Ever wrestled with one of those thorny problems for weeks only to wake up in the middle of the night with the answer? Thus was born Travelin’ Man, a web- based, one-click Asterisk® application that automatically reconfigures your Asterisk PBX to enable remote SIP phone access from your cellphone, iPad, remote PC, NetBook, or desktop telephone.

News Flash: Be sure to read our latest article introducing Travelin’ Man 3, a completely new security methodology based upon FQDN Whitelists and DDNS. In a nutshell, you get set-it-and-forget-it convenience and rock-solid VoIP security for your Cloud-based PBX or any PBX in a Flash server that’s lacking a hardware-based firewall and you get both transparent connectivity and security for your mobile or remote workforce.

If you’ve read the Incredible PBX series of articles on Nerd Vittles, you already know what a thorny problem remote phone access is if you want to preserve the overall security of your server. Indeed, our recommendation has been to leave SIP access closed on your hardware-based firewall because of the dangers inherent in activating remote SIP access. Now we have a better idea!

Today’s new approach works like this. First, we’ll run a little script that secures all of your extensions with permit entries locking down all these connections to the IP address range within your private network. Then we’ll open the SIP and RTP ports on your hardware and software firewalls and map these ports to your Asterisk server’s private IP address. With this setup, no one can attempt remote SIP logins to your server because Asterisk blocks all SIP extension connection attempts except those originating inside your LAN. To manage external phone connections to your server, the install script creates a new virtual Apache web server on your Incredible PBX using port 83. We’ll enable and map TCP port 83 on your hardware and software firewalls to your server as well. Web access with port 83 is limited to running the Travelin’ Man app to activate external phones.

Now we’re ready to set up access to your server for remote devices. For each extension you wish to enable for remote access, we’ll create a special web directory using an obscure, random file name which will serve as the web link for the Travelin’ Man web app. For example, in the diagram above, directory 184778 manages extension 501, directory 2389957h manages extension 701, and directory 6993h5j manages extension 702. This is accomplished by simply changing the extension number in the index.php script stored in each directory.

When one of these web links is accessed remotely, the PHP script will automatically reconfigure Asterisk to enable access to the designated SIP extension on your server using the remote IP address from which the web page was accessed. And, of course, there’s an additional layer of SIP security as well. You still need your extension credentials to actually log in to your server with a softphone to place and receive calls. The Travelin’ Man installation process takes only a couple minutes, and the remote SIP activation procedure takes just a couple seconds each time you want remote access from a different location. Here’s a quick example of how it actually works.

Let’s assume we want to use the new $3.95 Bria SIP softphone on an iPad to connect as extension 501 on our Incredible PBX back at home. The problem is that the dynamic IP address of your iPad changes at each new site on your itinerary. Some locations have WiFi while others only have 3G connections.

First, we’ll generate an icon to run Travelin’ Man from your iPad desktop. Use the same procedure with an iPhone or iPod Touch, and there’s a similar procedure for Android devices.¹ You only have to do this once. Start up Safari on the iPad to access the new port 83 web server at the random web address the installer created to support extension 501. That web address is something like this using your own FQDN²: http://myserver.dyndns.org:83/184778. After establishing the link once, we’ll hit the + button in Safari and choose Add to Home Screen. This creates the TravelMan icon on the iPad. See the screenshot below of our demo iPad setup which used extension 221 instead of 501.

Once configured, it’s just two clicks to enable your remote phone anywhere: click once on the TravelMan icon. When your IP address is confirmed, return to your Home Screen and click the Bria softphone icon to establish a SIP connection back to your server. Behind the scenes, the Travelin’ Man application will generate the required permit entry for your remote IP address mapping it to the designated extension on your server, and then it will reload your SIP settings to make your Asterisk server accessible to the Bria softphone in your hotel room. The entire process takes only a couple seconds.

