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The Most Versatile VoIP Provider: FREE PORTING
JUST RELEASED: Visit the Incredible PBX WikiIntroducing VPN in a Flash: The $499 Mobile Telephony Appliance with Asterisk
We’ve spent a lot of time designing turnkey Asterisk®-based systems from the early Asterisk at Home days until the latest Orgasmatron Builds1 for PBX in a Flash. So, trust us! Nothing comes close to the new VPN in a Flash Mobile Telephony Appliance. Having endured more than a decade of preparations for national emergencies, we are well aware of the need for well-designed telephony systems which can be deployed on a moment’s notice anywhere. We also appreciate the need for a versatile, portable communications appliance which can be toted from hotel room to hotel room providing secure VoIP communications back to the mothership. And we fully grasp the need of thousands of businesses to transparently deploy remote communications devices at far away places but in a way that they still can be supported from home base. With all that in mind, Tom King and I have spent the last several months designing this VoIP telephony appliance. Now let us introduce you to the new world2 of VPN in a Flash.
Until six months ago, the hardware simply wasn’t available to provide the GUI performance necessary to create such a portable appliance. But the Intel Atom® processor changed all of that. And now Acer has stepped up with an almost perfect mobile implementation of the Atom motherboard in the Aspire One® Netbook. Weighing in at just over two pounds, it’s totally portable but also a powerhouse. And it’s quiet.
On the software side, the stars all lined up when Fedora® introduced Fedora 10 last week, an almost perfect rendition of the Linux® operating system with every imaginable bell and whistle including a low-overhead KDE® GUI that rivals the very best of Windows® and Mac OS X®. Our challenge was to put all the pieces together and add the very best of the Asterisk® telephony world to the mix. And, of course, we wanted to accomplish all of this while staying true to our open source roots. We think this Fedora Remix3 meets that goal in spades! You certainly could build your own system from the ground up, and we would encourage you to download Fedora 10 and do that when you have a few months of free time on your hands. The new Fedora 10 build is a perfect platform for Asterisk and the latest state-of-the-art hardware. In the meantime, our rendition which configures everything to better support Asterisk in a mobile telephony environment should save you about 500 man-hours. Try it. You’ll see. 😉
We also wanted the new system design to include every imaginable communications bell and whistle on the planet including a flexible, turnkey virtual private network implementation, transparent support for wired and wireless networks, a built-in preconfigured softphone which is ready for business, and all of the Nerd Vittles utilities and FreePBX® functionality that has made PBX in a Flash such a hit.
Finally, a new Mondo backup script has been included that lets you clone your entire system to a $20 bootable USB flash drive for incredibly easy system recovery in the event of a hardware catastrophe. And the 2008 introductory price for these built-to-order systems: just $499 plus shipping to US-48 destinations. And there’s loads of documentation, too. With a little luck, a self-installing, bootable flash drive appliance for our friends outside of the United States should be available by early next year.
About the Face Lift. Well, it’s been a painful few days at Nerd Vittles Headquarters. Our former hosting provider, BlueHost, apparently hired a new recruit that deemed our CPU utilization unworthy… in the middle of the night last Thursday. He promptly shut down our site. For any of you considering shared hosting, this is one of the dirty little secrets of the industry. They may promise you unlimited disk storage and unlimited bandwidth, but they don’t really mean it. I’m reminded of Mark Twain’s old adage about bankers: "Bankers are the folks that hand you an umbrella when the sun is shining and want it back the minute it starts to rain." Internet hosting providers have some of the same gene pool unfortunately.
The sad part of the story is that BlueHost is one of the better providers in the United States, and we, in fact, have recommended them. Hundreds of our readers took us up on our BlueHost recommendation. It gets even worse. We provided free Asterisk support to the BlueHost folks about a year ago when they were attempting to reconfigure their queues. We even brought in a local consultant in their area to assist. Do you think we even got a return call from our fair-weather friends when we were trying to figure out why our site suddenly became a problem? Our site utilization has been fairly steady for more than two years! Suffice it to say, the phone never rang. But that’s all history now. Nerd Vittles has moved to our new high-performance server at WestNIC that also hosts the PBX in a Flash Forum, and we’re happy to be there.
