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The Most Versatile VoIP Provider: FREE PORTING
JUST RELEASED: Visit the Incredible PBX WikiGoogle Voice: Is the SIP and Asterisk Honeymoon Over?
"Well. That was quick." Not encouraging words to hear from your new best friend. Google doesn’t make many mistakes so let’s give their decision to shut down SIP connectivity to Google Voice a little more time to percolate before concluding that they’ve thrown the baby out with the bathwater. The knee-jerk reaction is simply to write off Google as having about as much technical and business savvy in the VoIP market as AOL demonstrated… twice. But that’s not the Google many of us have known and done business with. And it’s the antithesis of everything Google Android and the company have sought to promote.
Update: The original SIP interface to Google Voice described in this posting no longer works. A new approach that really works is now available on Nerd Vittles at this link.
For the record, let’s back up a minute and review what transpired. Last Monday we (and others) released a tutorial showing users how to almost transparently connect Google Voice to Asterisk® PBXs as either a SIP extension or a trunk. The beauty of this was that it added a great new, low-cost telephony provider to the worldwide mix. The short-term advantage to Asterisk users was that calls within the U.S. currently were free although Google already has announced that those darn "accountants" have told them that they’re going to be forced to charge for the service one day soon. Cough cough!
In the process of testing this SIP connectivity, what we discovered was the only layer of protection standing between your wallet and free worldwide phone calls for every creep on the planet was a 4-digit PIN. That translates into 10,000 SIP calls to break into any user’s account. Even without the assistance of BOTs, that afforded your shiny new Google Voice account less than an hour of protection with a well-written SIP dialer and no added protection from Google Voice. By Friday, Google had closed the hole and blocked all SIP connectivity except for Gizmo.
The simple solution to open up safe SIP connectivity to Google Voice would be the addition of either an IP address field or a SIP URI in the Google Voice configuration options. SIP calls to and from that address would be allowed. All other calls would be blocked.
And why is this a good idea? First, it promotes the SIP open source standard. See Andy Abramson’s blog for a thought-provoking analysis of where this could ultimately lead. Second, it brings Google Voice connectivity to an enormous pool of users most of whom are tech-savvy and influential in the VoIP marketplace. Millions of Asterisk systems already have been deployed worldwide. Third, it’s the right business decision. Can you spell S-K-Y-P-E? At a time when Skype is opening up its network to SIP connectivity through Skype for SIP and Skype for Asterisk not to mention corded and cordless telephones, what possible business case could be made for introduction of a closed-platform VoIP service with no outside connectivity except through MaBell landlines? Hello!
This may come as a shocker to the Google accountants, but the call pricing and the double-hoop outbound dialing through Click2Dial aren’t that great. Comparable SIP call pricing is available from thousands of providers worldwide. And voice transcription through the Click2Dial voicemail service is downright horrendous. We proved that quickly with our Google Voice demo system.
It comes down to this. The one truly distinguishing factor with Google Voice is Google. At a time when Google has been at the forefront of open source telephony in the cellphone space with Android, the current Google Voice design is a giant step backwards. Rumor has it that Ma Bell had an offering that rang phones in multiple locations about 70 years ago. It was called a Party Line. How are they doing with that? We hope Google does the right thing and opens its new service to safe SIP connectivity. It’s the right and the bright thing to do.
The Honeymoon Ain’t Over… The Return of Googlified Messaging With Free U.S. Calling
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
A Baker’s Dozen Asterisk Nuggets from the Forums
Whether you’re new to the Open Source VoIP Community or an old-timer, we wonder how many folks actually miss many of the terrific Asterisk® applications that are hidden in message threads on the various Asterisk forums around the globe. In honor of St. Patty’s Day, today we want to take a stroll through the PBX in a Flash forum just to demonstrate what you may be missing by not visiting the forums or subscribing to some of the better syndication feeds.
Skype Gateway to Asterisk. If you read our recent column on integrating a Skype gateway into your Asterisk server, good for you. But, if you missed the forum dialog which followed release of the article, you missed all sorts of enhancements and system integration tips which made the Skype gateway a much better fit on many systems.
