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 the Gizmo5 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.
Turning 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
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 PBX in a Flash project gets 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…
- VPN in a Flash: The Real Future of Internet Telephony
- AWstats for Asterisk and PBX in a Flash
- Asterisk 101: Some CallerID Tips & Tricks
- Roll Tide: Let Allison and Asterisk Plan Your Next Surfin’ Safari… xTide for Asterisk (TTS)
- The Asterisk Orgasmatron: A $199 Turnkey PBX Install in Under 15 Minutes, Part I
- The Asterisk Orgasmatron: A $199 Turnkey PBX Install in Under 15 Minutes, Part II
- Asterisk Hell: A Minefield Navigation Guide for Newbies
- Introducing AIM Call Out for Asterisk
- Introducing PBX in a Flash 1.2: The Leaner, Meaner Asterisk Machine
- The World’s Best Asterisk Phone… The Aastra 57i
- The World’s Best Asterisk Phone: And Now It’s Plug-and-Play
- Introducing TeleYapper 4.0: The Free, Asterisk 1.4 Message Broadcasting System (TTS)
- SIP Proxies Make Asterisk Shine and Save You Money
- Introducing Telephone Reminders for Asterisk 1.4 with Phone and Web Scheduling (TTS)
- Text-to-Speech Bonanza with Cepstral and Asterisk 1.4 (TTS)
- Introducing the Stealth AutoAttendant for Asterisk 1.4 and FreePBX
- Asterisk PBX Management Done Right
- Allison’s Text-to-Speech Trifecta: Cepstral, Asterisk 1.4 or 1.6, and FreePBX 2.4 (TTS)
- Build a $199 Turnkey (Green!) Asterisk 1.4 System in Less Than An Hour
- Get Your Email By Telephone: Introducing MailCall for Asterisk 1.4 (TTS)
- Some Asterisk Resolutions for the New Year
Whether it’s a satellite medical office, or a remote construction site, or a regional branch of your favorite bank, or a temporary broadcast studio in Beijing, or a home office or hotel room of an architect, lawyer, or candlestick maker, the little black box on the left is poised to revolutionize the way we all do business… securely. All you need is an Internet connection. Plug it in, turn it on, wait for the LCD to display the machine’s IP address, make a quick change of the IP address on your cordless or desktop phones to match the display, and presto. Instant telephony… to anywhere, from anywhere in the world! Total setup time until you make your first call: about a minute. And the system supports 25 extensions and 10 simultaneous IP calls assuming you have available bandwidth.
It’s been several months since we last addressed the “
When we began the PBX in a Flash project last November, one of our primary goals was to provide a LAMP (Linux, Apache, MySQL, PHP) and Asterisk-based, open source telephony server which could be easily upgraded without starting all over each time a new release of CentOS, or Asterisk, or FreePBX hit the street. We think we got it about right with our three utility programs: update-scripts, update-fixes, and update-source. 
Design Goal. Our objective with this project was to make it simple to install awstats without screwing up anything that already worked. We also didn’t want to add much performance overhead to your server. For that reason, we’ve configured awstats so that you can access it with your web browser and manually update the data any time you like. But we haven’t included a cron job to do it for you each night. A typical data refresh with awstats takes about 10 seconds so this shouldn’t be a life-altering delay for even the most impatient folks out there. It also gives you the very latest stats on demand whenever you’re in the mood. If you look at the screenshot to the left, you’ll see an Update now link on the top line. All you have to do is click on that link to refresh your awstats data anytime you want it. It’ll also show you when you last updated the awstats information. And, for those that didn’t know, all of this data is pulled from the Apache log: /var/log/httpd/access_log. If yours is named something else, then you’re obviously not going to see much in awstats other than a few samples which were generated on our server before we bundled things up.
Vitelity Special: Time Is Running Out. Remember, you only have until July 15 to get your half-price DIDs and 60 free minutes from our special 
If you're relatively new to Internet Telephony and VoIP, then it may come as a bit of a surprise when CallerID for incoming calls shows the phone number for both the name and number of the incoming caller or when names that popped up using your plain old telephone's CallerID service no longer do. The problem is that most telephone providers only deliver a CallerID number when sending calls. Thus it is left to your service provider to look up the incoming number in a directory and supply the matching name. To put it another way, CallerID numbers are pushed to recipients, but CallerID names must be pulled from in-house databases. With Ma Bell and siblings, this was easy because they had all of the records. With some Internet Telephony Hosting Providers, it's a different story. The reverse is also problematic. Even though you may provide a CallerID name and number, most telephone companies throw the supplied CallerID name in the bit bucket and do their own lookups. So... if you're not in their directory, your number or nothing will be supplied instead of your actual name. Just another last vestige of monopoly preservation. With Asterisk, the simplest solution to this mess is to do your own lookups. And today we have an updated CallerID directory lookup service to assist: the 
We began our 15-minute adventure last week with a turnkey install of Asterisk onto a $199 Everex gPC2 using a fully-customized version of PBX in a Flash. If you haven't yet read the first article,