Posts tagged: enum

Google Dips Its Toes in the Icy SIP Waters… and Retreats

In case you missed it, Google announced at the end of last week that it will discontinue support of Gizmo5 on April 3. Many of us suspected this was the death knell for Google support of SIP given the popularity of its recent Gtalk enhancements to Google Voice. Well, not so fast! As Todd Vierling pointed out on his blog this past Saturday, Google has quietly added outbound SIP support to reach any Google Voice number. So, assuming your Google Voice number is 678-123-4567, anyone in the world can now call you via SIP by dialing +16781234567@sip.voice.google.com.

For those using Asterisk® and FreeSwitch systems , here’s what you need to do immediately. Register all of your Google Voice numbers in the ENUM systems so that other Asterisk and FreeSwitch systems worldwide can connect with you using your new Google SIP URI without any communications charges. This also means that SIP phones such as the Nortel 1535 Color Videophone using services such as sip2sip.info can call you for free. And all they’ll need to do is dial your 10-digit Google Voice number!

To sign up for ENUM service, go to both e164.org and enumplus.org and register your 10-digit Google Voice number. Be sure to use the syntax shown above for the SIP URI (including the + symbol), or the calls will fail. It only takes a minute to register. ENUM is implemented for outbound calls by default in all Incredible PBX and Orgasmatron builds. So, just by registering your Google Voice number with these two sites, it means every ENUM-enabled server can place free SIP calls to your Google Voice number via ENUM before using any other outbound trunk for which there might be a charge.

Of course, everyone won’t register their Google Voice number with the ENUM services. So how do you call those folks via SIP without incurring charges for the call? For those that install Incredible PBX (beginning yesterday), it’s automagic. Just dial any 10-digit number, and Incredible PBX will attempt to place the call via SIP before falling back to Google Voice. The call processing is instantaneous so don’t worry about call delay. Remember, we’re living in a Digital World.

FreePBX Setup. If you have an existing FreePBX-based Asterisk system or an earlier release of Incredible PBX, here’s how to retrofit your system to support free SIP calling to Google Voice numbers. Whenever an Asterisk server attempts to place a SIP call, it sends a SIP Invite packet to the receiving server. In the case of Google, if the number is not one of theirs, you’ll immediately get a Congestion message from FreePBX. In the FreePBX design, this means that the attempt to place the outbound call will drop down to the next available trunk in the current Outbound Route. So the trick here is to create a custom trunk to handle the SIP calls to Google. And then we’ll add that trunk above your existing trunks in the Outbound Route that handles calls matching 1NXXNXXXXXX and NXXNXXXXXX. So the recommended Trunk Sequence in your Default Outbound Route would look like this:

1. ENUM
2. google-sip
3. gvoice
4. vitel-outbound
5. OtherProvider

Using a web browser, open FreePBX and choose Setup, Trunks, Add Custom Trunk. Create the new Google-SIP Trunk so that it looks like the following. Don’t forget the 1 Prepend and + Dial Prefix entries!

Click Submit Changes to save your entries and then reload FreePBX when prompted.

Now choose Setup, Outbound Routes, and choose your Default outbound route. Modify the Trunk Sequence so that it matches what was outlined above. Click Submit Changes to save your entries and then reload FreePBX when prompted.

You’re done. Enjoy your new SIP-based Google Voice calling addition.

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 Google Voice number. Enjoy!

 

March 8 Update. Well, that was a quick dip. Beware the Ides of March! It was almost exactly two years ago that Google shut down SIP connectivity the first time. Hopefully, we’re not in for another two year wait. Read our original article about this and have a chuckle. But it looks like they’ve done it again. To restore your system to normal functionality, remove the Google-SIP trunk from your Outbound Route and be sure to delete your Google Voice numbers from the SIP registries at e164.org and enumplus.org. To suggest this is short-sighted (not to mention monetarily wasteful) would be an understatement. But perhaps Google wasn’t prepared for the onslaught of delighted users. Let’s hope so. :roll:

March 16 Update. It’s working again this morning! But now it’s not morning, and we’re dead in the water once more. Did we mention this might qualify as E-X-P-E-R-I-M-E-N-T-A-L?? See the comments below for up-to-the-minute updates.

Security Reminder. We mentioned this two years ago, but it’s worth repeating since it still has not been addressed. Google protects phone access to your Google Voice account with only a 4-digit PIN. When unanswered calls roll over to their voicemail system, anyone has the option of pressing * to be prompted for this PIN. It only takes 10,000 calls at most to guess any PIN, and that doesn’t take very long with SIP and an automated dialer. Once someone has your PIN, in addition to listening to your voicmail messages, they also can press 2 to place an outbound call to anywhere in the world… on your nickel. So… don’t load up your account with your entire life savings unless you don’t mind losing it. :roll:

Originally published: Monday, March 7, 2011

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

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

 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up 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. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 

Some Recent Nerd Vittles Articles of Interest…

Posted in Technology, Telephony by , Monday, March 7, 2011 1:01 am Comments (18)

Tags: , , , , ,

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

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

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

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

So we had all the tools necessary to reengineer, design and build the all-new Incredible PBX for Asterisk 1.8. What used to be a somewhat kludgey, dual-call, dual-provider Google Voice implementation to take advantage of Google's free PSTN calling in the U.S. and Canada with Asterisk 1.4 and 1.6 is now a zippy-quick, Gtalk-based calling platform that rivals the best SIP-to-SIP calls on the planet and provides virtually instantaneous PSTN connections to almost anybody, anywhere. Trust us! Except for the price which is now free, you'll never know you weren't connected via Ma Bell's overpriced long-distance lines and neither will the Little Mrs. And, yes, our recommended $50 Nortel SIP videophone is plug-and-play on extensions 701 and 702.

Just download the latest PBX in a Flash ISO, burn to then boot from the PIAF CD, choose the Purple Edition to load Asterisk 1.8 and FreePBX 2.8, and then install the new Incredible PBX for Asterisk 1.8. In about an hour, you'll have a turnkey PBX with a local phone number and free calling in the U.S. and Canada via your own Google Voice account plus dozens and dozens of terrific Asterisk applications to keep your head spinning for months.

Thanks to its Zero Internet Footprint™ design, The Incredible PBX remains the most secure Asterisk-based PBX around. What this means is The Incredible PBX™ has been engineered to sit safely behind a NAT-based, hardware firewall with minimal port exposure to your actual server. And you won't find a more full-featured Personal Branch Exchange™ at any price.

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

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

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

Installing The Incredible PBX. The installation process is simple and straight-forward. Here are the 5 Easy Steps to Free Calling, and The Incredible PBX will be ready to receive and make free U.S./Canada calls immediately:

1. Install PBX in a Flash Purple Edition
2. Download & run The Incredible PBX 1.8 installer
3. Run passwd-master on your PIAF server
4. Map UDP 5222 on firewall to PIAF server
5. Configure a softphone or SIP telephone

Installing PBX in a Flash. Here's a quick tutorial to get PBX in a Flash installed. To use Incredible PBX for Asterisk 1.8, you must install the latest 32-bit version of PBX in a Flash.3 Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS operating system. That hasn't changed. But, once CentOS is installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities including all of the new Google Voice components. Just choose the new Purple Payload to get the latest Asterisk 1.8 release and all of the Google Voice goodies!

You can download the 32-bit PIAF from SourceForge or one of our download mirrors. Burn the ISO to a CD. Then boot from the installation CD and press the Enter key to begin.