CallerID Superfecta. One of the most perplexing issues facing those that implement VoIP telephony solutions is wrestling with CallerID issues which flow from the ongoing Baby Bell phonebook monopoly. Many of you may have tried our CallerID Superfecta application which provides CallerID lookups for FreePBX-based systems using AsteriDex, Google Phonebook, AnyWho, and WhitePages. But, if you’d explored the forum additions to CallerID Superfecta, you would have uncovered an incredibly slick FreePBX installer as well as support for WhoCalled.us and Telcodata plus SugarCRM as well as numerous fixes for syntax changes on the various lookup sources.
Faxing with Asterisk. Other than CalleriD, there’s probably no issue that generates more consternation in the Asterisk community than fax integration. We reintroduced nvfax for Asterisk 1.4 recently. But, if you’d been following the forums, you’d also know that HylaFax and AvantFax now can be easily integrated into PBX in a Flash thanks to the work of Joe Roper and Tony Shiffer.
A2Billing for Asterisk. Another application that’s been difficult to get working with Asterisk has been A2Billing, a sophisticated calling card and PBX billing system. There really never has been a clear, concise cookbook for getting the software installed and properly configured. Once again, thanks to Joe and Tony, this forum thread provides a step-by-step tutorial for getting every facet of A2Billing installed and properly configured.
Asterisk Stickies. This is another promising Asterisk web application for PBX in a Flash that pops up stickies when incoming calls are received. You then can add the contact to your phonebook and also generate the XML code to update the phone directory on Grandstream and Cisco phone sets. It also supports click-to-dial from the web interface. You can keep up with the progress of this developing application in this very active message thread.
Text-to-Speech FreePBX Module. Just today a new TTS module for FreePBX was introduced which lets you generate TTS announcements for use with any FreePBX-based Asterisk system.
Overhead Paging with Asterisk 1.4. Most workplaces need some sort of overhead paging system. With the tips in this thread and any Asterisk 1.4 server, it’s incredibly easy to implement.
Streaming Music on Hold. We introduced streaming audio for Asterisk over three years ago in the Asterisk 1.2 days. A new message thread has updated that technology to support Internet radio using any Asterisk 1.4 server.
Email Alerts on Trunk Failures. For those that rely upon Asterisk systems to do real work, it’s essential to know when access to your carrier has failed so that you can make adjustments to your outbound and inbound trunks. This thread provides a simple tutorial and script to get you started.
Outbound Emails with Asterisk and SendMail. Another one of our Top 5 most perplexing problems with Asterisk is getting an outbound email capability with SendMail working reliably. Part of this is the configuration hassles with SendMail. But service providers such as Comcast have made matters worse by blocking outbound access to port 25 on most non-business accounts. Here’s a message thread that will walk you through configuring SendMail to use Gmail as your outbound SMTP relay host, and you’ll never have an email problem again on your Asterisk server.
Voicemail Notification. Unified messaging may be everyone’s dream but the reality is that it would be nice to be called on your cellphone when a new voicemail arrived at your office. The Voicemail Notification System does just that. And this thread integrates the original design into a FreePBX module.
Configuration Editor for FreePBX. FreePBX stores much of its magic in Asterisk config files. At least in PBX in a Flash, we hide some of these files to protect the integrity of your system. In addition, changes made to some of these files will get overwritten the next time FreePBX is started since it populates a lot of the information in these config files from data stored in MySQL tables. For those that want to learn more about the FreePBX, there now is a configuration file editor which will let you view and edit any FreePBX config file on your system. You’ll find a complete tutorial in the forums.
Hotel-Style Wakeup Calls. A few weeks ago we covered Tony Shiffer’s new add-on module for FreePBX that provides hotel-style wakeup calls for Asterisk systems. This code actually had been available in the forum for several months and is yet another reason to frequently check the new message threads.