Nothing’s ever simple, of course. WestNIC employs PHP5 while BlueHost still was using PHP4. Even though cPanel made the server transition easy, our particular version of the WordPress blogging software was more than a little long in the tooth. Everything at first appeared to work fine. But it turned out that you could no longer read individual posts. Call us picky but that was a deal breaker. What to do? Suffice it to say that 17 version upgrades later, we’re now current. The only fatality was a few recent comments which got deleted by operator error… mine. 🙄
All good blogs deserve a facelift at least once every five years, don’t you think? Well, we’re about a month shy of our Fifth Anniversary, but it was worth the effort. And the performance boost is nothing short of amazing. We hope you agree. Enjoy!
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…
- If you don’t know what an Orgasmatron Build is, use the search function at the top of this page. [↩]
- And speaking of new worlds, lawyers love footnotes so you’d better get used to these little numbers. 🙂 We’ll break you in easy today. There are just a few of them. [↩]
- Fedora and the Infinity design logo are trademarks of Red Hat, Inc. Asterisk is a registered trademark of Digium, Inc. All other trademarks and registered trademarks are property of their respective owners. This software aggregation is neither provided nor supported by the Fedora Project and contains non-Fedora and modified Fedora content. Official Fedora software is available through the Fedora Project website. [↩]
The Lean, Mean Asterisk Machine: And Now It’s a Fax Machine
Hard to believe it’s been a year since PBX in a Flash hit the street, but today’s the Big Day! So Happy Birthday to us. With an estimated 100,000 downloads worldwide and over a million RSS feeds to our Kennonsoft User Interface each month, you might be wondering what keeps us going with all the reported venture capital behind Big Orange and Lime Green. Well, we’re glad you asked. Truth be told, it’s the cushy offices (in our kitchens) and the endless flow of generous contributions from grateful users. Heh, heh! Seriously, there are some real reasons that account for the popularity of PBX in a Flash. Bottom Line: It Just Works! And here’s a representative sample of other feedback from our fans:
- Currency – The PBX in a Flash distribution is always up to date. Our separate payload file makes it easy. No one else has anything close. So their builds are almost always long in the tooth.
- Upgradability – Unlike the competition, you don’t have to start all over each time a new version of Asterisk® or Linux hits the street. We’ll have more to say about our new SUSHI (Software Update Service – Hyperlinked, Interactive) in coming weeks.
- Flexibility – PBX in a Flash remains the only distribution that builds Asterisk from source. Even Digium®’s own distribution now uses RPMs. When you add new hardware or upgrade the Linux kernel to plug a security vulnerability, you’ll understand why this is critically important.
- Support – PBX in a Flash has the best support group in the business. It’s called the PBX in a Flash Forum, and it’s free. Unlike the competition, you don’t have to pay to get help on basic technical issues with our product. And you don’t normally wait more than an hour or two for a response. That’s what Open Source is all about!
- Security – We take security seriously. It’s our number one priority. When there’s a known problem, we don’t hide it or ignore it. We fix it right now. And the RSS Feed that’s part of our KennonSoft User Interface lets you know about it immediately. You can make your own comparisons and draw your own conclusions with regard to the other distributions.
- No Slimeware – We’re up front about the way we operate and why. We don’t create backdoors or Trojan Horses in our distribution that phone home for any reason. We notify users of issues through an RSS Feed. We believe it’s up to you, not Big Brother, to decide whether to protect your own system. As permitted by the GPL, we do encrypt some of our freeware installation scripts because of the conduct of some in this business that pass off the work product of others as their own.
- No Bugs – People chuckled when we began a year ago with this mantra because of the experience we all had in days of old. We still believe it and do our best to keep the PBX in a Flash distribution bug free. If you don’t believe it, visit our forums and then visit the others. Some bugs obviously are beyond our control, but we do endeavor to steer users toward stable versions of open source products that can be used reliably in almost any business environment.
So there’s a quick update on how we’re doing and why we do things the way we do. Unlike a year ago, there are lots of choices now in the marketplace. If you’re still on the fence, the nice part of the open source movement is that it doesn’t cost you anything to try several flavors and make your own decision. Ultimately, we think you’ll choose PBX in a Flash for all of the reasons we’ve mentioned.
2011 Update: This article has been updated to support Asterisk 1.8 using HylaFax, AvantFax, and IAXmodem. Click here for the latest article.
