack DID. Be sure a phone is connected to one of the existing extensions (701-715) on your PBX since you have to go through Google's confirmation drill to successfully register the number with GV. After the DID is confirmed, be sure there's a check mark beside this Google Voice destination so that incoming calls to your GV number will be routed to your Asterisk server.

While you're still in the Google Voice Setup, click on the General tab. Uncheck Enable Call Screening. Turn Call Presentation Off. And set CallerID to Display Caller's Number. Remember NOT to include a space in your Google Voice password! Finally, uncheck Do Not Disturb. Now click the Save Changes button.

Adding Your GV Credentials to PBX in a Flash. Now we're ready to insert your Google Voice credentials into PBX in a Flash. You'll need four pieces of information: your 10-digit Google Voice phone number, your Google Voice account name (which is the email address you used to set up your GV account), your GV password (no spaces!), and your 11-digit RingBack DID (beginning with a 1) from either IPkall or SIPgate. Don't get the 10-digit GV number mixed up with the 11-digit RingBack DID, or nothing will work. 🙂 Now log back into your server as root and issue the following commands. Check your entries carefully. If you make a typo in entering any of your data, press Ctrl-C to cancel the script and then run it again!!

cd /root
wget http://pbxinaflash.net/orgasmatron/configure-gv
chmod +x configure-gv
./configure-gv

Updating pyGoogleVoice. Since this article was initially released, Google has made some changes in the way Google Voice processes incoming calls. To address this, you'll need to update the version of pyGoogleVoice installed with this build. While still logged into your server as root, issue the following commands:
cd /root
wget http://pygooglevoice.googlecode.com/files/pygooglevoice-0.5.tar.gz
tar zxvf pygooglevoice-0.5*
cd pygooglevoice-0.5
python setup.py install

Modifying Your RingBack Inbound Route. The last step in the setup process is to reroute your gv-ringback incoming route so that it points to a custom context to process your Google Voice ringback calls transparently. Log back into FreePBX with a web browser and choose Setup, Inbound Routes, gv-ringback. Change the Destination for these calls to Custom Destinations: Custom GV-Park. If you're using SIPgate instead of IPkall, be sure to change the other settings to look like this:

Description: gv-ringback
DIDNumber: *Your 10-digit-SIPgate-Number*
CallerId: *Your 10-digit-Google-Voice-Number*

Save your changes by clicking the Submit button and then reload your dialplan when prompted.

Choosing a VoIP Provider. For this week, we'll point you to some things to play with on your new server. Then, in the subsequent articles below, we'll cover in detail how to customize every application that's been loaded. Nothing beats free when it comes to long distance calls. But nothing lasts forever. So we'd 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 VoIP world is new territory for some of you. Unlike the Ma Bell days, there's really no reason not to have multiple VoIP providers especially for outbound calls. Depending upon where you are calling, calls may be cheaper using different providers for calls to different locations. So we recommend having at least two providers. Visit the PBX in a Flash Forum to get some ideas on choosing alternative providers.

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

  • D-E-M-O - Check out the Nerd Vittles Orgasmatron Demo
  • Z-I-P - Enter a five digit zip code for any U.S. weather report
  • 6-1-1 - Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 - Get the latest news and sports headlines from Yahoo News
  • T-I-D-E - Get today's tides and lunar schedule for any U.S. port
  • F-A-X - Send a fax to an email address of your choice
  • 4-1-2 - 3-character phonebook lookup/dialer with AsteriDex
  • M-A-I-L - Record a message and deliver it to any email address
  • C-O-N-F - Set up a MeetMe Conference on the fly
  • 1-2-3 - Schedule a regular or recurring phone reminder
  • Dial *68 - Schedule a hotel-style wakeup call on any extension

Google Voice Speed Dials. For frequently called numbers, you can add speed dials by inserting entries in the [from-internal-custom] context of extensions_custom.conf in the /etc/asterisk folder that look like the example below where 333 is the speed dial number and 6781234567 is the area code and number to call. Be sure to reload your Asterisk dialplan to activate them.

exten => 333,1,Dial(local/6781234567@custom-gv,300)

Congratulations! You now have what we hope will be flawless and free U.S. calling on your Asterisk system using Google Voice. No gimmicks, no strings, no cost. Enjoy!

Finally, one additional word of caution. Both Google Voice and this call design are set up for a single call at a time. There are no safeguards to prevent multiple calls, but that may violate the Google Voice terms of service.

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

Continue reading Part II.

Continue reading Part III.

Continue reading Part IV.

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.


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

 

Special Thanks to Our Generous Sponsors


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

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

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

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

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

Some Recent Nerd Vittles Articles of Interest...

Asterisk on Steroids: The Orgasmatron Installer, Part IV

Monday, May 25, 2009

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Read Part I and download the software.

Read Part II.

Read Part III.


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

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

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

 

Special Thanks to Our Generous Sponsors


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

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

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

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

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

Some Recent Nerd Vittles Articles of Interest...

Asterisk on Steroids: The Orgasmatron Installer, Part III

Tuesday, May 5, 2009 / 1 Comment on Asterisk on Steroids: The Orgasmatron Installer, Part III

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

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

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

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

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

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

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

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

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

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

t38pt_rtp=no
t38pt_tcp=no
t38pt_udptl=no

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Continue reading Part IV (Monday, May 25).

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

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

 

Special Thanks to Our Generous Sponsors


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

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

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

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

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

Some Recent Nerd Vittles Articles of Interest...

Asterisk on Steroids: The Orgasmatron Installer, Part II

Monday, April 20, 2009 / 5 Comments on Asterisk on Steroids: The Orgasmatron Installer, Part II

In our last column, we introduced you to the new Orgasmatron Installer for PBX in a Flash. After a one-week break to prepare for our visit to the Atlanta Asterisk® Users Group 3d Annual InstallFest, we're back in the saddle today to flesh out the new baby.