Mac OS X Scripting Package. Since publication, a new link to a Treasure Trove of Goodies for Mac OS X has been posted including a link to the new Mac OS X Scripting Package and Asterisk binaries for Mac OS X from Sven Slezak at Mezzo.
Syndication Syntax. Many forums provide a syndication feed link, but many do not. For vBulletin-based forums, the basic syntax for an RSS feed looks like this:
http://fqdn.com/forum/external.php
You can refine the type of feed you want by specifying the type: RSS, RSS2, ATOM, or XML. For example, to pull down a feed from the PBX in a Flash forum, here’s the syntax for the various formats that are supported:
http://pbxinaflash.com/forum/external.php?type=RSS
http://pbxinaflash.com/forum/external.php?type=RSS2
http://pbxinaflash.com/forum/external.php?type=ATOM
http://pbxinaflash.com/forum/external.php?type=XML
You can further refine the feed by narrowing it down to a particular forum of interest. For example, to retrieve the latest threads from the PBX in a Flash Open Discussion forum, the syntax looks like this:
http://pbxinaflash.com/forum/external.php?type=RSS2&forumids=2
Finally, here’s the list of forum ID numbers for the PBX in a Flash forum:
2 – Open Discussion
3 – Help
4 – Endpoints
5 – Trunks
6 – Providers
7 – Wish List
9 – Bug Reporting & Fixes
10 – Add-On Install Instructions
Something We Missed? There are hundreds of additional Asterisk apps hiding in the woodwork. Please share your discoveries by posting a comment and link below. Enjoy!
Want a Bootable PBX in a Flash Drive? Our Atomic Flash bootable USB flash installer for PBX in a Flash has been quite the hit. Special thanks to all of our generous contributors! Atomic Flash provides all of the goodies in the VPN in a Flash system featured last month on Nerd Vittles. You can build a complete turnkey system using almost any current generation PC with a SATA drive and this USB flash installer in less than 15 minutes!
If you’d like to put your name in the hat for a chance to win a free one delivered to your door, just post a comment with your best PBX in a Flash story.1
Be sure to include your real email address which will not be posted. The winner will be chosen by drawing an email address out of a hat (the old fashioned way!) from all of the comments posted over the next several weeks.
And it still isn’t too late to make a contribution of $50 or more to the PBX in a Flash project and get a free Atomic Flash installer delivered to your door as our special thank you gift. See this Nerd Vittles article for details.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
- This offer does not extend to those in jurisdictions in which our offer or your participation may be regulated or prohibited by statute or regulation. [↩]
Why Wait? Build Your Own Skype Gateway to Asterisk
As the world awaits the much-hyped Digium® commercial offering of a Skype for Asterisk® gateway, we began exploring existing alternatives last week that provide individuals1 the same functionality at what will no doubt be considerably less cost. The Gizmo5 offering is darn-near perfect for those that need a quick solution and don't mind spending $20 a year to let someone else wrestle with the technical complexities that invariably accompany maintenance of cross-platform gateways such as this one.
NOTE: Article has been updated. For the latest edition, follow this link.
We, of course, like the technical challenge that comes with the territory so today we turn our attention to Greg Dorfuss' SipToSis software which forms the lynchpin of Gizmo's offering and which lets any Asterisk user create much the same gateway at no cost other than the expense of any Skype Out calls you may choose to make. When we're finished, you'll be able to call any Skype user in the world from any extension on your Asterisk server by entering either a Skype username or any 10-digit telephone number preceded by an 8 to take advantage of SkypeOut calling rates. You'll also be able to receive incoming calls from any Skype user on any extension of your Asterisk system. In short, what you get is a transparent interface to several hundred million Skype users from your Asterisk server. And the time to set this up is less than 15 minutes assuming you already have one of the Nerd Vittles Orgasmatron Builds in operation. These are available for Dell PowerEdge servers, Everex gPC desktop systems and minis, and most of the new Atom-based systems. If you have a garden-variety FreePBX-based Asterisk server such as PBX in a Flash, trixbox, or Elastix, just add another 5 minutes to reconfigure a few things.