Welcome Back Faxing. That brings us to today’s topic: adding a fax machine to your PBX in a Flash system. With all the distributions, there have been numerous fax options. And the one word that describes most of them is P-A-I-N-F-U-L. We’ve been searching for a way to return to the good ol’ Asterisk@Home days with NVfax. It just worked. Well, today it works again with PBX in a Flash and Asterisk 1.4. And, yes, it should work on the other distributions as well. I’ve had mixed emotions about whether to protect the install script, but I’ve chosen to release it in unencrypted format because I think we all can benefit from the contributions of others while still giving credit to those that contribute. And, yes, I know there’s a difference of opinion about this… for some very good reasons. But the Nerd Vittles contribution to VoIP technology has always been distribution agnostic, and we’ve decided to keep it that way. We’re equally delighted that Philippe Lindheimer has left the hooks in FreePBX to support NVfax so, once you complete this install, you can manage incoming fax calls from the comfort of the FreePBX user interface… even in distributions which no longer call it FreePBX. Ever wonder why these folks didn’t also rename Asterisk while they were in the lobotomy business?
How It Works. There are two pieces to the new faxing mechanism. For inbound faxing, you simply set FreePBX to use NVfax to listen for a fax tone on inbound trunks. We’ve found that 5 is the magic number for detecting a fax tone on most inbound calls. YMMV! You also can dial local extension 329 (F-A-X) and the extension will listen for an incoming fax. In either instance, if a fax tone is detected, the call is routed to a fax context that converts the incoming fax to a PDF document which is then sent to your email address specified in your Fax Handling setup for each Inbound Route on your system. The correct answers for Fax Handling are Fax Extension: System, Fax Email: any email address that works, Fax Detection Type: NVFax, and Pause After Answer: 5. Don’t forget to also enter the Fax Machine Settings under the Setup->General Settings tab in FreePBX. For outbound faxing, we can’t recall this ever working with NVfax, but it does now. Here’s how to set things up. Create a PDF document of anything you wish to send by fax. Name the document so that it corresponds with the phone number of the fax destination, e.g. 6789991234.pdf would mean you plan to send the PDF document to a fax device at the following phone number: 678-999-1234. Now place the document in the /tmp directory on your server. Next, pick up a phone on your system and dial 32948 (F-A-X-I-T). When prompted for the destination fax phone number, key in 6789991234. Once you receive an acknowledgment that your fax has been sent, hang up. It doesn’t get much easier than that.
Prerequisites. Well, there are lots of them. But a stock installation of Asterisk with CentOS works great so long as you also have outbound emailing working and you’ve installed a text-to-speech engine. Either Flite or Cepstral works just fine. All of the bundled distributions should suffice. We actually only use TTS to generate the voice prompts for the outbound faxing so, if you don’t need that functionality, no TTS engine is required. If you need help with outbound emailing, see our PBX in a Flash knol. There also are setup instructions for Gmail and Comcast in the PBX in a Flash forum.
Installing the Fax Software. We’ve written a script which handles all of the heavy lifting for you. Just log into your server as root and issue the following commands:
cd /root
wget http://pbxinaflash.net/source/fax/fax.pbx
chmod +x fax.pbx
./fax.pbx
In less than a minute, you should be all set.
Configuring the Fax Software. First, edit the [faxit] context in /etc/asterisk/extensions_custom.conf to plug in your actual fax number to be displayed on outbound faxes. It should be the 17th line up from the bottom of the file. Save your changes and reload Asterisk: amportal restart. Now load FreePBX using your favorite browser and make the Fax Machine entries in Setup->General Settings. Remember that your return email address must match your server domain name that you set up in /etc/hosts to get outbound email flowing, e.g. pbx.dyndns.org. Next, for each of your Inbound Routes in which you wish to enable fax detection, edit the entry and fill in the Fax Handling options we previously mentioned. To repeat, the correct answers are Fax Extension: System, Fax Email: any email address that works, Fax Detection Type: NVFax, and Pause After Answer: 5. Finally, add Misc Destinations for Fax (329) and FaxIt (32948). Reload your dialplan, and you should be ready to go.