For those that are new to all of this, let's briefly review what the Orgasmatron Installer has added to your Lean, Mean Asterisk Machine. Faxing and email now work out of the box. More than a dozen extensions and a number of hosting provider trunks are preconfigured as well. Delivery of CallerID names with numbers is now available from a half dozen providers of your choice. And, of course, the Flite text-to-speech engine is preconfigured with Cepstral TTS only a few keystrokes away. Also included are FreePBX 2.5, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. And here's the complete list with all of your new Nerd Vittles applications:

Security First! Because your phone bill matters, today we begin with security. The design of virtually all of the open source Asterisk PBX aggregations is to leave SIP and IAX ports on your new server exposed to the Internet. This is done to facilitate communications with your hosting providers as well as telephone extensions which may be connected to your server from the other side of the globe. The wrinkle with this design is that, if a bad guy can guess an extension number on your system and its password, they get a free ticket to do whatever could be done from that extension on your PBX. In the case of one unlucky company, this resulted in a phone bill of over $100,000. For details, read our Primer on Asterisk Security. So... Security Matters!

Anyone obviously can download PBX in a Flash and the Orgasmatron Installer. Thus, you need to assume that everyone on the planet knows your default passwords. We walked you through changing some of the important ones with the passwd-master script last week. Use it regularly. Now let's turn our attention to your extensions and trunk passwords.

Extension Security. There now are a couple of ways to secure your extensions from the bad guys. First, you need to establish very secure passwords for your extensions and voicemail boxes. Second, you need to specify the IP addresses that are authorized to access every extension on your PBX. And third, remember do repeat this drill every time you add a new extension to your system.

To change an extension password, open FreePBX using a web browser pointed to the IP address of your server: http://ipaddress/admin/. On PBX in a Flash systems, you'll be prompted for a username (maint) and whatever password you set when you ran passwd-master. Now click the Setup tab and then the Extensions option. You'll see the list of configured extensions on your PBX in the right column. Click on each of those extensions, and you'll see a form like this:



The password for this extension is stored in a field called secret. Make up a very secure password for every extension on your PBX. You will embed this password in the telephone connected to this extension. There's no other place you'll need it so a long and complex numeric password is essential.

The authorized IP addresses for this extension are stored in a field called permit. The way this works is that you first specify which IP addresses should be denied access (the deny field), and then you poke a little hole in the dike, if you're smart, to permit only one or a few IP addresses to connect to the extension. Leave the deny entry as it is. The default permit entry 0.0.0.0/0.0.0.0 opens the floodgates. It means any IP address can log into this extension. To restrict extension access to IP addresses on a private LAN of 192.168.1, the entry would look like this: 192.168.1.0/255.255.255.0. To further restrict extension access to a specific IP address (recommended!), the entry would look like this: 192.168.1.44/255.255.255.255. Use a permit entry that makes you sleep well at night. After all, it's your phone bill.

The third entry you'll want to change is further down the same data entry form, and that's the Voicemail Password field. This entry determines who can actually retrieve voicemails left for this extension. Set it accordingly.

Once you've made the three changes above, save your entries by clicking the Submit button at the bottom of the form. Repeat the drill for every extension, and then click the orange Apply Configuration Changes tab at the top of the screen and then Continue with Reload to reload your Asterisk dialplan.

Trunk Security. Securing the trunks on your PBX is equally important to securing extensions. Keep in mind that, with your trunk credentials, anyone can set up your trunk on their PBX to make calls on your nickel! Unlike the extensions, there are no working usernames and passwords in the default trunks with one exception. If you plan to use the providers we've preconfigured, simply insert your own username, fromuser, and secret settings in the fields provided, and you'll be making calls in a matter of seconds. The process is similar to the one we used for extensions. Choose Setup, Trunks and then click on each trunk and make your entries. Submit your entries and then reload the dialplan when you're finished.

In the case of the remote-peer trunk, this trunk is designed to make it extremely easy to interconnect Asterisk servers for interoffice communications. But it also means that a bad guy can easily interconnect with your server and start dialing. If you don't plan to connect to another Asterisk server, delete this trunk! If you do plan to connect to another Asterisk server, change the trunk secret and IP address of the host to which you are connecting. Do NOT leave the default secret in either the outgoing or incoming settings! Also change the password for the outbound route: Remote-Host. You may want to ultimately remove this password if you actually start interconnecting servers. Otherwise, users will have to enter this password whenever they may a call to an extension on the interconnected Asterisk server.

To interconnect your server to another server, you would simply add a new trunk called main-peer on the other server that looks like this (using your new password and correct IP address):


Configuring AsteriDex. AsteriDex is plug-and-play for most users. However, as configured, your AsteriDex web site is reachable from the Internet if you have mapped port 80 on your hardware-based firewall to your PBX in a Flash server or if you don't have a hardware-based firewall and your server is directly exposed to the Internet. If you don't mind people seeing your contact list or making prank calls that ring your extensions, this may be okay. If it's of concern to you, the easiest security precaution is to rename the asteridex4 directory to an obscure name that only you know, e.g. bahbah143. Here are the commands to issue after logging into your server as root. By using all of these commands, AsteriDex still will be accessible through FreePBX and the PBX in a Flash GUI:

cd /var/www/html
mv asteridex4 bahbah143
sed -i 's|asteridex4|bahbah143|' admin/modules/asteridex/page.asteridex.php
sed -i 's|asteridex4|bahbah143|' welcome/.htindex.cfg

The other adjustment you may need to make to AsteriDex is to configure who can access the Admin tab to add, modify, and delete entries in your database. As configured, the Admin tab is available to any computer with an IP address that begins with 192.168. This may not match your private subnet, and not all 192.168 IP address are non-routable. So you may wish to tighten this restriction to match your internal subnet. In the /var/www/html/asteridex4 folder (or whatever name you've chosen above), you'll find a configuration file: config.inc.php. Simply edit this file and change the $local_net entry. You also can set the long distance prefix ($LDprefix), your CallerID number ($CallerID), and the default extension to ring for click-to-dial from the web interface ($INtrunk and $defaultExt). The extension to dial can now be set from the web interface as well. Unless you really know what you're doing, leave everything else the way it is.

CallerID Superfecta. Most hosting providers deliver CallerID numbers as part of your payment for using their DIDs. Almost none deliver CallerID names without an additional charge. CallerID Superfecta is designed to fill that gap... for free. A number of us have worked on this project for years. And it now has been integrated directly into FreePBX. There are two steps to getting everything working properly on your new PBX. First, you need to identify which CallerID lookup sources you wish to use on your system. Then, you need to specify CallerID Superfecta as the lookup source on each Inbound Route where you want CallerID names looked up for incoming calls.