Prerequisites. For today's project, we're assuming you have an existing FreePBX-based Asterisk server with either CentOS 5.2 or the Fedora 10 Remix featured in our Atomic Flash build. You'll need both a keyboard and mouse! For inbound Skype calling to work with other implementations including generic PBX in a Flash systems, you'll need to create a SIP URI for your Asterisk server: mothership@127.0.0.1. We've previously explained how to set one up in this article. The Atomic Flash installer, VPN in a Flash build, and the Orgasmatron II and III builds include this SIP URI functionality out of the box. You'll also need Java 1.5. To see if it's included in your distribution, issue the following command: rpm -q jdk. Finally, we're assuming you already have an existing Skype account. If not, download the Skype software for your Mac or Windows PC, and sign up. Try out a demo from any Skype phone. Just call nerdvittles.
Installing JAVA. If your particular Asterisk distribution doesn't have JAVA 1.5 or higher installed (rpm -q jdk), here's how to do it. Go to Sun's Java SE Development Kit 6u12 website, choose Linux for the platform, and agree to the license agreement. Then click Continue. Download jdk-6u12-linux-i586-rpm.bin and copy it to your Asterisk server. Make the file executable (chmod +x jdk-6u12-linux-i586-rpm.bin) and then run it. Scroll down the wordy license agreement AGAIN and type yes. Java 1.6 then will be installed on your system. Whew!
Basic Installation. Now we're ready to get started. Log into your Asterisk server as root and issue the following commands.
cd /root
mkdir skype
cd skype
wget http://www.skype.com/go/getskype-linux-centos
#Atomic Flash builds including VPN in a Flash
#should skip the next 3 yum commands.
#Then pick up again after the next comment line.
yum install xorg-x11-server-Xvfb
yum install qt4
yum install xterm
#everyone continues on from here
yum install libXScrnSaver.i386
wget http://pbxinaflash.net/source/skype/siptosis.tgz
rpm -ivh skype*
cd /
tar zxvf /root/skype/siptosis.tgz
FreePBX Design. The FreePBX setup that we recommend goes something like this. For outbound Skype calls, you have two choices.
1. To place a call to a regular phone number using SkypeOut (which costs you money), you'll simply dial 8 plus the area code and number. Our foreign friends will have to adjust their dialplans and /siptosis/SkypeOutDialingRules.props accordingly. Today's setup assumes 10-digit phone numbers!
2. To place a call to a Skype username using a softphone that supports SIP URI dialing such as X-Lite, you simply precede the Skype username with an asterisk, e.g. *echo123 will connect you to the Skype Call Testing Service.
For incoming Skype calls, the default setup routes those calls to a SIP URI: mothership@127.0.0.1. Whether you point this URI to an extension, ring group, or IVR is up to you. In the default Orgasmatron and Atomic Flash builds, the mothership URI is pointed to the Stealth AutoAttendant, an IVR that plays a welcoming message and then transfers the call to a ring group if no digit is pressed by the caller.
Configuring FreePBX. To put this setup in place, use a web browser to access FreePBX on your Asterisk server. You'll need to create a Custom Trunk and then an Outbound Route.
1. Choose Setup, Add Trunk, Add Custom Trunk. Fill in the form so that it looks like the following using your own CallerID number obviously:
When you're finished, click the Submit Changes button and then reload the dialplan when prompted.
2. Next choose Setup, Outbound Routes, Add Route. Fill in the form so that it looks like this:
When you're finished, click the Submit Changes button. Be sure to move this new OutSkype route to the top position in your Outbound Routes listing in the right margin! Then reload the dialplan when prompted.
3. If you're not using one of our custom Asterisk builds, add a new DayNight Control 1 option while you're still in FreePBX. Just specify where you want calls routed for Day mode and Night mode. Then, here's the easy way to activate SIP URI support on your Asterisk/FreePBX server. Copy the [from-sip-external] context from the extensions.conf file in /etc/asterisk. Now copy the content into extensions_override_freepbx.conf. Be sure to preserve the context name in brackets! Now make it look like the following. The additions we're making are shown in bold below:
[from-sip-external]
;give external sip users congestion and hangup
; Yes. This is _really_ meant to be _. - I know asterisk whinges about it, but
; I do know what I'm doing. This is correct.