Testing Things Out. The easiest way to assure that your system is properly configured is to attach a real fax machine to an FXS device on your system. Then send a fax to extension 329 (F-A-X). You should receive the fax via email shortly thereafter. That’s only half the battle unfortunately. If you want to receive faxes from outside your PBX, you also need to find a VoIP provider that properly supports faxing. Suffice it to say, all VoIP providers are not created equal when it comes to fax support. Our Best of Nerd Vittles article on faxing will provide some suggestions as well as a few tips and tricks. If you have a standard POTS line connected to an FXO device on your Asterisk server, that’s an even better option. Just make certain that fax detection is enabled on the inbound route for that line.
Don’t be misled by the brevity of this article. It in no way is a measure of the effort that it’s taken to make NVfax work again. One way that you can show your appreciation for the good deeds of others is through the Donate link at the top of our page. There’s no obligation, of course, but it does keep the Little Mrs. from regularly asking, "Tell me again why you do this?" Enjoy and thanks in advance.
Getting Started with PBX in a Flash. There’s a great deal of literature on PBX in a Flash that is yours for the taking. But we wanted to mention a terrific new series of articles in Mark Berry’s blog that are especially well suited for those just learning about VoIP. Have a look. We think you’ll agree.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
Introducing Noojee Click for Asterisk: The Free Click-to-Dial Solution for Firefox Using AJAM
AJAM is a new technology available in Asterisk® 1.4 which allows web browsers or other HTTP-enabled applications and web pages to directly access the Asterisk Manager Interface (AMI) via HTTP. You can read muppetmaster's complete writeup at voip-info.org. Because of Apache, it was Asterisk's mini-web server that rarely was needed or enabled with PBX in a Flash, Elastix, trixbox, or any other Asterisk aggregation... until recently. Then along came a terrific app called Noojee Click for Firefox from Noojee Telephony Solutions in Australia. What this little gem provides is Asterisk click-to-dial functionality with AJAM for any phone number found on any web page you happen to be browsing with Firefox. See, for example, the U.S. Senate web site (shown below). You'll note that little Noojee icons are displayed beside each senator's telephone number. So give Barack and John a call, and tell them what's on your mind. To call your favorite senator, just click on the desired icon. And, because it's a Firefox Addon, it's operating system agnostic so it works well on almost any desktop computer. We're going to walk you through getting everything set up today with Asterisk, but we have one cautionary note.
WARNING: We strongly urge you NOT to expose AJAM or the Asterisk Manager Interface to public access over the Internet or to untrusted users! Doing so almost certainly will compromise the integrity of your Asterisk system without a significant amount of additional work (not covered in this article) to harden and broaden the number of passwords and to better secure these services. Having said that, Noojee Click is a terrific tool for use on a private intranet where you trust ALL of the users or via VPN access where you trust ALL of the users with VPN access. And that's what we'll cover today in this tutorial.
Prerequisites. In order to follow along in this tutorial, you'll need a properly configured Asterisk 1.4 system as well as the Noojee Click Addon for Firefox which must be installed and properly configured on your client machines. We also are assuming that your Asterisk 1.4 implementation includes a relatively current version of FreePBX that is functioning reliably. We've given up installing every release of every Asterisk aggregation on the planet. So... what follows assumes you're using PBX in a Flash. If the other aggregations are properly configured, the same instructions should work without any changes. But we haven't tested them so proceed at your own risk.
Activating AJAM. Asterisk 1.4 as compiled on PBX in a Flash systems comes with AJAM support built in but not activated. Here are the three steps to activate and test that it's working properly.
1. Copy the sample AJAM config file into the /etc/asterisk directory:
cp /usr/src/asterisk/configs/http.conf.sample /etc/asterisk/http.conf
2. Using your favorite editor, edit http.conf so that it looks like the following. Be sure to change the private IP address 192.168.0.236 to reflect the private IP address of your Asterisk server or, in the alternative, you can use the VPN IP address of your Asterisk server .
;
; Asterisk Builtin mini-HTTP server
;
;
[general]
;
; Whether HTTP interface is enabled or not. Default is no.
;
enabled=yes
;
; Whether Asterisk should serve static content from http-static
; Default is no.
;
;enablestatic=yes
;
; Address to bind to. Default is 0.0.0.0
;
bindaddr=192.168.0.236
;
; Port to bind to (default is 8088)
;
bindport=8088
;
; Prefix allows you to specify a prefix for all requests
; to the server. The default is "asterisk" so that all
; requests must begin with /asterisk
;
prefix=asterisk
;
; The post_mappings section maps URLs to real paths on the filesystem. If a
; POST is done from within an authenticated manager session to one of the
; configured POST mappings, then any files in the POST will be placed in the
; configured directory.