Open FreePBX with your web browser and navigate to Setup, CID Superfecta. You'll get a form that looks like this:


With the exception of AsteriDex and SugarCRM lookups which are almost instantaneous, keep in mind that each lookup takes a little time and slows down receipt of your inbound call. So long as you have a good Internet connection, you shouldn't have a problem using all of the sources. The way the CallerID Superfecta works is that, once it gets a name match in any of the sources beginning with AsteriDex and SugarCRM, it ends the lookups and provides the CallerID name it found to Asterisk for display on the extensions which are ringing in the designated inbound route. Filling out the form is self-explanatory for the most part. Tick off the lookup sources you wish to use. If you plan to use whocalled.us, you'll need to sign up for an account and provide your credentials before the lookup will work. With SugarCRM, fill in the blanks to match your implementation of SugarCRM. Click the SAVE button when you have CallerID Superfecta configured to meet your needs.

The final step in implementing CallerID Superfecta is to designate it as the CallerID Lookup Source for your Inbound Routes. Click on Setup, Inbound Routes and a list of your existing routes will be displayed in the right column. As installed, there will only be one: Any DID / Any CID. Click on this entry to display the form. Scroll down to the CallerID Lookup Source dropdown box and choose CallerID Superfecta. You'd do the same with any other inbound route you create down the road. Click the Submit button and reload your dialplan to enable CallerID Superfecta. Now sit back and wait on your first call.

CallWho for Asterisk. CallWho for Asterisk is a little script we put together to make it easy to look up and dial the numbers of people in your AsteriDex database. When you dial 4-1-2, you'll be prompted to enter the first three letters of the name of the person you wish to call. Once you key in the three letters, CallWho for Asterisk will look up every matching entry in your AsteriDex database and read you the list of matches. For example, if you had Joe Schmo and Joe The Plumber in your database, CallWho would say something like this:

Press 1 for Joe Schmo.
Press 2 for Joe The Plumber.

When you press 2, CallWho will place a call to Joe The Plumber. Not sure why you'd ever want to do that, but now you understand the way it works.

Before CallWho for Asterisk will work at all, you need to run the script which associates three letter codes with every entry in your AsteriDex database. And, whenever you add new entries to your database, you need to run it again. Using a web browser, here's the program to run. Be sure to use the correct IP address for your Asterisk server and your newly designated AsteriDex location instead of asteridex4:

http://192.168.0.44/asteridex4/dialcode.php

Cepstral TTS for Asterisk. PBX in a Flash is delivered with the Flite text-to-speech engine already enabled. But, unless you like the voices of Lurch and Fred Munster, you may wish to cough up a little cash and install Cepstral on your server. Cepstral now has a synthesized voice of Allison which exactly matches all of the other voice prompts in Asterisk. I'm embarrassed to report that we can't seem to get the correct installation script deposited in our Orgasmatron builds... ever! So, if you want to use Cepstral, here are the steps to download the real, working installation script and to install Cepstral:

cd /root/nv
rm install-cepstral
wget http://pbxinaflash.net/source/cepstral/install-cepstral
chmod +x install-cepstral
./install-cepstral

Once the 65MB download completes, you'll be prompted to agree to the license. You do this by pressing the Enter key to scroll down the license agreement. When you reach 100%, type yes to continue with the install. Press Enter to accept /opt/swift as the install directory. Very important: Type y to create the directory. The default is No which will mess up the installation. Now type yes to complete the install. Once the install completes, you can purchase a license for the Allison voice at this link. Under Voices, choose Language: US English, Voice: Allison-8kHz, and Platform: Linux. For non-commercial use, the $30 voice registration is all you need. For commercial use, you also need to acquire Concurrency Licenses which authorize a certain number of simultaneous voice ports on your system for Cepstral voices. These run $50 per port in 2-port multiples and are in addition to the $30 Allison voice license. For Nerd Vittles readers, you can save 15% on your purchase by sending an email to sales at cepstral.com explaining how you plan to use Cepstral and requesting the discount code.

We'll have an in-depth article on Cepstral in coming weeks. For those that want a head start, each of the Nerd Vittles text-to-speech applications typically includes dialplan code and one or more PHP/AGI scripts. The dialplan code can be found in /etc/asterisk/extensions_custom.conf. When you scroll through the dialplan code you will see entries like the following for each of the TTS applications:

exten => 611,5,Flite("Enter a 3 character airport code.")
;exten => 611,5,Swift("Enter a 3 character airport code.")
exten => 611,6,Read(APCODE,beep,3)
exten => 611,7,Flite("Please hold a moment.")
;exten => 611,7,Swift("Please hold a moment.")

The semicolon at the beginning of a line tells Asterisk this is a comment and to ignore it. To change the voice from the Munsters to Allison, just comment out the Flite lines and uncomment the Swift lines by deleting the leading semicolons. When you're finished making the changes, save the file and then reload your dialplan: asterisk -rx "dialplan reload". So, in the example above, the code would now look like this:

;exten => 611,5,Flite("Enter a 3 character airport code.")
exten => 611,5,Swift("Enter a 3 character airport code.")
exten => 611,6,Read(APCODE,beep,3)
;exten => 611,7,Flite("Please hold a moment.")
exten => 611,7,Swift("Please hold a moment.")

You also need to modify the PHP/AGI scripts that go with each application. All of these files are stored in /var/lib/asterisk/agi-bin. Typically the filenames begin with nv- and end in .php:
-rwxrwxr-x 1 asterisk asterisk 6835 Sep 16 2008 nv-callwho.php
-rwxrwxr-x 1 asterisk asterisk 201 Jul 12 2006 nv-config-555.php
-rwxrwxr-x 1 asterisk asterisk 201 Apr 2 13:08 nv-config.php
-rwxrwxr-x 1 asterisk asterisk 14329 Feb 10 2008 nv-mailcall.php
-rwxrwxr-x 1 asterisk asterisk 6072 Sep 24 2008 nv-mailit.php
-rwxrwxr-x 1 asterisk asterisk 10490 Apr 20 10:34 nv-news.php
-rwxrwxr-x 1 asterisk asterisk 6545 Apr 12 15:10 nv-today.php
-rwxrwxr-x 1 asterisk asterisk 21537 Apr 2 13:07 nv-weather.php
-rwxrwxr-x 1 asterisk asterisk 12043 Apr 2 13:07 nv-weather-world.php
-rwxrwxr-x 1 asterisk asterisk 22243 Apr 2 13:07 nv-weather-zip.php

In each of these scripts, you'll find a variable near the top that controls the TTS engine: $ttspick = 0 ;

To use Cepstral as the TTS engine instead of Flite, just change the $ttspick value from 0 to 1 and save the file.