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)
exten => s,1,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?from-trunk,${DID},1)
exten => mothership,1,Goto(app-daynight,1,1)
exten => pbxinaflash,1,Goto(app-daynight,1,1)
exten => e164,1,Goto(from-trunk,e164,1)
exten => fax,1,Goto(from-trunk,fax,1)
exten => s,n,Set(TIMEOUT(absolute)=15)
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,Playtones(congestion)
exten => s,n,Congestion(5)
exten => h,1,NoOp(Hangup)
exten => i,1,NoOp(Invalid)
exten => t,1,NoOp(Timeout)
Finally, reload your Asterisk dialplan, and we're finished with Asterisk and FreePBX setup:
asterisk -rx "dialplan reload"
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 Asterisk server since we have to run Skype in graphics mode. This may look a bit 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.
For those on a CentOS-based system, here are the steps:
1. Start up X-Windows: xinit
NOTE: 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
EndSectionSection "Device"
Identifier "Card0"
Driver "vesa"
EndSectionSection "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
2. Next we need to start up Skype, log in, set it to autologin whenever we start it, and then in the Skype configuration, set it to run minimized: skype
NOTE: Place a Skype Test Call (echo123) to be sure your audio settings are set correctly. If a 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! If you're using a Dell server such as the PowerEdge T100, you're probably S.O.L. without an advanced degree in Rocket Science.
HINT: Intel Atom-based motherboards are a piece o' cake!
3. Once you've got Skype working and all of the Skype settings configured above, shut down Skype.
4. Now start Skype again in background mode: skype &
5. Be sure to write down the PID for Skype in case you need to kill the job if something goes wrong. 🙂
6. You should have a command prompt back now. So issue these commands:
cd /siptosis
./SipToSis_linux
7. A message from Skype should pop up asking if you want to authorize external use of Skype: yes. Also be sure to mark the Checkbox to save this setting for future connections!
8. Now go to a softphone connected to an extension on your Asterisk server and dial *echo123
9. 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.
Finally, here are a few navigation tips for managing your Asterisk console on CentOS systems:
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 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 you're positive everything is working reliably and you've rebooted and tried it again just to make certain there are no prompts, here's how to fire up your Skype gateway whenever you reboot your server.
Log into your server as root and issue the following commands:
cd /root
wget http://pbxinaflash.net/source/skype/start-skype
chmod +x start-skype
echo "/root/start-skype" >> /etc/rc.d/rc.local
reboot
For those using one of our Fedora 10 builds such as VPN in a Flash or the Atomic Flash installer, these systems have a full implementation of X-Windows and KDE. Just start the system in mode5 (graphics mode), log in, run Skype in one window and start up SipToSis in a terminal window using the commands in Step 6 above. Authorize external use of Skype when prompted.
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. One final note of caution. Do NOT expose UDP port 5070 to the Internet unless you first secure this port with a username and password to avoid Internet intruders using your gateway as a free Skype dialing platform! You do not need 5070 exposed to the Internet to implement today's gateway solution for inbound or outbound Skype calling from your Asterisk server so we recommend you keep it securely behind at least one firewall.
Where To Go From Here. Well, those are the basics. You now can make one outbound Skype call at a time from your Asterisk server, and you can receive an inbound Skype call on any Asterisk extension when Skype users call your regular Skype name. If you want to start up your own business (like Gizmo5), then you'll need to do some tweaking. What you'll need is the STS Trunk Builder toolkit which is free, but proprietary. Enjoy!
Want a Bootable PBX in a Flash Drive? Our Atomic Flash bootable USB flash installer for PBX in a Flash has been quite the hit. Special thanks to all of our generous contributors! Atomic Flash provides all of the goodies in the VPN in a Flash system featured last month on Nerd Vittles. You can build a complete turnkey system using almost any current generation PC with a SATA drive and this USB flash installer in less than 15 minutes!