;
;[post_mappings]
;
; In this example, if the prefix option is set to "asterisk", then using the
; POST URL: /asterisk/uploads will put files in /var/lib/asterisk/uploads/.
uploads = /var/lib/asterisk/uploads/
;
3. Now issue the following commands to configure and restart Asterisk and make sure AJAM is functioning properly:
mkdir /var/lib/asterisk/uploads
chown asterisk:asterisk /var/lib/asterisk/uploads
amportal restart
asterisk -rx "http show status"
You should receive a response that looks something like the following:
HTTP Server Status:
Prefix: /asterisk
Server Enabled and Bound to 192.168.0.236:8088
Enabled URI's:
/asterisk/httpstatus => Asterisk HTTP General Status
/asterisk/manager => HTML Manager Event Interface
/asterisk/rawman => Raw HTTP Manager Event Interface
/asterisk/static/... => Asterisk HTTP Static Delivery
/asterisk/mxml => XML Manager Event Interface
Adjusting Security Settings to Permit Noojee Click Access. Two of the default security settings on PBX in a Flash systems would prevent Noojee Click running on any PC inside or outside your private network from accessing your Asterisk server to place a call. We want you to be fully aware that we are loosening up security to permit this access so that you'll know how to reverse it if you change your mind. First, the IPtables firewall blocks TCP port 8088 access to your Asterisk server from any external machine. Second, the default Asterisk Manager configuration blocks access to the Asterisk Manager Interface except from the Asterisk server itself. So the next two sections will walk you through adjusting both the IPtables firewall setup and the Asterisk Manager configuration to permit Noojee Click access.
Adjusting IPtables for TCP Port 8088 Access. We always recommend that your Asterisk server be installed behind a hardware-based firewall/router with all web access blocked. IPtables is configured to permit access to port 80 and several other web ports; however, this is intended to allow private intranet access to your server, not public Internet access. We're going to unblock TCP port 8088 with the same cautionary note. Do NOT expose TCP port 8088 to the public Internet! If you cannot live without Internet access to your server, use a VPN tunnel to make the connection so that all of the data is secured and the connection does not expose unencrypted data and passwords to the public Internet.
To open TCP port 8088 on your IPtables firewall, add the following line to the bottom of /etc/sysconfig/iptables just above the COMMIT line:
-A INPUT -p tcp -m tcp --dport 8088 -j ACCEPT
Then restart the IPtables service:
service iptables stop
service iptables start
Adjusting Asterisk Manager for Local Subnet Access. Giving any user Asterisk Manager access is equivalent to handing over the keys to your Asterisk castle. If you have any doubt about the integrity of any user on the subnet on which your Asterisk server is running, don't follow these instructions. Instead, consult an expert and limit access by individual IP addresses with separate account names and passwords for each trusted machine on your network, e.g. permit=192.168.0.31/255.255.255.255. This also could be the VPN address of any remote machine. To enable Noojee Click access to the Asterisk Manager for your entire local subnet, edit /etc/asterisk/manager.conf and add a new context at the bottom of the file that looks like the following using the actual subnet address of your intranet instead of 192.168.0.0. AND be sure to use a very secure password:
[noojee]
secret = YourVerySecurePasswordGoesHere
deny=0.0.0.0/0.0.0.0
permit=192.168.0.0/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
In the [general] context of the same file, add the following entry:
webenabled = yes
Save your changes and restart Asterisk: amportal restart.