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

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

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

NOTE: For previous users of PBX in a Flash, be aware that this new version automatically runs update-programs and update-fixes for you. You still should set your FreePBX passwords by running passwd-master after The Incredible PBX installer finishes!

Configuring Google Voice. You'll need a dedicated Google Voice account to support The Incredible PBX. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So why take the chance. Keep this account a secret!

We've tested this extensively using an existing Gmail account, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Set up a dedicated Gmail and Google Voice account, and use it exclusively with The Incredible PBX. Google Voice no longer is by invitation only so, if you're in the U.S. or have a friend that is, head over to the Google Voice site and register. If you're living on another continent, see MisterQ's posting for some tips on getting set up.

You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work... in either direction. Google used to permit outbound Gtalk calls using a fake CallerID, but that obviously led to abuse so it's over! You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don't skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you'd like in Settings, Voice Setting, Phones. But...

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

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

  • Call Screening - OFF
  • Call Presentation - OFF
  • Caller ID (In) - Display Caller's Number
  • Caller ID (Out) - Don't Change Anything
  • Do Not Disturb - OFF

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

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

cd /root
wget http://incrediblepbx.com/incrediblepbx18.x
chmod +x incrediblepbx18.x
./incrediblepbx18.x
passwd-master

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

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

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

The Google Voice Account Name is the Gmail address for your new dedicated account, e.g. joeschmo@gmail.com. Don't forget @gmail.com! The Google Voice Password is the password for this dedicated account. The Gmail Notification Address is the email address where you wish to receive alerts when incoming and outgoing Google Voice calls are placed using The Incredible PBX. And your FreePBX maint Password is the password you'll use to access FreePBX. You'll actually set it by running passwd-master after The Incredible PBX completes. We need this password to properly configure the CallerID Superfecta for you. By the way, none of this confidential information ever leaves your machine... just in case you were wondering. :roll:

Now have another 15-minute cup of coffee, and consider a modest donation to Nerd Vittles... for all of our hard work. :wink: You'll find a link at the top of the page. While you're waiting (and so you don't forget), go ahead and configure your hardware-based firewall to support Google Voice. See the next section for what's required. Without completing this firewall configuration step, no calls will work! When the installer finishes, READ THE SCREEN just for grins.

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

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

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

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

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

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

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

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

Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you'll want a real SIP telephone such as the $50 Nortel color videophone we've recommended above. You'll also find lots of additional recommendations on Nerd Vittles and in the PBX in a Flash Forum. If you're like us, we want to make damn sure this stuff works before you shell out any money. So, for today, let's download a terrific (free) softphone to get you started. We recommend X-Lite because there are versions for Windows, Mac, and Linux. So download your favorite from this link. Install and run X-Lite on your Desktop. At the top of the phone, click on the Down Arrow and choose SIP Account Settings, Add. Enter the following information using your actual password for extension 701 and the actual IP address of your Incredible PBX server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

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

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

Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust the settings in /etc/asterisk/sip_custom.conf. Just uncomment the first two lines by removing the semicolons. Then replace 173.15.238.123 with your public IP address, and replace 192.168.0.0 with the subnet address of your private network. There are similar settings in gtalk.conf that can be activated although we've never had to use them. In fact, we've never had to use any of these settings. After making these changes, save the file(s) and restart Asterisk with the command: amportal restart.

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

Adding Multiple Google Voice Trunks. Thanks to rentpbx on our forums, adding support for multiple Google Voice trunks is now a five-minute operation. Once you have your initial setup running smoothly, hop on over to the forums and check out this Incredible solution.

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

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

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

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

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

Enabling Google Voicemail. Some have requested a way to retain Google's voicemail system for unanswered calls in lieu of using Asterisk voicemail. The advantage is that Google offers a free transcription service for voicemail messages. To activate this, you'll need to edit the [googlein] context in extensions_custom.conf in /etc/asterisk. Just modify the last four lines in the context so that they look like this and then restart Asterisk: amportal restart

;exten => s,n(regcall),Answer
;exten => s,n,SendDTMF(1)
exten => s,n(regcall),Set(DIAL_OPTIONS=${DIAL_OPTIONS}aD(:1))
exten => s,n,Goto(from-trunk,gv-incoming,1)

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

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

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

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

And here's a sample from an install we just completed. We'll have more details and additional utilities for your use in coming weeks. Stay tuned!


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

Special Thanks. It's hard to know where to start in expressing our gratitude for all of the participants that made today's incredibly simple-to-use product possible. Please bear with us. To Mark Spencer, Malcolm Davenport, and the rest of the Asterisk development team, thanks for a much improved Asterisk. To Philippe Sultan and his co-developers, thank you for finally making Jabber jabber with Asterisk. To Leif Madsen, our special thanks for your early pioneering work with Gtalk and Jabber which got this ball rolling. To Philippe Lindheimer & Co., thanks for FreePBX 2.8 which really makes Asterisk shine. To Lefteris Zafiris, thank you for making Flite work with Asterisk 1.8 thereby preserving all of the Nerd Vittles text-to-speech applications. To Darren Sessions, thanks for whipping app_swift into shape and restoring Cepstral and commercial TTS applications to the land of the living with Asterisk 1.8. And to our pal, Tom King, we couldn't have done it without you. You rolled up your sleeves and really made Asterisk 1.8.0 sit up and bark. No one will quite understand what an endeavor that was until they try it themselves. You've made it look so easy. And, finally, to our dozens of beta testers, THANK YOU! We've implemented almost all of your suggestions.

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

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

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

Originally published: Monday, November 1, 2010

VoIP Virtualization with Incredible PBX: OpenVZ and Cloud Solutions

Safely Interconnecting Asterisk Servers for Free Calling

Adding Skype to The Incredible PBX

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

Adding Remotes, Preserving Security with The Incredible PBX

Remote Phone Meets Travelin' Man with The Incredible PBX

Continue reading Part II.

Continue reading Part III.

Continue reading Part IV.

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



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

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

 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up 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. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 

Some Recent Nerd Vittles Articles of Interest...

  1. For 64-bit systems with Asterisk 1.8, use the Cepstral install procedures outlined in this Nerd Vittles article. [↩]
  2. If you use the recommended Acer Aspire Revo, be advised that it does NOT include a CD/DVD drive. You will need an external USB drive to load the software. Some of these work with CentOS, and some don't. Most HP and Sony drives work; however, we strongly recommend you purchase an external DVD drive from a merchant that will accept returns, e.g. Best Buy, WalMart, Office Depot, Office Max, Staples. You also can run The Incredible PBX on a virtual machine such as the free Proxmox server. Another less costly (but untested) option might be this Shuttle from NewEgg: $185 with free shipping. Use Promo Code: EMCYTZT220 [↩]
  3. HINT: Version 1.7.5.6 recommended, but 1.7.5.5.3+ ISOs also work just fine. [↩]

Posted in Technology, Telephony by , Friday, December 10, 2010 1:00 am Comments (91)

Tags: , , , , , , , ,

The Incredible PBX: Adding a Free Skype Gateway to Asterisk

Last week we got The Incredible PBX all set up with free worldwide SIP calls, free U.S./Canada PSTN calls using Google Voice with SIPgate or IPkall, and rock-solid Asterisk® security using our new Zero Internet Footprint™ design. Because of licensing restrictions, we couldn't include Skype out of the box. If you're an individual and not a business, today we'll walk you through adding free Skype calling worldwide to your Incredible PBX. With today's addition, the Incredible PBX now provides free calling to nearly a billion phones around the world via Skype, SIP, ENUM, FreeNUM, and U.S./Canada PSTN connections. Yowza!