If your company happens to have a dozen traveling salesmen, then you’d simply assign a dedicated extension to each employee and create secure directory names for each person (e.g. 2389957h and 6993h5j in diagram above) with a copy of the Travelin’ Man app configured for that employee’s extension number. Now your entire mobile workforce has connectivity back to the home office from any location on the globe. And, when an employee leaves the company and another arrives, just create a new name for the old employee’s web directory to preserve the security of your system (e.g. 184778 in our example becomes 78hd773). Keep in mind that each time the Travelin’ Man app is run for any extension, it wipes out any previously authorized IP address entry for that extension. Thus, the security of your Incredible PBX is always preserved.

Prerequisites. Before proceeding with today’s install, you must be running a stock install of Incredible PBX with PBX in a Flash behind a properly-secured, hardware-based firewall³. We recommend the latest version of Asterisk 1.4 because it addresses a SIP vulnerability that might cause you problems if malformed SIP packets are targeted at your server. The current release of PBX in a Flash (1.7.5.5 Silver) is ideal, but any version of PBX in a Flash can be brought current with Asterisk using the update-source and update-fixes tools. Travelin’ Man assumes that you have the Incredible PBX base install of extensions: 501 plus 701-715. You can obviously add more or remove some, but you’ll need to manually adjust sip_custom_post.conf to reflect your actual extension list after the install completes.

The installer has been encrypted for your/our own protection. In source form, the script would allow anyone to defeat the Incredible PBX requirement. Doing so would mean the required IPtables security component would not be in place and properly configured to protect the underlying system from attack. So we’ve opted to play Big Brother to avoid potential security problems for all of us down the road. This article clearly explains all the necessary components if some folks want to roll their own version. We just don’t want the responsibility if something goes horribly wrong. As Forrest Gump would say, "Shit Happens." If you don’t believe it, check out the latest security scramble in the trixbox forums.

Installation. Now we’re ready to get started. So log into your Incredible PBX as root and issue the following commands:

cd /root
wget http://incrediblepbx.com/travelinman.tar.gz
tar zxvf travelinman.tar.gz
./travelinman.x

NOTE: If you’re using PIAF2 with CentOS 6.2, you’ll need to use the updated version of Travelin’ Man because of a syntax change in the Apache config file:

cd /root
wget http://incrediblepbx.com/travelinman2.tar.gz
tar zxvf travelinman2.tar.gz
./travelinman2

The first step in the install procedure is to lock down access to all of your extensions to your private LAN subnet. In case you ever want to do this on another server not running the Incredible PBX, here’s a link to our privip.sh shell script that shows how to do it. This should work on most FreePBX-based Asterisk systems.

Once the extensions are locked down, the script will modify your IPtables and Apache configurations to permit web access on port 83. Next, it will adjust your Asterisk setup to support the Travelin’ Man permit scheme. This involves reworking of sip_custom_post.conf so that permit settings for individual extensions can be stored in files named 501.inc, 701.inc, etc. Finally, the installation procedure will set up a single web site to support extension 501 with a randomized directory name for remote access.⁴ This setup will be stored in /var/www/travelman. To activate support for additional extensions, you would simply copy the subdirectory giving it a new random name: cp -r dir1 dir2. Then edit config.php in the new subdirectory and change the $extension entry.

To complete the install, you must reconfigure your hardware-based firewall and map the following ports to the private IP address of your server:

TCP 83
UDP 5060
UDP 10000-20000

When the installation is completed, it will show you how to access the new web site for extension 501 using either a fully-qualified domain name or a public or private IP address. Now just follow the steps at the beginning of this article to set up your Android or iDevice, and test things out. Enjoy!

Reminders: Be sure to review the comments to this article and the related support forum thread for a week or two for late-breaking enhancements and issues. Also, Incredible PBX comes preconfigured with call forwarding activated for extension 501. Don’t forget to either disable it or set up a real call forwarding number for extension 501 if you want your cellphone to ring. From any extension on your server, just dial *72501 to set up call forwarding. To cancel call forwarding and pass calls directly to the registered 501 softphone, dial *74 and enter 501. Also be aware that the default RingAll ring group (700) configuration on Incredible PBX systems does not include extension 501. So add 501 if you want your remote extension to ring for incoming calls.