If you'd like to put your name in the hat for a chance to win a free one delivered to your door, just post a comment with your best PBX in a Flash story.2
Be sure to include your real email address which will not be posted. The winner will be chosen by drawing an email address out of a hat (the old fashioned way!) from all of the comments posted over the next several weeks.
And it still isn't too late to make a contribution of $50 or more to the PBX in a Flash project and get a free Atomic Flash installer delivered to your door as our special thank you gift. See this Nerd Vittles article for details.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest...
- Skype and this suggested implementation are intended for individual use. Your use is, of course, governed by the Skype Terms of Service. [↩]
- This offer does not extend to those in jurisdictions in which our offer or your participation may be regulated or prohibited by statute or regulation. [↩]
Now It’s a No-Brainer: Free Skype Calling with Asterisk
Many of you may recall that last August we showed you an easy way to implement free calling to millions of cellphones using Gizmo5 and Asterisk®. Vaporware aside, it's been a quiet six months in the Skype for Asterisk department. But now the folks at Gizmo5 have outdone themselves once again. This time it's their new OpenSky service. Here's how it works. You can call as many Skype users as you like and talk for up to five minutes for free.
If there are people you frequently call and the time limit is a problem for you, then you can cough up $20 a year and make as many 2-hour Skype calls as you like to your ten best friends. If you've got more friends than that or if you plan to use this for something other than a home Asterisk system, then there are reasonably priced plans to accommodate you. $320 a year gets you 20 accounts to an unlimited number of Skype users with the same 2-hour per call limit. $800 buys you 50 accounts, and $1600 buys you 100 accounts per year.
Getting Started. The easiest way to integrate this into your existing Asterisk system is to sign up for a free Gizmo5 account and then follow our previous tutorial to set up your outbound trunk.
Once you have everything working, you're ready to add a few numbers on your Asterisk system for your Skype pals. Here's the easy way, and we'll cover some more sophisticated implementations in a subsequent article. Assuming you have a friend with a Skype username of joeschmo, here's what you need to do to call Joe by dialing 563 (J-O-E) from any extension on your Asterisk system.
Edit the /etc/asterisk/extensions_custom.conf file on your system and add the following line within the [from-internal-custom] context:
exten => 563,1,Dial(SIP/skype_joeschmo@proxy01.sipphone.com)
If you also use softphones which support SIP URI dialing, then you might want to add another entry like this in the same context:
exten => joeschmo,1,Dial(SIP/skype_joeschmo@proxy01.sipphone.com)
Now just reload your Asterisk dialplan, and you're ready to start calling your Skype buddies around the world from any Asterisk extension.
asterisk -rx "dialplan reload"
The FreePBX Alternative. As has been pointed out in a comment, you can accomplish much the same thing using newer versions of FreePBX without having to muck around in extensions_custom.conf. Just add an Extension, choose the Custom type, provide an Extension Number, a Display Name, and optional SIP Alias. Then insert the following in the dial field, save your entries, and reload the dialplan when prompted.
skype_joeschmo@proxy01.sipphone.com
$20 Buys You Skype Calling Aliases. One of the major drawbacks of Skype always has been the alphanumeric Skype names which make it next to impossible to place Skype calls using regular telephones. Well, Gizmo5 has solved that, too. With your $20 annual subscription which gets you 2-hour Skype calls to your 10 best friends for a year, you now can define new phone numbers to match against your 10 favorite Skype friends. For example, for a user named John Doe, you might choose 564-6363 (JOHN-DOE). Once you sign up for the $20 Skype subscription and configure this alias in your Gizmo account, you can reach John Doe on Skype by dialing 1-333-564-6363 through your Gizmo5 trunk from any Asterisk extension. In your Asterisk setup, just create an outbound route for Gizmo calls with the following dial strings, and you're all set.
1333NXXXXXX
333NXXXXXX
Special thanks to JPE on the PBX in a Flash Forum for the original tip and to Adrian at Gizmo5 Operations for the alias demo. Enjoy!
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