If you use the recommended hardware, today's setup procedure takes less than 10 minutes! Once it's complete, inbound and outbound Skype calling is totally transparent on your Incredible PBX. To reach a Skype number, just dial * plus the user's Skype name from any phone with an alphanumeric keypad. To place a Skype Out call (fees apply), dial 8 plus the user's area code and number. When your 500 million friends on Skype contact you using your Skype name, all of your Incredible PBX phones will ring just like any other inbound call. What's the difference in today's solution and Digium®'s commercial Skype for Asterisk product? For openers, our solution is $66 cheaper. It's free! And, if you're an individual, you won't need Skype's commercial Business Control Panel to make calls. Functionally, the results with your Incredible PBX Skype implementation are identical.1

To make the Skype Magic work, you'll need three pieces of software in addition to The Incredible PBX obviously: Sun's 6u12 Java SE Development Kit, Skype's Static Edition for Linux plus an existing Skype account, and Greg Dorfuss' SipToSis product which manages the Skype Gateway to Asterisk.

As far as hardware is concerned, we're assuming you're using our recommended $200 Acer Aspire Revo to host your Incredible PBX. With other hardware, your mileage may vary because CentOS 5.4 may or may not support your audio card and graphics mode with your video card. Both are required to get Skype working properly under X-Windows. If you have problems with some other type of hardware, take a look at the tips in our previous article on Setting Up a Skype Gateway to Asterisk as well as the comments. Better yet, visit your neighborhood Best Buy and purchase an Aspire Revo for a hassle-free install.

Installing JDK. Using your favorite browser, go to Sun's 6u12 Java SE Development Kit website, choose Linux for the platform, and agree to the license. Click Continue. Download jdk-6u12-linux-i586-rpm.bin and copy it to the /root directory of your Incredible PBX. Next, make the file executable (chmod +x jdk-6u12-linux-i586-rpm.bin). Then run it: ./jdk-6u12-linux-i586-rpm.bin. Scroll down the wordy license agreement AGAIN and type yes. Java 1.6 then will be installed on your system. Check to be sure Java was properly installed with this command: rpm -q jdk.

Installing Skype and SipToSis. Now we're ready to load the remaining components. While still logged into your Incredible PBX as root, download and run the skype-setup script2:

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

Activating Your Skype Gateway. Now we're ready to place your Skype gateway in production. You'll need to perform these steps from the console on your Incredible PBX since we have to run Skype in graphics mode. This may look complicated. It's really not. It's just a bit tedious to figure out the sequence of steps, but we've done that part for you.

WARNING: Be sure that you use a dedicated Skype account on this server! Do not run the same Skype account on any other server or desktop, or it fails!

1. Start up X-Windows: xinit3

2. Start up Skype. While still logged into your server as root, issue the following commands:

cd /root/skype/skype_static-2.0.0.72
./skype

Now log in to Skype with your Skype name and password. Be sure to set Skype to autologin whenever it is started. Then, in the Skype configuration option, set Skype to always run minimized. Save your settings.

Place a Skype Test Call4 to echo123 to be sure your audio settings are set correctly. Again, with the Aspire Revo, this won't be a problem assuming you have plugged in a microphone and speakers. These can be disconnected after you're sure things are working properly. HINT: Intel Atom-based motherboards are a piece o' cake!

Once you've got Skype working and all of the Skype settings configured above, shut down Skype.

3. Restart Skype in Background Mode: ./skype &

Be sure to write down the PID for Skype in case you need to kill the job if something goes wrong. :-) If you forget the PID, you can obtain it with this command: pgrep skype. You can kill Skype with the following command using your actual PID instead of 12345: kill 12345.

4. Start up SipToSis: Press Enter if the command prompt doesn't reappear. Then...

cd /siptosis
./SipToSis_linux

A message from Skype will pop up asking if you want to authorize external use of Skype: yes. Important: Be sure to select the Checkbox to save this setting for future connections!

5. Testing Skype. Go to a softphone (X-Lite recommended!) connected to an extension on your Incredible PBX and dial *echo123. You should be connected to the Skype Call Testing Service. Try *nerdvittles for the Nerd Vittles Demo.

Assuming you have a little money in your Skype Out account, go to any extension connected to your Asterisk server and dial 8 + your home phone number. This will place the outbound call through SkypeOut at 2¢ a minute.

Reboot your server when you're sure everything is working properly.

GUI Tips. Here are a few navigation tips for managing your Asterisk console on your Incredible PBX:

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

2. Ctrl-Alt-F7 gets you back to the SipToSis/Skype session. You can kill SipToSis by holding down Ctrl-C for several seconds. To decipher your SipToSis PID: pgrep -f SipToSis. To kill SipToSis: kill pid# (that you wrote down). To kill Skype: kill pid# (that you wrote down). To restart Skype: skype & and to restart SipToSis, just issue the command again: ./SipToSis_linux

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

Automating the Skype Gateway Startup. Once everything is working reliably, reboot your server again, log in as root, and issue the command: /root/skype-start. Place a test call again using a softphone on your Incredible PBX. If everything works fine, you now can add the skype-start command to your server's startup script, and you're all set.

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

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

SIP/joeschmo@127.0.0.1:5070

Security Warning. Do NOT expose UDP port 5070 to the Internet by opening a port on your hardware firewall. You do not need UDP 5070 exposed to the Internet to implement today's gateway solution for inbound or outbound Skype calling from your server!

Enjoy!

Update: As of May 1, you now can set your Google Voice number as your Skype CallerID number. Previously, Google Voice blocked the verification SMS messages, but no longer. Thanks, @zsafwan.

Adding Multiple Google Voice Trunks to The Incredible PBX



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.

 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up 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. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 

Some Recent Nerd Vittles Articles of Interest...

  1. Skype and this suggested implementation are intended for individual use. Your use is, of course, governed by the Skype Terms of Service. [↩]
  2. Here are the actual commands in the skype-setup script if you'd prefer to execute them one at a time:

    cd /root
    mkdir skype
    cd skype
    wget http://www.skype.com/go/getskype-linux-beta-static
    tar jxvf skype_static*
    yum install xorg-x11-server-Xvfb
    yum install qt4
    yum install xterm
    yum install libXScrnSaver.i386
    wget http://pbxinaflash.net/source/skype/siptosis.tgz
    cd /root
    wget http://incrediblepbx.com/skype-start
    chmod +x skype-start
    cp skype-start skype/.
    cd /
    tar zxvf /root/skype/siptosis.tgz
    cd /root


    [↩]

  3. Starting xinit won't be a problem on the Aspire Revo. But, if xinit won't start on your particular machine, you may need to create /etc/X11/xorg.conf. Here's a generic config file that should work fine for our purposes:

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

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

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

    [↩]

  4. If the test call fails with a bad audio message, go into Options, Sound Devices and reconfigure your Audio settings until you can place the test call successfully. Otherwise, none of the rest will work! [↩]

Posted in Technology, Telephony by , Monday, April 26, 2010 5:23 am Comments (30)

Tags: , , , , , , ,

Introducing ISN: Free SIP Dialing From Any Asterisk Phone

Wouldn't it be nice to pick up any telephone on your Asterisk® system and place free SIP calls to anywhere in the world by dialing joe@sip.asterisk.com or any SIP URI? The problem, of course, is that most phones don't include alphanumeric keyboards much less the @ symbol. Well, not to worry. A group of Asterisk gurus headed up by John Todd came up with a clever plan using DNS that lets you dial any SIP URI using the 10 numeric keys plus the asterisk key on any standard telephone keypad. Today, we'll show you how to set up your Asterisk system to support ISN's (aka ITAD Subscriber Numbers).