Email That Works With SendMail. It's always been a knuckle drill to get your new server to reliably send outbound emails. Assuming your Internet service provider doesn't block downstream mail servers, the Orgasmatron Installer will get this working reliably. You can test it out by logging into your server as root and issuing the following command using your real email address. If you get the email, you can move on.
echo "test" | mail -s testmessage yourname@gmail.com

If you didn't get the email, you probably have a provider such as Comcast that blocks port 25 in many areas of the country. The easiest way to solve this is to set up a free Gmail account and use Gmail to deliver outbound messages from your server. This message thread on the PBX in a Flash Forum will walk you through the setup process. There's also a Comcast solution if you'd prefer not to use Gmail.

Stay Tuned. Your eyes are probably glazing over about now. I know mine are. So we'll quit here for today. In our next episode, we'll tackle the rest of the goodies that make up the Orgasmatron Installer. Enjoy!

Continue reading Part III.

Continue reading Part IV (Monday, May 25).

Tip of the Week. Ever wanted a 20-seat conference bridge for under $9 a month with a local phone number in any of 2600+ rate centers all over United States? You can add load balancing and automatic failover for an extra $1 per month. After you use the Orgasmatron Installer, just set up a conference extension in FreePBX and then head over to the PBX in a Flash Forum to read all about the latest rage in DID providers.

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

 

Special Thanks to Our Generous Sponsors


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

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

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

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

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

Some Recent Nerd Vittles Articles of Interest...

Asterisk on Steroids: Introducing the Orgasmatron Installer

Thursday, April 9, 2009 / 15 Comments on Asterisk on Steroids: Introducing the Orgasmatron Installer

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

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

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

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

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

Getting Started. Even though the installation process is now a No-Brainer, you are well-advised to do some reading before you begin. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some precautions to protect your phone bill. Start by reading our Primer on Asterisk Security. Then read our PBX in a Flash and VPN in a Flash knols. If you’re still not asleep, there’s loads of additional documentation on the PBX in a Flash documentation web site.

Installation. Here’s a quick tutorial to get you started. First, install the 32-bit version of PBX in a Flash with Asterisk 1.4. Boot your system from the installation CD and type ksalt to begin. When your machine reboots, remove the CD and choose option A to load the most stable payload. When the install completes, reboot your system once again and login as root with the password you chose when you built your system. Now issue the following commands to bring your system current and protect your system passwords: update-scripts, update-fixes, passwd-master. You now have a PBX in a Flash base install. On a stand-alone machine, it takes about 30 minutes. On a virtual machine, it takes about half that time.

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

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

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

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

http://192.168.0.123/admin/

You’re NOT done yet!

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

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

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

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

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

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

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

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

Choosing a VoIP Provider. For this week, we’ll point you to some things to play with on your new server. Then next week, we’ll cover in detail how to customize every application that’s been loaded. For openers, we recommend you set up an account with Vitelity using our special link below. This gives your PBX a way to communicate with every telephone in the world, and it also gets you a real phone number for your new system… so that people can call you. Here’s how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you’re calling. If you’re in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask.

The VoIP world is new territory for some of you. Unlike the Ma Bell days, there’s really no reason not to have multiple VoIP providers especially for outbound calls. Depending upon where you are calling, calls may be cheaper using different providers for calls to different locations. So we recommend having at least two providers. Visit the PBX in a Flash Forum to get some ideas on choosing alternative providers.

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

  • D-E-M-O – Check out the Nerd Vittles Orgasmatron Demo
  • Z-I-P – Enter a five digit zip code for any U.S. weather report
  • 6-1-1 – Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 – Get the latest news and sports headlines from Yahoo News
  • T-I-D-E – Get today’s tides and lunar schedule for any U.S. port
  • F-A-X – Send a fax to an email address of your choice
  • 4-1-2 – 3-character phonebook lookup/dialer with AsteriDex
  • M-A-I-L – Record a message and deliver it to any email address
  • C-O-N-F – Set up a MeetMe Conference on the fly
  • 1-2-3 – Schedule a regular or recurring phone reminder
  • Dial *68 – Schedule a hotel-style wakeup call on any extension

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

Continue reading Part II.

Continue reading Part III.

Continue reading Part IV (Monday, May 25).

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

 

Special Thanks to Our Generous Sponsors


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

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

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

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

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

Some Recent Nerd Vittles Articles of Interest…

Another Dell with Asterisk, Dude: Introducing the Orgasmatron III for Dell’s New PowerEdge T100

Tuesday, December 16, 2008 / 18 Comments on Another Dell with Asterisk, Dude: Introducing the Orgasmatron III for Dell’s New PowerEdge T100

Dell finally kissed its SC440 server goodbye last week so we've been scrambling for a replacement VoIP candidate for Asterisk® that has performance sufficient to serve as a 50 to 100-user small business PBX without breaking the bank. It turns out Dell's PowerEdge T100 introduced in September is strikingly similar to the SC440 both in performance, design, and even price, but it scales a bit better. If it walks like a duck, quacks like a duck, and is priced like chicken feed, that's good enough for us.