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The Incredible PBX: Basic Installation Guide

Adding Skype to The Incredible PBX

Adding Incredible Backup… and Restore to The Incredible PBX

Adding Multiple Google Voice Trunks to The Incredible PBX

Adding Remotes, Preserving Security with Incredible PBX

Continue reading Basic Installation Guide, Part II.

Continue reading Basic Installation Guide, Part III.

Continue reading Basic Installation Guide, 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.

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

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

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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.
 

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Some Recent Nerd Vittles Articles of Interest…

1. To create a desktop icon for Travelin’ Man on Android devices, navigate to the link with your browser. Then save the link as a Bookmark by clicking the Star icon in your browser then click Add. Return to the Home Screen and, from the screen on which you wish to add the icon, touch and hold your finger on the screen. When the Add to Home Screen menu appears, choose Shortcuts then Bookmarks and select the link you previously saved. As with iDevices, you only have to do this once. [↩]
2. FQDN = Fully-qualified domain name [↩]
3. We recommend the dLink Router/Firewall. Low Cost: $35 WBR-2310  Best: DGL-4500 [↩]
4. If you’d like to download the web site code independently from the Travelin’ Man install procedure, here’s the link. [↩]

We interrupt our Incredible PBX coverage this week to bring you a summer roundup of the best and worst VoIP softphones for use with an iPhone, iPad, or iPod Touch in conjunction with Asterisk®. We’ve tested all of these products with Asterisk sitting behind a NAT-based firewall/router which introduces some additional wrinkles unless your softphone and server are connected through a virtual private network. We’ll leave the VPN discussion for another day. None of these products has native support for the iPad although all will work with any iPad as will any standard iPhone app in either 1X or 2X mode.

The three four products we’ll be evaluating are Acrobits SIP Softphone, the WiFone from Snizmo.com Ltd., the Media5-fone, and CounterPath’s just-released Bria softphone. All support SIP dialing, and the WiFone provides IAX connectivity as well. We were a bit surprised that, despite their reliance on SIP to connect calls, SIP URI support was minimal to non-existent in all but the Bria product. Before diving into the individual products, we should note that, in conjunction with our product evaluations, we received no compensation or discounted/free software from any source. We are a beta site for CounterPath’s next Bria release.

Acrobits Softphone. The Acrobits Softphone requires iPhone OS 3.0 or later and was recently updated on June 3, 2010. The softphone only supports SIP but works with both WiFi and 3G connections which makes it a perfect complement to current generation iPhones as well as the iPad-3G. The softphone also supports push notifications for inbound calls until multitasking is available with iOS 4.0. Multiple SIP accounts can be registered, and the softphone has SIP proxy, VPN, and STUN server support, a must with Asterisk sitting behind most NAT-based routers. G.711, GSM, and iLBC audio codecs are supported in the standard configuration, and we experienced excellent call quality using WiFi with no DTMF issues. As with all of these VoIP phones, 3G call quality was all over the map depending upon the reliability of your nearest cell tower. SIP URI’s can be called by cutting-and-pasting dial strings from entries in the Contacts list email address fields provided the SIP URI destination name is numeric. Quirky but it works. There’s also a speed dial feature for your 12 favorite contacts. Flexible dial strings are supported to smooth the path for international calling. With iOS 3.1, a bluetooth headset can also be used. The application sells for $7.99 in the App Store, and G.729 support can be added for an additional $9.99. G.729 is a must-have if you’ll be using a 3G network for most of your VoIP calls.

While call quality is obviously subjective, the Acrobits Softphone was our personal favorite for daily use. We routinely use it on an iPad to check Asterisk voicemails and to make outbound calls through our home Asterisk server while traveling. Setup is as simple as entering the IP address or FQDN¹ of your Asterisk server and an extension number and password to handle the calls. We added a public STUN server entry because of our NAT-based Asterisk setup.