Overview. In laymen's terms, the trick to ISN dialing is that we pass a number such as 1234*1061 to a DNS server that knows how to translate the numeric sequence into a SIP URI that looks like this: 1234@sip.pbxinaflash.com. In short, it takes the number after the asterisk and resolves it to a fully-qualified domain name which is preconfigured at freenum.org. And the result is inter-domain numeric SIP addressing using ordinary telephone instruments. For our recommended setup, you'll actually dial ISN numbers like this: **1234*1061. The leading asterisks will tell FreePBX to treat this as an ISN dial string.1

Prerequisites. We're assuming that you already have one of the FreePBX-enhanced Asterisk aggregations in place such as PBX in a Flash. If not, start there and then run the Orgasmatron Installer which provides all of the SIP URI functionality you'll need for this project. If you're not using PBX in a Flash, then review our tutorial on SIP URI's which will walk you through getting this functionality set up on your FreePBX-enhanced Asterisk server.

Adjusting Your Phones to Support ISN Dialing. We'll be using a somewhat different dial plan to make ISN calls so you'll probably have to adjust the default dialplan on your actual phones or ATA to get this to work. If you can place ISN calls with a softphone but you get a fast busy when you dial the same number on your hardware-based phones, then it's a dialplan problem. For Aastra phones, you can access the Aastra dialplan settings with a web browser. Just go to the IP address of the phone and login with admin:22222. Click on the Preferences option and you should see Local Dial Plan at the top of the page with an entry that looks like this: x+#|xx+*. Just change it to: x+#|xx+*|'*'xx+* and click the Save Settings button. No reboot of the phone is required. Notice that we've enclosed the asterisk in single quotes in the third option. That's the trick to getting Aastra phones to recognize * as part of an actual dial string. If you're using other phones, consult your user's guide for tips on modifying your dialplan to accommodate an asterisk as the first character in the dial string.

Enabling Outbound ISN Dialing. There are a number of ways to get ISN outbound dialing to work with Asterisk. We're going to show you a couple of methods. You can either set up a trunk and outbound route to handle the calls, or you can add an extension to your system which actual prompts for the ISN number when you dial that extension. There are also two ways to look up ISN numbers at freenum.org. The preferred method is using DNS queries with the new Asterisk ENUMLOOKUP function. An alternative method (which is especially useful with older versions of Asterisk that do not support ENUMLOOKUP) is to use FreeNUM's external public resolver to map ISN dial strings to SIP URIs. With PBX in a Flash and Asterisk 1.4.21.2 or later, both methods work.

Implementing the Trunk Method for ISN Dialing. With this option, you'll be able to pick up any (properly configured) phone on your Asterisk system and dial **1234*1061 to complete a free ISN SIP call. To set this up, we'll add a new trunk and outbound route in FreePBX. Then we'll insert a dialplan script in extensions_custom.conf to finish up. Once you reload your Asterisk dialplan, you'll be good to go.

Open FreePBX in a web browser, and choose Admin, Setup, Trunks, Add Trunk, Add Custom Trunk. Leave the General Settings blank for now. In the Dial Rules, insert X.*X. (be sure to include trailing period!) and, for the Custom Dial String, insert: local/$OUTNUM$@freenum. Click the Submit button to save your settings and reload the dialplan when prompted. Now add an Outbound Route called OutFreeNUM. For the Dial Pattern, use **|X.*X. with the trailing period again. For the Outbound Route Dial Pattern, you can get more elaborate so that you don't have to dial the ** prefix. Just be aware that this may not work with all handsets (including the Aastra's). It does work well with Zoiper softphones. Here's the dial pattern we actually use. With this dial pattern, you can dial most ISN numbers directly with no prefix, e.g. 16781234567*1061 works fine.

**|X.*X.
1NXXNXXXXXX*X.
NXXNXXXXXX*X.
XX*X.
XXX*X.
XXXX*X.
XXXXX*X.
XXXXXX*X.
XXXXXXX*X.

For the Trunk Sequence, choose local/$OUTNUM$@freenum. Save your entries and reload the dialplan once more.

Finally, log into your server as root and edit extensions_custom.conf in /etc/asterisk. At the bottom of the file, insert the following code:

[freenum]
exten => _X.,1,Set(TIMEOUT(absolute)=10800)
exten => _X.,2,NoOp(Number to Call: ${EXTEN})
exten => _X.,3,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})
exten => _X.,4,GotoIf($["${isnresult}"=""]?6:5)
exten => _X.,5,Dial(SIP/${isnresult},40,r)
exten => _X.,6,Background(ss-noservice)
exten => _X.,7,Congestion
exten => _X.,8,Hangup
exten => h,1,Hangup
exten => i,1,Hangup
exten => T,1,Hangup

Make sure you eliminate the line-wrap on line 3 above. Then save the file and reload your dialplan: asterisk -rx "dialplan reload". Now place a test call by dialing: **1234*1061. If the call doesn't connect to Nerd Vittles' demo site, check the Asterisk CLI and fix any reported errors.

Implementing the Extension Method for ISN Dialing. With this option, you'll be able to pick up any phone on your Asterisk system and dial FREE (3733) to place an ISN call. You'll be prompted to enter the number using the following format: 1234*1061. Note that there are no leading asterisks with this method. Instead of using ENUMLOOKUP to find the ISN number, we'll use FreeNUM's external public resolver to do the ISN translation into a SIP URI.

Log into your Asterisk server as root and edit extensions_custom.conf in /etc/asterisk. At the bottom of the file, insert the following context:

[custom-freenum]
exten => s,1,Answer
exten => s,2,Wait(2)
exten => s,3,Background(pls-entr-num-uwish2-call)
exten => s,4,Read(NUM2CALL,beep,30)
exten => s,5,GotoIf($["foo${NUM2CALL}" = "foo"]?10)
exten => s,6,Set(TIMEOUT(absolute)=10800)
exten => s,7,Background(pls-hold-while-try)
exten => s,8,Dial(SIP/${NUM2CALL}@public.freenum.org,30,m)
exten => s,9,Congestion
exten => s,10,Hangup
exten => h,1,Hangup
exten => i,1,Hangup
exten => T,1,Hangup

Now move to the top of the file and insert the following line in the [from-internal-custom] context:

exten => 3733,1,Goto(custom-freenum,s,1)

Save the changes you've made to the file and then edit (or create, if necessary) sip_custom.conf and insert the following line:

promiscredir=yes

Save the file and then restart Asterisk: amportal restart. Now place a test call by dialing 3733. When prompted for the ISN number, enter 1234*1061 and press # to avoid the timeout delay. Be aware that on non-FreePBX systems, this code would go in sip.conf; however, that file gets overwritten with any FreePBX reload. Hence the reason that we've placed the code in sip_custom.conf.

Creating a SIP URI for Your Asterisk Server. Before you can receive any inbound calls with ISN dialing, you'll need at least one SIP URI for your Asterisk server. The format of a SIP URI is much like an email address: somename@yourdomain.dyndns.org or somenumber@yourdomain.dyndns.org. Step 1 is to register a fully-qualified domain name (FQDN) for your Asterisk server. Step 2 is to actually set up the SIP URI's on your server.