In early December, we got our first new T100: a Dual Core Intel® Pentium®E2180, 2.0GHz processor with 1MB Cache, an 800MHz FSB, two 80GB 7.2K RPM Serial ATA 3Gbps 3.5-in Cabled Hard Drives connected to the onboard SATA controller, 512MB of 667MHz DDR2 RAM, a DVD-ROM Drive, and an On-Board Single Gigabit Network Adapter for $299. Sound familiar? It should. The T100 special pricing was virtually identical to the $299 special on the SC440 except Dell now has thrown in a DVD-ROM drive in lieu of the SC440's CD-ROM drive. For $19 more, you can bring the system up to 2GB of RAM which is an excellent idea. If you missed out, don't fret. There will be another deal in a week or two. Even the regular pricing on this unit with a Celeron 1.8GHz processor, 2 gigs of RAM, and two 80GB drives is only $339. And international pricing is equally competitive. We haven't yet seen the $199 single-drive U.S. price that appeared regularly with the SC440, but it shouldn't be too long given the current economy.

As for scaling, if you're interested in a growth path, you'll love the T100 compared to the SC440. It supports numerous processors up to the Quad Core Xeon 2.83GHz with 2x6M Cache and 1333MHz FSB as well as two one-terabyte SATA drives (just don't buy them from Dell 😯 ). And, unlike the SC440, the T100 accepts up to 8GB of RAM. So the remaining question: "Will the SC440 Orgasmatron II build work with the T100?" And the answer is "sort of." But have no fear, we've put Humpty back together again and have added even more bells and whistles to the new Orgasmatron III custom-designed for the T100 today. It now includes your own, free and private Hamachi VPN cloud for up to 16 computers.

To get email alerts when the T100 again goes on sale, go to techbargains.com. Then click on Send Email Deal Alert and fill out the form entering T100 as your search term. Be sure to confirm the alert by replying to the email.

If you want a cash rebate on your Dell purchase, use our eBates link to Dell or click on the coupon image in the right column of this article. It takes less than 30 seconds to sign up, and you get $5 (and so do we!) plus you receive 2% cash back on your Dell small business purchases which can be deposited directly into your PayPal account.

We expect these units will follow in the footsteps of their SC440 cousin and go on sale roughly every two weeks... so be ready! The T100 also is good news for our international friends because Dell now markets this machine virtually everywhere in the world at very competitive prices. It's selling for 40% off in the U.K. and 299€ in many European countries as we speak.

For long-time readers, you already know that we've identified what we believe to be the perfect Asterisk SIP phone, the Aastra 57i. But both of our previously anointed small business/home servers on which to run a production Asterisk system for 50-100 employees, the Everex gPC2 (aka "The WalMart Special") and the Dell SC440, are no more. So this build brings us current with Dell's very latest offering in the low-cost, high-performance server category and builds on the SC440 tradition of providing a quantum leap in performance and reliability compared with traditional home PCs. The ISO images you'll be downloading were captured as a backup on the flash drive of our new T100 lab machine. You can expect at least twice the performance on the PowerEdge T100 compared to the WalMart Special. Today's Orgasmatron III Build provides a preconfigured T100 installation on a 2-disk ISO image backup of the whole system using Mondo. And, NO, it won't work with any other hardware! Once you download the ISO images and burn your CDs, it's a 15-minute No-Brainer to install the entire image onto your own T100. Wait to install any add-on cards until after you complete the Orgasmatron install. You must have a T100 configured as above, or this Mondo restore may not work. So accept no substitutes, or you may end up with an Electronic Brick instead of an Orgasmatron.

We've preconfigured some extensions on your new system as well as outbound and incoming trunks from some terrific providers including our second homegrown entry for VoIP terminations. Joe Roper and his business partner in Spain now offer a terrific IAX VoIP termination service. You can choose penny a minute service in the U.S. and most of Canada, or you can opt for premium VoIP service at about 2¢ a minute in the U.S. International rates also are VERY reasonable! You literally can sign up for service, plug in your phones, and have a system in full operation in under an hour.

If you've missed our previous Orgasmatron articles, suffice it to say this is the Ultimate Kitchen Sink for Asterisk. From the time you insert the CD 'til you have a functioning Asterisk PBX with all the bells and whistles imaginable... just 15 minutes! In fact, it will take less time to create your new system than it will take you to finish reading this article. Please do BOTH! The Orgasmatron III includes PBX in a Flash 1.3 in all its glory including Asterisk 1.4.21.2 running under CentOS 5.2 with a version of Zaptel that actually works with legacy cards, plus the newly released FreePBX 2.5, a full-function fax server, a full-disk backup and restore solution (that actually works!), the latest Hamachi VPN software, every imaginable Nerd Vittles text-to-speech application for Asterisk, and so much more. Complete documentation for the TTS apps is available here.

  • Inbound and Outbound VoIP Faxing Using nvFax... finally!
  • FONmail for Asterisk to send voice messages to any email address on the planet
  • AsteriDex RoboDialer and Telephone Directory
  • Telephone Reminders with Support for Recurring Reminders and Web-based TTS Reminder Messages
  • NewsClips for Asterisk featuring Dozens of Yahoo News Feeds (TTS)
  • Weather Reports by Airport Code (TTS)
  • Weather Reports by ZIP Code (TTS)
  • Worldwide Weather Forecasts (TTS)
  • xTide for Asterisk (TTS)
  • MailCall for Asterisk: Get Your Email By Telephone (TTS)
  • TeleYapper 4.0 Message Broadcasting System
  • CallWho for Phone Lookup and Dialing of Entries in the AsteriDex Database (TTS)
  • TFTP Server with preconfigured setups for 10 Aastra 57i SIP telephones

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

  • Stealth AutoAttendant with Welcome and Application IVRs
  • Key Telephone Support Using Park and Parking Lot
  • Intercom/Paging Support
  • Bluetooth Proximity Detection with Automatic Call Forwarding to Cell Phone
  • DISA
  • Blacklisting with Web and Telephony Interfaces
  • CallerID Name Lookups from Numerous Providers
  • Weekly Automated System Backups to a Flash Drive
  • One Touch Day/Night Service
  • Music on Hold
  • Voicemail with Email Delivery of Messages and Pager Notification
  • Voicemail Blasting
  • Cell Phone Direct Dial
  • Call Forward: All, Busy, No Answer
  • Call Waiting
  • Call Pickup
  • Zap Barge
  • Call Transfer: Attended and Blind
  • Dictation Service with Email Delivery
  • Do Not Disturb
  • Gabcast
  • Phonebook Dial by Name
  • Speed Dial
  • Flite Text to Speech (TTS)
  • Windows Networking with SAMBA
  • Linux Firewall and Fail2Ban with SSH, HTTP, and SIP/IAX login protection
  • PBX in a Flash Software Update Service To Keep Your System Current
  • One-Click Cepstral TTS Install with Allison... Just Type install-cepstral