Snizmo’s WiFone. A very close runner-up in voice quality was the WiFone from Snizmo.com Ltd. This softphone has the added advantage of supporting both SIP and IAX2 connections to Asterisk. If security and ease of use matter most to you, then you can’t go wrong with this softphone. IAX2 connections are much less vulnerable to attack from the Internet and are considerably easier to configure because of the elimination of thorny NAT issues. If we had found this softphone first, we probably would have looked no further. As you can see from the screenshot, this softphone supports multiple SIP and IAX connections and is easily set up using the configuration menu. For our European friends, it also supports SMS using a dozen different providers. Echo cancellation and STUN support are available, and G.711 and GSM codecs can be individually configured for SIP and IAX connections. An Outbound Proxy is also available as well as support for international dial strings and prefixes if you need it.

For SIP accounts, simply provide the server address, a username, and password. Authorization name, SIP port, and proxy server settings are optional. For IAX accounts, server address, username, and password are the only required entries. Each account can be toggled ON and OFF to meet your individual requirements. SMS Settings provides a listing of a dozen SMS providers. Simply add your username, password, and a CallerID and SMS just works. The contacts list also synchronizes with your Mac Address Book as well as MobileMe. The call quality of both SIP and IAX connections using WiFi was excellent. 3G support is not yet available. The web-based tutorial is excellent, and the application is available in the App Store for $6.99. An international version also is available.

We could not get the SIP URI functionality to work because the Contacts list phone numbers do not support SIP URI syntax, and there’s no way to manually enter or cut-and-paste a dial string from an email address in the Contacts list. While the polish of the application was not quite up to the Acrobits Softphone, the call quality was uniformly excellent with the SIP URI limitation that we’ve noted.

Media5-fone. Our final softphone in today’s roundup is Media5-fone from Media5 Corporation. It can be downloaded from the App Store for $4.99. While the application is exclusively a SIP phone, it does have preconfigured setups for dozens of providers in the event your requirements extend beyond the Asterisk universe. Unfortunately, there is no STUN support in the current version which makes it unsuitable for use with Asterisk implementations that sit behind NAT-based routers. Multiple SIP connections are supported as are second call, call waiting, and call toggle. In the current version, both SIP over WiFi and 3G are supported using iLBC, G.711, Enhanced G.711, G.722, and iSAC codecs. SIP Info, RFC 2833, and RTP Inband DTMF methods are configurable for each SIP account. Dialing prefixes are flexible and the phone has language support for English, Arabic, French, German, Italian and Spanish which facilitates international use. The phone also includes a nice implementation of visual voicemail; however, the SIP password and voicemail password would have to be the same to function properly with Asterisk. Automatic gain control and echo cancellation also are supported. With the addition of STUN and SIP URI support, Media5-fone would be a worthy competitor.

Update: CounterPath’s Bria. As luck would have it, CounterPath released their new Bria softphone for the iPhone today. It also is iPod Touch and iPad-compatible and supports both WiFi and 3G. The softphone is available at an introductory price of $3.99 in the App Store. It’s the best bargain in the softphone market. G.729 support can be added for an additional $8.99. G.722 wideband support reportedly is coming in August. You may recall CounterPath’s terrific and free X-Lite offerings for Windows, Macs, and Linux. They’ve been one of our favorite developers ever since, and we are actually serving as a beta tester for their next release. As usual, the Bria interface offers what is hands-down the best UI in the business. The voice quality of the calls is impeccable. Our only criticism is that out-of-the-box, Bria doesn’t work for placing outbound calls with Asterisk. Registration of credentials works fine, inbound calling works great, but outbound calls to either an extension, a phone number in the Address Book, or a SIP URI all just hang with no error message or notation in the log. Only after tracing down an obscure link on their web site did we discover the problem. It turns out that one simple change of a single default setting gets things working as they should. To make the change to support Asterisk, click Settings, Advanced Settings, Network Traversal Strategy, User Specified. Then change ICE:ON to ICE:OFF. Click the Advanced button, and then Apply Changes. Aside from this one default configuration glitch, the Bria softphone would be our Editor’s Choice. We highly recommend you make your purchase while the softphone still is available at the introductory price. For an excellent review, see Alec Saunder’s Blog today.

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

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

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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.
 

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Some Recent Nerd Vittles Articles of Interest…

1. FQDN = Fully-Qualified Domain Name [↩]

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.

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


Continue reading Part II.

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

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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!

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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.

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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.

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