If you already have a registered domain, then we recommend you create a sip subdomain: sip.yourname.org. Then point that subdomain to the IP address of your Asterisk server. If your Asterisk server has a dynamic IP address, then register a subdomain with a service such as dyndns.org and point that domain at your Asterisk server. We've previously covered how to install software on your Asterisk server to make sure your FQDN always resolves to the correct dynamic IP address. Here's the link for DNS-O-Matic.

Once you have FQDN covered, you're ready to set up a SIP URI. With Orgasmastron builds of PBX in a Flash, the work already has been done for you. You should already have a SIP URI of mothership@yourFQDN. For everyone else, the drill involves moving a copy of the [from-sip-external] context into extensions_override_freepbx.conf in /etc/asterisk so that it can be edited without risking an overwrite from FreePBX. To find out the location of the [from-sip-external] context, issue the following commands while logged into your server as root:

cd /etc/asterisk
grep from-sip-external *

The result will look something like this:

extensions.conf:[from-sip-external]
extensions_override_freepbx.conf:[from-sip-external]
sip_general_additional.conf:context=from-sip-external

If the middle line is there, the context already has been copied over. Otherwise, list out the file showing [from-sip-external] which varies depending upon your version of FreePBX: cat extensions.conf. Now cut-and-paste the entire [from-sip-external] context into extensions_override_freepbx.conf. Then edit the override file and add an entry for each SIP URI you wish to create. The entries should be inserted just below the exten => s,1... line. Here are some samples:

exten => 16781234567,1,Goto(from-trunk,${DID},1)

This entry would let you control the routing of 16781234567 by creating a new incoming route in FreePBX with a DID entry of 16781234567. Then you can point the SIP URI to any FreePBX resource, e.g. an extension, ring group, IVR.

exten => e164,1,Goto(from-trunk,e164,1)

This entry would route e164@yourFQDN to the Inbound Route created for a DID number entry of e164.

exten => 18431234567,1,Goto(custom-windyhouse,s,1)

This entry would route incoming calls to 18431234567@yourFQDN to s,1 in a custom context called [custom-windyhouse] in extensions_custom.conf.

exten => 17065439876,1,Dial(SIP/17066313456@sip.otherdomain.com)


This entry would route incoming calls to 17065439876@yourFQDN to another SIP URI.

exten => 12021234567,1,Dial(local/12029876543@from-internal)

This entry would route incoming calls to 12021234567@yourFQDN to a cellphone at 12029876543 using your Asterisk dialplan to choose an appropriate trunk for the call.

exten => 18883331212,1,Dial(SIP/skype_joe@proxy01.sipphone.com)

This entry would route incoming calls to 18883331212@yourFQDN to a Skype user named joe using the free Gizmo5 gateway.

Once you've made all desired SIP URI entries, save the override file and reload your Asterisk dialplan.

Using the PBX in a Flash ITAD Number. So you're probably asking, "What's in this for me?" Well, a couple of things actually. First, if you're a PBX in a Flash user, we want you to join our free calling network. We already have reserved the 1061 ITAD number for our group. Just cut-and-paste the form below, fill in the blanks, and email it to us. We'll set up an ISN number for your server (one per customer, please) so that others can contact you without spending a dime. The other option is to obtain your own ITAD number for your organization and set it up on your own server. We'll get to that in a minute.

If you want to join our club (and we really don't mind if you're not using PBX in a Flash), then cut-and-paste the form below into your email and fill it out. And here's the email link. Once we receive your request, we'll set up an ISN number for you that matches your existing phone number. So, if your phone number is 16781234567, your new ISN number will be 16781234567*1061. Please include your international codes with your phone number. Before we activate your ISN number, we'll place a test call to your SIP URI to verify it's working. Please be sure it is before applying. :-)

Name:
Mailing Address:
Phone Number:
SIP URI for Your Server: _____________@_____________________________
ISN Number (leave blank):
Publish Entry in Directory? Yes or No (choose one)

Obtaining Your Own ITAD Number. We know there are lots of you that prefer to do things yourself. And that's perfectly fine. We're going to quickly show you how. But, if you want to be included in the PBX in a Flash directory, please send us the form above with your own ISN contact number once you get things working.

To get your own ITAD number, visit this link and follow the instructions for requesting your own number. It's easy, but detail matters so do it right the first time! Within a few days, you'll get your shiny new number. And, in a few more days, freenum.org will notify you that your account has been established.

Setting Up An ISN Account at FreeNum.org. Once you receive your login credentials from FreeNUM, log in to your account. Leave the DNS Wildcard setting the way it is. All you have to do is insert your fully-qualified domain name in the FQDN placeholder. For example, if your FQDN were sip.big.edu, then the last part of the DNS entry should look like this:

sip:\\1@sip.big.edu!" .

Save your entry and wait an hour. Then test it by dialing your new ISN number or, after logging into your server as root, use a command like the following. Turn your SIP URI around from 6781234567*1061 so that it looks like this:

dig @freenum.org NAPTR 7.6.5.4.3.2.1.8.7.6.1061.freenum.org.


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

Aretta Introduces Free NetPBX. In an industry first, Aretta Communications is rolling out a free Asterisk hosted solution known as NetPBX Free Edition. The only cost is for the minutes you use, and the free hosted service will support one inbound or outbound call at a time. Everything including the SIP trunking is preconfigured so the system is literally plug-and-play. We'll provide a more in-depth review once we've had some time to play.

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.

 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up 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. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 

Some Recent Nerd Vittles Articles of Interest...

  1. The dial string has been modified a bit to mesh with special dial codes in FreePBX. See the comments for details. [↩]

Posted in Networking, Technology, Telephony by , Wednesday, June 24, 2009 4:51 pm Comments (8)

Tags: , , , , , ,

Adding Post-Dial Processing to Asterisk and FreePBX Dialplans

Last week we introduced a couple of new free calling options for Asterisk®: ENUM and Gizmo5's Backdoor Dialing. But one of the limitations of the Gizmo5 service in particular was the need for a 0101 prefix in order to trigger a free call as opposed to a pay per minute call to the same number. This highlighted a pretty serious limitation in the way FreePBX processes most outbound calls. As we indicated, the process goes something like this. After a caller dials a number, FreePBX searches through its Outbound Routes (from top to bottom) looking for a match on the dial string. Once it finds one, FreePBX then initiates calls beginning with the first trunk in the trunk priorities list for that outbound route. If the call is completed, no further call processing takes place. And a completed call includes a call that is either answered or rings busy. If a call is not completed, FreePBX continues to drop down the available trunks list and repeats the process until a call is either completed or the trunk list is exhausted. The one exception to this scenario was support for ENUM. In that situation, a lookup occurs after a call is dialed to see if it can be placed as a free SIP call. We'd like to do the same thing with Gizmo5's Backdoor Dialing. What we want to do is query the Gizmo5 database to determine whether a number to be called is a free call. If it is, then we want to modify the route for processing the outbound call to take advantage of Gizmo5's free calling option. And we'll also need to change the phone number by adding a 0101 prefix.

Since our last column, another serious limitation in FreePBX post-call processing was mentioned on the PBX in a Flash Forums. With a number of commercial PBXs, it's possible to specify post-dial processing for emergency calls. For example, in an office environment, if an employee dialed 911, it would be helpful to alert a receptionist in some way so that immediate first aid could be attempted and also to give the receptionist a heads up so that he or she could direct emergency responders to the appropriate location in a building. As written, FreePBX doesn't provide an easy way to handle this.

So our objective today is to provide a couple applications which address these limitations. And the apps also will document a methodology for overcoming other post-dial processing issues which may arise using the existing FreePBX framework.