Prerequisites. As mentioned, you'll need a T100 configured with the specs outlined above including the 2GB RAM upgrade. We also recommend an 8GB USB flash drive on which to store automatic weekly backups of your new system. Just plug it into your new machine, and follow the simple steps below to activate Mondo. Every Sunday night, you'll get a new backup in ISO format on your flash drive. If something goes wrong on your system, copy the ISOs to CDs and reboot with Disk 1. It doesn't get any easier than that. And you can always check on the latest backup by issuing the command: usbcheck

Pay to Play. Greed has finally set in at Nerd Vittles. After all, Christmas is just around the corner! The download of this two-disk ISO image will set you back a whopping $10. In addition to covering the bandwidth and storage costs for the builds themselves, it also seems only fair that those using the builds help cover the hardware costs associated with these technology refreshes. When you compare our pricing to the Lime Green PBX offering from Dell... well, you don't really wanna know! There's one other little difference. Once you download our image from DreamHost, you are more than welcome to pass it along to as many of your friends and business acquaintances as you like. You can even do it electronically through the DreamHost Files Forever program. And, if you're inclined to host this image for your fellow man at no cost, be our guest... and thank you!

Bottom line: With a little patience waiting on Dell's next special, for about $300 and some lunch money, you'll have the slickest, newest, fastest, most reliable PBX and fax machine on the planet with rock-solid weekly backups and, of course, the availability of our one-of-a-kind PBX in a Flash Software Update Service! In fact, this may very well be The PerfectPBXâ„¢ even if we do say so.

Getting Started. Once you have your T100 in hand, take it out of the box, plug it into your LAN with DHCP and DNS support and Internet connectivity. You'll need a USB keyboard for typing temporarily. We also strongly recommend that you always keep your system running behind a NAT-based firewall/router. We strongly recommend the dirt-cheap dLink WBR-2310 WiFi router which handles NAT issues with VoIP masterfully. Don't redirect any ports to the machine and don't turn the PC on just yet.

Download the two ISO images for the T100 from here. Unzip the file and create two CDs from the ISO images. If you don't know how to create a CD from an ISO image, read that section from our previous article. In fact, read the whole article. It'll help you immensely down the road.

Once you've created your two CDs, turn on the T100 and quickly insert Disk 1 into the DVD drive and close the drive. When prompted, press F11 to choose the boot device and select the DVD-ROM drive. You'll note that the default T100 setup now apparently looks for a network boot device so you'll need to do a little BIOS reconfiguring, but you can do that at your convenience. F2 gets you into the T100 BIOS setup. Then choose Integrated Devices and, using the space bar, change Embedded Gb NIC from Enabled with PXE to simply Enabled. Press the escape key twice and then choose Save and Exit.

For now, choose the DVD-ROM drive as the boot device and proceed with the Mondo restore. If you don't see a Mondo Rescue screen within a minute or less, turn the machine off and then back on again. At the Mondo Rescue main screen, type nuke and press the Enter key. This will erase, repartition, and reformat your hard disk in case you didn't know. This is normal. If you get any kind of errors about incorrect drive or partition names and you really do have a T100, ignore them. Otherwise, halt the install by pressing CTL-ALT-DEL and remove the CD. You'll need to install PBX in a Flash using our standard ISO which is available here. Otherwise, go have a cup of coffee and come back in about 10 minutes. You'll be prompted to insert Disk 2 and press Enter to finish the install. When the second CD finishes, eject it and wait for the prompt. Then type "exit" and press Enter. Your T100 will reboot, and you're ready to go.

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

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

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

Now, using a web browser, go to the IP address of your new PBX in a Flash server. Click the Admin tab, the password is password. Then choose the FreePBX Administration button. Log in as maint with your new maint password. Before you do anything else, change ALL of the 10 extension passwords to something very secure... as if your phone bill depended upon it! Click Setup, Extensions and then choose each extension, modify BOTH the device secret and Voicemail Password, and click Submit. When you finish all the extensions, then reload the dialplan to save your changes. Finally, change your DISA password to something very, very secure: Setup, DISA, DISAmain, PIN. Reload your dialplan once again to save your changes.

Regardless of what you may read elsewhere, the Orgasmatron III has all the very latest security patches as of today. If you want more security, take our advice and add a hardware-based firewall/router between your Internet connection and your new Orgasmatron III and don't expose port 80 (the web interface) to the Internet!

Permanently Setting the IP Address. There are different schools of thought on whether to use a fixed or dynamic IP address. Most hardware-based routers support DHCP IP address reservations. The simplest way to permanently secure the existing IP address for your server is to reserve it on your router. If you'd prefer to assign your own IP address, we have included the deprecated netconfig utility which can be run after logging into your server as root. Sometimes you will need to run it once, enter your settings, reboot, and then repeat the drill. Then you should be all set. Either way, you need a permanent IP address for your machine when all is said and done. Once you have a permanent IP address, hop on over to dyndns.org and sign up for your own fully-qualified domain name (FQDN), e.g. mypbx.dyndns.org. You're going to need it for a whole host of things with your new PBX, and dyndns.org is about the easiest way to do it. Once you have your FQDN and DynDNS username and password, log in as root and edit: /etc/ddclient/ddclient.conf. Search (Ctl-W) for ***. Fill in your username and password and uncomment those two lines. Then search for *** again, uncomment the next three lines and fill in your fully-qualified domain name. Save the file and service ddclient restart. To make sure everything worked, issue the following command: ddclient -force. Assuming there are no errors, issue the following command to start ddclient each time your server reboots: /sbin/chkconfig --add ddclient. Now the IP address of your Asterisk server will always resolve to your FQDN from DynDNS. And anyone can call you via SIP for free using the following SIP URI: mothership@yourFQDN.dyndns.org. You can take this a step further and sign up for a free incoming phone number at ipkall.com. For your account type, choose SIP. For your SIP phone number, enter: mothership. For your SIP proxy, enter the fully-qualified domain name (FQDN) for your server, e.g. mypbx.dyndns.org. Choose a password and enter your real email address, and they will beam you a Washington state phone number within a day or so. You can't beat the price!