The trick to adding today's hooks into the Asterisk dialplan is to understand that Asterisk loads identically named dialplan contexts only once. Taking advantage of this, FreePBX provides a mechanism for users to insert custom code to replace default FreePBX contexts. All of these configuration files are stored in /etc/asterisk. For today, the context we want to modify is [macro-dialout-trunk]. This is the FreePBX macro that does the heavy lifting once a call has been placed and a trunk route has been selected to handle the call. With FreePBX 2.3, the macro is in extensions.conf. In FreePBX 2.4 and 2.5, the context is in extensions_additional.conf. In both cases, what we want to do is copy the entire contents of the existing context into the bottom of extensions_override_freepbx.conf. If you're using an editor to cut-and-paste the code, be sure you get the code that is located outside the left and right margins of your editor. And the context ends on the line before the next context begins. In the case of FreePBX 2.3, the next context is [macro-agent-add]. In the case of FreePBX 2.4, the next context is [macro-dialout-dundi]. And, in 2.4, there is now a comment which indicates where each context ends: ; end of [macro-dialout-trunk].

What we want to do is insert a line or two of custom code in this context which you've copied into extensions_override_freepbx.conf. The purpose is to run our custom code after the number to dial and trunk ID have been passed to this macro. Then, in the case of the Gizmo5 application, we'll run out to the Internet to determine if this call should be handled as a free call. If so, we'll change the trunk ID number to match your Gizmo5 trunk, and we'll change the number to dial by prefixing the existing number with 0101. The only gotcha with the Gizmo5 Backdoor Dialing is that every number must be tested at least once by someone (not necessarily you) in order to populate the Gizmo5 free calling database. You can check as many numbers as you like at this link. In the case of our 911 emergency application, we'll check to see if the number being dialed is 911. If so, we'll send an email or text message to an address that you define with an alert that extension 1234 just placed a call for emergency assistance to 911.

If you're using FreePBX 2.3, the custom code below should be inserted after the third "exten" line in the context, i.e. after the following line of code:

exten => s,n,Set(ROUTE_PASSWD=${ARG3})

If you're using FreePBX 2.4, the custom code below should be inserted after the first "exten" line, i.e. after the following code:

exten => s,1,Set(DIAL_TRUNK=${ARG1})

And the code to be inserted looks like this for Asterisk 1.4:

exten => s,n,AGI(nv-outbound.php|${ARG2}|${ARG1})
exten => s,n,AGI(nv-gizmo.php|${ARG2}|${ARG1})

For Asterisk 1.6, it should look like this:

exten => s,n,AGI(nv-outbound.php,${ARG2},${ARG1})
exten => s,n,AGI(nv-gizmo.php,${ARG2},${ARG1})

Now we need to add a couple of PHP scripts to your system and set a few configuration options, and you'll be ready to go. While logged into your server as root, issue the following commands:

cd /var/lib/asterisk/agi-bin
wget http://pbxinaflash.net/source/gizmo/nv-gizmo.zip
unzip nv-gizmo.zip
rm nv-gizmo.zip
wget http://pbxinaflash.net/source/gizmo/nv-outbound.zip
unzip nv-outbound.zip
rm nv-outbound.zip
chown asterisk:asterisk *.php
chmod +x *.php
asterisk -rx "dialplan reload"
grep OUT_ /etc/asterisk/extensions_add* | awk '/ = / { print $0 }'

The last line of code above is used to decipher the trunk numbers associated with each of your trunks. What we need to know is the trunk number for the Gizmo5 trunk that you set up in last week's tutorial. Write it down and then edit nv-gizmo.php: nano -w nv-gizmo.php. Look down the screen about 5 or 6 lines for the line that reads $GIZMO_TRUNK = "21" ; and replace 21 with the number you wrote down for your actual Gizmo5 trunk. In the next two lines, insert your actual Gizmo5 username and password between the quotation marks. Don't change anything else. Save your changes: Ctrl-X, Y, and then press the Enter key.

With the other application, nv-outbound.php, we need to be sure it's working with your phone system before you actually need it. And we don't place test calls to 911. So here's the drill. Edit the file: nano -w nv-outbound.php and insert your email address or text message address in the $email variable between the quotes. Then move to the next line and insert a telephone number with the area code that you can dial from a phone on your system to test that the notification is working. For example, put in your cell phone number. Once you save your changes, pick up a phone on your system and call your cellphone. You should receive an email notification within a few seconds. Once it's working, edit the application again and change the $number2monitor to "911" and you're all set. Enjoy!

VPN in a Flash Update! We've had over 100 reservations for our new VPN in a Flash system. We're very close to having a manufacturer in place so hopefully we'll have more good news in a week or two. We have begun the documentation for the new product, and we encourage you to take a look and offer any questions or comments you may have on our forums. The documentation is in the new Google Knol format and can be reviewed here. It's not too late to get in the queue and place a reservation for a system. Just send us a note, and we'll keep you posted as the release date approaches. It'll hold your place in line with absolutely no obligation to purchase.

Coming Attractions. We're very close to signing on a new VoIP provider for PBX in a Flash users that will provide penny-a-minute calls in the U.S. and Canada as well as all-you-can-eat plans for just over $10 a month with an annual contract. We're also only a week or two away from a new version of AsteriDex with Outlook synchronization and a TTS dialer for AsteriDex queries from any connected Asterisk phone. Stay tuned!

Hosting Provider Deal of the Century. Just an FYI that the Nerd Vittles hosting provider, BlueHost, has raised the bar again on hosting services. For $6.95 a month, you can host unlimited domains with unlimited web hosting disk storage and unlimited monthly bandwidth. Free domain registration is included for as long as you have an account. It really doesn't get any better than that. And their hosting services are flawless! Just use our link. You get a terrific hosting service, and we get a little lunch money.

New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. Until October 15, you can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up 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. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.

Some Recent Nerd Vittles Articles of Interest...

Posted in Technology, Telephony by , Wednesday, September 3, 2008 3:00 am Comments (5)

Tags: , , ,

Free Asterisk Calls to Zillions of Phones with ENUM and Gizmo5′s Backdoor Dialing

It’s been a while since there’s been much to cheer about in the free calls department with Asterisk®. But today, to kick off the new school year, we have lots of good news and some simple tricks to add zillions of free phone numbers to your Asterisk repertoire. In fact, you’ll be able to call almost any non-AT&T cellphone or landline in the United States at no cost. Remember that when you buy your next cellphone! Special thanks to Cliff on the PBX in a Flash Forums for heads up.

Some early readers of Nerd Vittles may remember sipphone.com which morphed into Gizmo5.com. In January of this year, Gizmo5 struck peering deals with a number of telephone providers that already routed their calls over the Internet. And it’s a pretty impressive list that includes more than 10% of the phones and cellphones in the United States according to Gizmo5′s bean counters. There’s Access One, Airadigm, Allegiance, Alltel, Cablevision Lightpath, Cat Communications, Cbeyond, Cellcom, Cellular Properties, Centennial Wireless, Choice One, Cincinnati Bell Wireless, Cinergy Communications, Cingular, CityNet, Cleveland Unlimited, Comcast Digital Voice, Commpartners, Conversent Communications, Cox Communications, CP Telecom, CTC Communications, Dobson Cell, Eureka, Globalcom, Heartland Communications, Illinois Valley, ITC Deltacom, LDMI, McLeod, Metro PCS, Mpower, Nationsline, Nextel, Nextera Communications, Paetec, RCN, Sprint PCS, Talk America, Telnet Worldwide, T-Mobile, US Cellular, Verizon Wireless, and XO. Whew! And the program is constantly being expanded. Toll-free numbers and Gizmo5-to-Gizmo5 calls also are free using Gizmo5. You can check whether your frequently called numbers are free calls by simply entering the phone numbers at this link.

Thus was born what Gizmo5 calls Backdoor Dialing. Just dial 0101 and the 10-digit number of your choice. If it’s free, the call goes through. If not, you get a message that the number is not yet supported and click. The beauty of the program is that your total investment to use the free service with Asterisk is a one-time fee of $10 for a bucket of CallOut minutes to activate your account. Sometimes this takes a day for the credit to appear, particularly if you use PayPal to cover the cost. The good news is you can spend most of the $10 making calls to any phone in the world, many for under 2¢ per minute, using just about any computer on the planet. Just leave a few cents in the pot to keep your free Backdoor Dialing service enabled. From our testing, we’d rate the Gizmo5 call quality as excellent on both the free and the pay-per-minute calls! Complete rate tables are available here.

Gizmo5 provides free softphones for Windows, Macs, and Linux as well as numerous cell phones and mobile devices including Treo, Nokia, and many more (not the iPhone… yet!). All of the softphones make it extremely easy to place SIP calls, e.g. joeschmo@mypbx.dyndns.org. And you can place these calls all day long at no cost. See our tutorial for step-by-step instructions on setting up your own SIP addresses on your Asterisk server. The softphones also include Conferencing, SMS, and Instant Messaging with AIM, Yahoo, MSN, Google, and MySpace.

As with many of these services, they weren’t designed for Asterisk, but nothing in their fine print precludes Asterisk use so today we’ll show you how. Will the program last forever? Who knows, but it’s free for now. And the cost of admission is too good to resist. You’re obviously not going to dial every number you frequently call twice just to see if the call is free. That’s why you’ll want to use a robodialer such as AsteriDex for your outbound calling. Then it’s easy to adjust the phone numbers of your friends with Sprint, T-Mobile, or Verizon cellphones so that you never have to pay for those calls again. Just add a prefix of 0101 to the numbers, and you’re done. And they can call you on your Gizmo5 CallIn number through Asterisk if you’ve enabled the CallIn Service and chosen a number. It’s under $3 a month with an annual subscription. Or the calls can be returned using the CallerID number displayed by Gizmo5 when you call your friends. Toll charges may apply in this case due to the Gizmo5 area code.

So let’s get started. Step 1 is to download and install a free softphone of your choice and follow the prompts to sign up for your account. There’s really no reason not to install a Gizmo5 softphone on every computer you own. If you don’t use it, there’s no cost. If you ever need it, it’ll be there for you. Step 2 is to make a $10 purchase of CallOut minutes. While you’re waiting on the credit to appear (and it usually takes less than a day), let’s set up Asterisk. You’ll need your new account name, password, and phone number from Gizmo5 to get started.

Setting Up a FreePBX Trunk for Gizmo5. If you’re using a product such as PBX in a Flash that includes FreePBX, then open FreePBX in your browser and choose Setup->Trunks->Add SIP Trunk. Leave the General Settings blank. For the Dialing Rules, if you just want free calling through your Gizmo5 trunk, plug in values below. For regular calls as well, add 1NXXNXXXXXX or an entry that is suitable for each country you wish to call.

1800NXXXXXX
1822NXXXXXX
1833NXXXXXX
1844NXXXXXX
1855NXXXXXX
1866NXXXXXX
1877NXXXXXX
1888NXXXXXX
800NXXXXXX
822NXXXXXX
833NXXXXXX
844NXXXXXX
855NXXXXXX
866NXXXXXX
877NXXXXXX
888NXXXXXX
0101+NXXNXXXXXX
0101NXXNXXXXXX

Name the Trunk: Gizmo5. Make the following entries in Outgoing Settings Peer Details:

disallow=all
allow=ulaw
auth=md5
authuser=youracctnameNOTyourphonenumber
canreinvite=no
context=from-trunk
dtmfmode=auto
fromdomain=proxy01.sipphone.com
fromuser=youracctnameNOTyourphonenumber
host=proxy01.sipphone.com
insecure=very
nat=yes
qualify=yes
secret=yourpassword
type=peer
username=youracctnameNOTyourphonenumber

Clear out the Incoming Settings and use the following syntax for the Registration String. Then Save your setup and Reload Your Dialplan. NOTE: Don’t use any registration string unless you want incoming call support. By not registering, you can use your softphones whenever you need it to also make outbound calls. If you register with Gizmo5 using a registration string, then it knocks out use of a softphone since you can’t have two simultaneous registrations to the same account. But registering allows those you call with this service to call you back conveniently… although not necessarily for free from the caller’s phone.

youracctname:yourpassword@proxy01.sipphone.com/yourphonenumber

Setting Up a FreePBX Outbound Route for Gizmo5. While still in FreePBX, choose Setup->Outbound Routes->Add Route. Name the route: OutGizmo5. Then enter the following Dial Pattern: 0101NXXNXXXXXX. Choose SIP/Gizmo5 as your Trunk Sequence. Then click Submit Changes and Reload Your Dialplan.

Setting Up a FreePBX Inbound Route for Gizmo5. While still in FreePBX, choose Setup->Inbound Routes->Add Incoming Route. Name the route: Gizmo5 and plug in your 10-digit DID number in the appropriate field. Then Set a Destination for the incoming calls. That’s it. Save your entries by clicking the Submit button and then Reload Your Dialplan.

Making a Free Call with Gizmo5. Once your DialOut credit appears on your softphone or in your Gizmo5 web account, you’re ready to start making calls. From any phone connected to your Asterisk server, just dial 0101 plus the 10-digit phone number. On the Asterisk CLI, you should see the call routed out through your SIP/Gizmo5 trunk. If you get a congestion tone and you’re sure your DialOut credit has been posted to your account, then check your username and password entries in your Trunk setup. Be sure to use your account name and NOT your Gizmo5 phone number for your username, authuser, and fromuser entries. But, if that doesn’t work, try using your Gizmo5 phone number instead of your assigned user name. Some have reported quirks in which actually works. For us, the assigned user name did the trick. Also make certain that the disallow all entry is above the allow=ulaw in versions of FreePBX after 2.3, or no calls will ever be successful.

Photo courtesy of the Chicago Historical Society and the Library of Congress American Memory ProjectTurning Non-Free Numbers into Freebies. There’s always some enterprising individual that figures out a quick way to beat the system even when many calls already are free. Suppose the number you wish to call isn’t yet available through Backdoor Dialing. The only trick is to have a pool of numbers from a provider with a peering arrangement with Gizmo5… and, of course, an Asterisk or FreeSwitch server to forward the calls and handle the number translation. You can read about RingBranch’s implementation, and then you can sign up for the service here.

There’s another way to turn non-free calls into freebies. This is Gizmo5′s "All Calls Free" Plan which is available in 60 countries. Landlines and mobile phones are supported in 17 countries while landlines only are supported in 43 more. U.S., Canadian, and Chinese landlines and cellphones are included in the program in addition to those of the Pope and the other residents of Vatican City. God works in mysterious ways! Here’s the complete list of countries that are supported.