Getting Phones to Work Reliably. If you or the the person at the other end of your calls only hears half the conversation or if your calls get abruptly disconnected after a few minutes, it's probably because you forgot to add IP addresses to tell SIP how to communicate with your Asterisk server sitting behind a firewall. Edit /etc/asterisk/sip_custom.conf and add an entry for your external IP address and also for your local (internal) subnet where Asterisk resides. Then restart Asterisk: amportal restart.

externip=68.28.142.83
localnet=192.168.0.0/255.255.255.0

If you have a dynamic IP address and you set up ddclient above with your fully-qualified domain name, we've created a little script to keep these entries up to date automatically. Just edit the following file:

/var/lib/asterisk/agi-bin/ip.sh

Fill in the correct entries for your fqdn and localnet. Then uncomment the last line in /etc/crontab which runs ip.sh once every 5 minutes.

Adding Plain Old Phones. Before your new PBX will be of much use, you're going to need something to make and receive calls, i.e. a telephone. For today, you've got several choices: a POTS phone, a softphone, or a SIP phone (highly recommended). Option #1 and the best home solution is to use a Plain Old Telephone or your favorite cordless phone set (with 8-10 extensions) if you purchase a little device (the size of a pack of cigs) known as an SPA-2102. It's under $70. Be sure you specify that you want an unlocked device, meaning it doesn't force you to use a particular service provider. Once you get it, plug the device into your LAN, and then plug your phone instrument into the SPA-2102. Note that this adapter supports two-line cordless phones! Your router will hand out a private IP address for the SPA-2102 to talk on your network. You'll need the IP address of the SPA-2102 in order to configure it to work with Asterisk. After you connect the device to your network and a phone to the device, pick up the phone and dial ****. At the voice prompt, dial 110#. The device will tell you its DHCP-assigned IP address. Write it down and then access the configuration utility by pointing your web browser to that IP address.

Once the configuration utility displays in your web browser, click Admin Login and then Advanced in the upper right corner of the web page. When the page reloads, click the Line1 tab and then repeat this drill for the Line2 tab if you want to connect the device to two extensions on your Asterisk system. Scroll down the screen to the Proxy field in the Proxy and Registration section of the form. Type in the private IP address of your Asterisk system which you wrote down previously. Be sure the Register field is set to Yes and then move to the Subscriber Information section of the form. Assuming you're using the preconfigured extensions starting with 701, do the following. Enter House Phone as the Display Name. Enter 701 as the User ID. Enter your actual password for this extension in the Password field, and set Use Auth ID to No. Click the Submit All Changes button and wait for your Sipura to reset. In the Line 1 Status section of the Info tab, your device should show that it's Registered. You're done. Now repeat the drill for Line2 using extension 702. Pick up a phone and dial 1234# to test out BOTH extensions.

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


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

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

Log back into your server as root. Switch to the TFTP directory: cd /tftpboot. You'll notice that there are config files for up to 15 phones. Simply choose the extension number you wish to use for each phone AND rename each file (filenames are 701.cfg to 715.cfg) to the MAC address of each phone.cfg. Do NOT use hyphens or colons in the MAC address. Edit each of the .cfg files and replace the SIP line1 password with the new password you created for the extension using FreePBX. One final step and you'll be ready to load up your phones. We need to set the correct IP address to tell each phone where your server is located. So... issue the following command using the IP address of your new server instead of 192.168.0.123. Leave the rest of the command as it is!

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

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

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

TCP 80 - HTTP (needed to access the web sites on your server from the Net)
TCP 22 - SSH (needed if you want remote SSH access)
TCP 9001 - WebMin (needed if you want remote WebMin access... not recommended!!!)
UDP 10000-62000 - RTP (needed for SIP communications)
UDP 5004-5037 - SIP (ditto)
UDP 5039-5082 - SIP (ditto)
UDP 4569 - IAX2 (needed for IAX connection between Asterisk servers)

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

For outbound calling, we recommend you establish accounts with several providers. We've included the necessary setups for Joe Roper's new service for PBX in a Flash as well as Vitelity and AOL. To register for the service, just visit the web site and register. To sign up to the service in the USA and be charged in US Dollars, please sign up here. To sign up for the European Service and be charged in Euros, sign up here.

In addition to being one of the least expensive providers, there's also the premium service option. You can prefix any number with 000 to try it out. Give it a try. We think you'll be pleased with the service AND the pricing. DIDs for inbound service are not yet available, but Vitelity has lots of them, and there's a link below to get you started.

Vitelity: One of the Best Providers on the Planet. If you're seeking the best flexibility in choosing an area code and phone number plus reasonable entry level pricing plus high quality calls, then Vitelity is a winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. For PBX in a Flash users, sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. You can't beat the price (except with us) and the call quality is excellent as well. We've tried just about everybody.

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

To test things out, pick up a phone configured on your system and dial an area code and number of someone in the United States or Canada. Now get someone to call you using your new number. Presto! You have inbound and outbound phone service. And, if you'd like to see just how good SIP service can be, pick up a phone on your system and dial D-E-M-O. This will connect you to the PBX in a Flash hosted demo applications server at Aretta Communications.

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

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

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

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

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

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

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

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

canreinvite=no
context=from-pstn
dtmfmode=rfc2833
insecure=port,invite
nat=yes
type=user

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

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

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

Choosing a VoIP Provider That Supports Faxing. We've included a reliable fax solution in this build. You can review the details in this Nerd Vittles article. To test your machine, you can connect a real fax machine to one of the extensions using an SPA-2102. Then send a fax to extension 329 (F-A-X). But first you must configure your email address in two places using FreePBX: Setup, General Settings, Email address to have faxes emailed to AND Setup, Inbound Routes, any DID / any CID, fax Email. Once you've saved your settings, send the fax and see if it's delivered to your email address. If it works reliably, then the fax and email applications on your machine are configured correctly. Unfortunately, that's only half the battle. To receive faxes from outside your system, you'll also need a DID from a provider that supports faxing. And then it's still only about a 90% proposition... on a good day. We've tested this with many, many VoIP providers. Some work. Many don't. Some, such as Vitelity, offer a faxing service for a fee. Guess what? Their regular VoIP setup doesn't support faxing. Our old friends at Telasip.com still support faxing. We've also had good luck with Future-Nine and Teliax. You can read our fax dissertation here for more details. With the exception of the trunk setup covered in the article, all of the remaining setup steps already have been completed on your new server!

Interconnecting Two Asterisk Servers. We've preconfigured this build to support an IAX interconnect to a second PBX in a Flash system. The trunk setup for the second machine to match the setup on this build can be printed out. The filename is /root/MainPeerTrunkSetup.gif.

Choosing a Preferred Provider. Finally, you'll need to decide whether to use PIAF-USA or AOL or Vitelity as your primary terminations provider. HINT: Joe's new service is the cheapest! So we've set things up this way. This is handled in FreePBX in the Outbound Routes tab under the Default entry. You can adjust easily these in any way you like by adding trunks or moving entries up and down the list to change their priority. Just be sure to leave ENUM at the top of the list since ENUM calls are always free. If a free call isn't possible, your server will automatically drop down to the next trunk in the priority list. Don't add Vitelity to the list unless you have actually created a Vitelity account since they handle unsuccessful connections in a non-standard way which will cause FreePBX not to drop down to the next trunk to attempt a connection.

Activating the Stealth AutoAttendant for Inbound Calls. By default, all incoming calls are routed to the Day/Night Code 1 context which allows you to toggle calls between a Day setting and a Night setting by pressing *281. The Day setting for Code 1 is set to our Stealth Autoattendant which plays a brief greeting during which you can choose other options or direct dial extensions on your system before the call is passed to Ring Group 700. To change the options, edit MainIVR.

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

WARNINGS: If you update the version of Mondo shipped with this distribution to the current version using either yum or a standalone RPM, you will break your backup system. The advantage of the newer version is that it can create bootable flash drives with your backup image. The disadvantage is that the restore process croaks and locks up your machine. So don't update for the time being. We'll let you know when it's safe to upgrade.

Particularly if you have more than one drive in your system, be aware that the device name for your USB flash drive may differ from the setting of /dev/sdb1 that is preconfigured in this backup. This depends upon the number of internal hard disks and the Dude that built your Dell.

To safely activate backups on a stock T100 configured as we've outlined above, here are the mandatory steps:

1. Format every USB stick you plan to use for backups. Insert the USB flash drive into the right USB slot on the front of your Dell T100. Log into your server as root and type: /root/usbformat.sh. Your USB flash drive is now formatted. Repeat the process for any additional USB flash drives. WARNING: Do not use this script if you have added additional drives on your system as it may inadvertently reformat the wrong drive! The script assumes you have one or two internal SATA drives and one USB stick inserted in the right USB slot on the front of your Dell T100.

2. Assign the proper device name to Mondo and activate it: With a formatted USB flash drive in place, log into your server as root and type: /root/usbdevice.sh. You're all set. A backup will be made each Sunday night. If no flash drive is present, the backup will be saved in /etc/usbmondo.

3. Run a test backup: With a USB flash drive in place, log in as root, and type: /etc/cron.weekly/disk-backup.cron. To be sure it worked, see #4.

4. Check the contents of your USB stick regularly! Plug it into the front right USB port, log in as root, and type usbcheck. It's a good practice to check this on Mondays to be sure you got a fresh backup on Sunday night!!

Other Backup Options. Of course, there are some other backup options. FreePBX is preconfigured to make an automatic backup of your FreePBX data once a week. This is controlled by the settings in Tools, Backup and Restore, WeeklyBackup. It currently is set to make a backup every Wednesday morning. You also may want to consider off-site backups. Amazon's S3 service is preconfigured including all necessary software and scripts. All you need is an account and password. For detailed instructions, see this Nerd Vittles' article.

Installing Cepstral on Your New Server. If you want real text-to-speech with Allison's familiar voice, then you'll need to buy Cepstral. It's dirt cheap for single, non-commercial use. To install it, run install-cepstral from the command prompt while logged in as root. At one point you'll be asked whether to create a missing directory for the Cepstral installation. Be sure to type y at the prompt rather than just pressing the Enter key. Instructions for registering your copy of Cepstral are displayed when the install completes. For complete documentation, read our previous tutorial.

Creating Your Own Hamachi VPN Network. We've saved the best for last today. This latest Orgasmatron III build includes the Hamachi VPN network software. All you have to do is initialize it. Once configured, you can add as many as 16 computers (including Windows, Mac, and Linux machines) to your own private virtual private network. Communications between all of your systems then will be encrypted by simply connecting to the other systems using their VPN network addresses (5.x.x.x). For complete setup instructions, take a look at our VPN in a Flash knol on Google. The entire setup takes less than 5 minutes.

News Flash: As we put this article to bed last night, we tried one final experiment. We took the bootable USB flash drive from our VPN in a Flash build for the Aspire One NetBook that was featured last week and plugged it into the Dell T100. Guess what, Dude? Twelve minutes later we had a perfect clone of the Aspire One build on our new Dell T100. So, if you're looking for a state-of-the-art operating system with a fantastic GUI interface to pair up with Asterisk and PBX in a Flash, we may have another surprise for you to ring in the new year with your new T100. And it should work splendidly on the older SC440 as well as other machines with any industry-standard SATA drive. For 2009, PBX in a Flash perhaps should be renamed PBX on a Flash. Imagine carrying a full-featured, preconfigured PBX around on your keychain. Now that should impress even your nerdiest friends. There still are a few kinks with the latest version of Mondo which have forced us to build our own custom patches to get a successful restore, but we're oh so close... Stay tuned!


Special Thanks. As another year comes to a close, we want to take a moment to thank all of you for reading Nerd Vittles. About 50,000 folks from 137 countries around the globe read Nerd Vittles every week. The Nerd Vittles Official Flag above shows all of your home towns. Incidentally, the countries are ordered by the number of actual visitors from each country.

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

Continue reading Part II...

 

Special Thanks to Our Generous Sponsors


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

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

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

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

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