To qualify a landline or mobile number for free calling (by dialing with the usual country code prefixes), you both have to be "active" Gizmo5 subscribers, your landline and mobile numbers must be listed on your account, and you must enter each other in your respective Buddy Lists. Then free calls using your Asterisk Gizmo trunk can be made to the "regular" phone numbers of all your pals whether the called person is online with Gizmo or not. Be aware that you can’t call your own numbers for free, and there is lots of additional "fine print" in this program. Nothing precludes your spouse having his or her own Gizmo5 account, however. You’ll need to wade through the rules carefully to take advantage of the free calling. It is possible, but it’s not easy. If you have relatives in Europe, Australia, or the Far East, you might want to have a look here. Just do a search for "All Calls Free." Your Gizmo5 softphone also will report your current All Calls Free Status.

Add Free Calls to 40 Million Asterisk Servers with e164.org. While we’re on a roll of free calling, here’s a simple way to add free calling to 40 million Asterisk servers around the world. Just add your name and phone numbers to the e164.org registry at no cost and configure FreePBX with ENUM support. Then outbound calls to numbers in the e164 registry will always be free as well. The whole setup takes less than 10 minutes. Here’s how.

The first step in setting up ENUM is to create a SIP address for your Asterisk server. The format looks like this: myname@somedomain.com. You’ll need either a fully-qualified domain name (FQDN) if your server has a static IP address or an FQDN issued through a dynamic DNS service such as dyndns.org if you have a dynamic IP address, e.g. pbx.dyndns.org. In the latter case, your router keeps dyndns.org apprised of changes in your external IP address so that pbx.dyndns.org always resolves to the correct IP address of your Asterisk server. Incidentally, with any hosted domain using a registrar such as omnis.com, it’s easy to add a subdomain DNS entry and point it to your Asterisk server, e.g. sip.joeschmo.com. That won’t cost you a dime other than the annual $6.95 domain registration fee which you’re already paying anyway.

Step two is to add your new FQDN address with a name of your choice to your Asterisk server. Then Asterisk will know how to process incoming SIP calls to that address. Read the Rolling Your Own section of our article on SIP Proxies for the procedure using FreePBX. It only takes a minute or two to set up. Let’s assume for purposes of this tutorial that you’re going to use the following destination address on e164.org for your server: e164@pbx.dyndns.org. An advantage to this type naming scheme is you can always keep straight the source of your incoming SIP calls. Thus your /etc/asterisk/extensions_override_freepbx.conf file should include a line in the [from-sip-external] context that looks like this: exten => e164,1,Goto(from-trunk,e164,1)

This tells Asterisk to route incoming SIP calls to e164@pbx.dyndns.org to the FreePBX Incoming Route for e164. And to complete the routing of the inbound calls to this address, add an Inbound Route in FreePBX called e164 that includes a destination of your choice for these SIP calls, e.g. an extension, a ring group, or an IVR already configured on your system. Just a footnote that e164.org requires you to enter a confirmation PIN when you set up the SIP routing to your server. So, at least initially, make the destination for your e164 SIP calls an extension that you can answer to obtain your PIN. You can safely ignore the FreePBX warning that you’re entering an odd type of inbound route by clicking OK. But you knew that.

Now let’s get you signed up with an account on e164.org. Go to the web site and click the Sign Up tab. Go through the sign up drill and then log into your new account. Then click the Phone Numbers tab and Add your phone numbers to e164. For each number, enter the area code and number. Then click the Next button. You’ll be warned about not having the number you’ve specified redirected to an IVR. If you already have this DID redirected to an IVR, change the routing temporarily to an extension that you can answer to obtain your PIN before you press Next to proceed. You’ll then be prompted for the SIP address to contact your server. Leave the default SIP protocol and plug in the address you created, e.g. e164@pbx.dyndns.org (using your own FQDN, of course). As soon as you click the Next button, your phone should start to ring, but there may not be a message when you answer. Hang up and wait for the second call within 15 minutes. It will include your PIN. Now click on the Phone Numbers tab and update your phone entry by choosing Enter PIN and typing your assigned PIN. Your phone number now has been activated with the e164 service. To complete the setup, you’ll want to click on the Do Not Call option and make your selections. You also can decide whether to list yourself in the ENUM White Pages directory.

Remember that the real purpose of this drill was to avoid charges when you place outbound calls to numbers in the ENUM directory. We merely added your numbers to e164.org so that others could benefit as well. So the final step before you can start saving money is to configure FreePBX to handle ENUM lookups for outbound calls from your server. One more observation may be helpful. You’ll recall that one of the limitations of FreePBX has always been that once an outbound route was chosen for a call, if the call was completed using the first destination trunk in that route, then the call processing ended there. ENUM adds a new wrinkle because we basically want to connect to ENUM to check for a free route and, if no matching entry is found, then we want the next trunk to process the call. As luck would have it, FreePBX has been tweaked to allow this scenario. All you have to do is create an ENUM trunk and then place it first in your sequence of trunks for each of your outbound routes. If an ENUM entry is found for the number you’re calling, the call will be routed as a free call with a direct SIP connection. Otherwise, the call processing will continue and the call will be routed using the next trunk specified in your outbound route.

There are two steps in FreePBX to implement ENUM. First, we need to create a special ENUM trunk. And second, we need to adjust our outbound routes to use the ENUM trunk first, and then the series of trunks you already have specified in each outbound route. NOTE: You obviously wouldn’t do this for an emergency 911 outbound route.

In FreePBX, click Setup, Trunk, Add ENUM Trunk. Enter your desired CallerID for these calls. Set a maximum number of channels, if desired, and then leave the other entries blank in most cases. Save your settings and reload your dialplan. Now click Setup, Outbound Routes and adjust the sequence of trunks for each of your existing routes. Be sure to put ENUM in the top position of each desired route. We also recommend adding a new Free Calls route so that users on your system can dial 0 and then a number to place a call through ENUM and then Gizmo5. If neither has a route for calling the party for free, the call will fail. The dial patterns might look like this for U.S. calls:

0|1NXXNXXXXXX
0|NXXNXXXXXX

The trunk list would look like this:

0 ENUM
1 SIP/gizmo5

Continue reading Part II.

Today’s Must Read: 101 Things You Can Do With Asterisk

VPN in a Flash Update! We’ve had over 100 reservations for our new VPN in a Flash system since last week. We’re very close to having a manufacturer in place so hopefully we’ll have more good news in a week or two. We have begun the documentation for the new product, and we encourage you to take a look and offer any questions or comments you may have on our forums. The documentation is in the new Google Knol format and can be reviewed here. It’s not too late to get in the queue and place a reservation for a system. Just send us a note, and we’ll keep you posted as the release date approaches. It’ll hold your place in line with absolutely no obligation to purchase.

Coming Attractions. We’re very close to signing on a new VoIP provider for PBX in a Flash users that will provide penny-a-minute calls in the U.S. and Canada as well as all-you-can-eat plans for just over $10 a month with an annual contract. We’re also only a week or two away from a new version of AsteriDex with Outlook synchronization and a TTS dialer for AsteriDex queries from any connected Asterisk phone. Stay tuned!

Hosting Provider Deal of the Century. Just an FYI that the Nerd Vittles hosting provider, BlueHost, has raised the bar again on hosting services. For $6.95 a month, you can host unlimited domains with unlimited web hosting disk storage and unlimited monthly bandwidth. Free domain registration is included for as long as you have an account. It really doesn’t get any better than that. And their hosting services are flawless! Just use our link. You get a terrific hosting service, and we get a little lunch money.

New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. Until October 15, you can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up 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. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.

Some Recent Nerd Vittles Articles of Interest…

Posted in Technology, Telephony by , Monday, August 25, 2008 2:00 am Comments (9)

Tags: , , , , ,

Ringbinder theme by Themocracy