Testing AJAM on Your Server. The easiest way to be sure you have a correct setup on your system is to try to access AJAM with a browser. First, install the text-based lynx browser. Issue the command: yum install lynx. Once installed, type lynx to start it up. Then choose G and enter the following URL using your actual IP address and password from above, of course:
http://192.168.0.236:8088/asterisk/manager?action=login&username=noojee&secret=YourVerySecurePasswordGoesHere
Installing and Configuring Noojee Click. Now that AJAM is humming along, you're ready to install Noojee Click on each of your desktop machines. On every machine, you'll need to fire up Firefox and go to this link. Click on the provided link to install Noojee Click for Asterisk. Firefox will display the following message just under the tab bar: 'Firefox prevented this site (www.noojee.com.au) from asking you to install software on this computer.' Click the 'Allow' button on the right hand side of your browser to allow Noojee Click to be installed. Then repeat the process again. You'll need to restart Firefox to finish the install. When Firefox reloads, you'll see the Noojee logo in the bottom right hand corner of the Firefox status bar. Click on it and choose Configuration. Enter the following settings using the IP address of your Asterisk server, noojee for the username, and whatever password you chose above for Asterisk Manager access:
ServerType: AJAM (Asterisk 1.4+)
Host: Internal IP Address of Your Asterisk Server
Port: 8088
Username: noojee
Password: YourVerySecurePasswordGoesHere
Phone Extension: the extension number where you will pick up this outbound call (works like AsteriDex!)
Context: from-internal
Enable Autoanswer: your choice
Phone Type: your choice (use Aastra for softphones)
Dial Prefix: only if required by your existing trunk setup
International Prefix: only if required by your setup
Pattern: leave the ones that are there and add the following for calls in the U.S.
XXX-XXX-XXXX
(XXX) XXX-XXXX
Enable Logging: your choice
Press the Escape key to save your settings. Now access a web page with some phone numbers and click on the Noojee icon beside a phone number to place an outbound call. The extension you specified in the Configuration should begin to ring. Answer the call, and the outbound call will be placed. Enjoy!
Hosting Provider Mega Deal. 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 Fonica Special. If you want to communicate with the rest of the telephones in the world, then you'll need a way to route outbound calls (terminations) to their destination. For outbound calling, we recommend you establish accounts with several providers. We've included two of the very best! These include Joe Roper's new service for PBX in a Flash as well as our old favorite, Vitelity. To get started with the Fonica service, just visit the web site and register. You can choose penny a minute service in the U.S. Or premium service is available for a bit more. Try both. You've got nothing to lose! In addition, Fonica offers some of the best international calling rates in the world. And Joe Roper has almost a decade of experience configuring and managing these services. So we have little doubt that you'll love the service AND the support. To sign up in the USA and be charged in U.S. Dollars, sign up here. To sign up for the European Service and be charged in Euros, sign up here. See the Fonica image which tells you everything you need to know about this terrific new offering. In addition to being first rate service, Fonica is one of the least expensive and most reliable providers on the planet.
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...
It’s a Dell With Asterisk, Dude: Introducing the Orgasmatron II for the Dirt-Cheap Dell SC440
About 20 years ago, we began our migration from proprietary DEC VAX minicomputers by acquiring two of the first Dell servers ever manufactured. They were serial numbers 6 and 7. And we were just about as excited about the transition as the folks on Sesame Street. The machines were quickly named Bert and Ernie, and there was even a scrolling LCD display on the units showing the machine names. Considering that this all occurred inside a federal courthouse, it was revolutionary at the time. Michael Dell has gotten a little richer since that initial $10,000 investment (a bargain at the time!), and I'd have to say Dell servers have improved a good bit as well. So we finally bit the bullet last week and bought two of Dell's SC440 servers when they went on sale for a whopping $199 each. For this price, you got Dual Core Intel® Pentium®E2180, 2.0GHz processors with 1MB Cache, an 800MHz FSB, an 80GB 7.2K RPM Serial ATA 3Gbps 3.5-in Cabled Hard Drive connected to the onboard SATA controller, 512MB of 667MHz DDR2 RAM, a 48X CD-ROM Drive, and an On-Board Single Gigabit Network Adapter. For $19 more, you got 2GB of RAM. We hope some of you also took advantage of the offer because today we're releasing our plug-and-play Osgasmatron II for the SC440. Nov. 13 NEWS FLASH: Dell once again is offering the SC440 with Dual-Core processor but it now includes a second hard disk for $299 until November 19. It's still a great deal on a top-notch server. To get email alerts when the SC440 again goes on sale, go to techbargains.com and search for SC440. Then click on Send Email Deal Alert. Be sure to confirm the alert by replying to the email. These units have gone on sale roughly every two weeks since early September.
If you've missed the last two week's articles, the Orgasmatron II is the Ultimate Kitchen Sink for Asterisk®. It includes PBX in a Flash 1.3 in all its glory plus the newly released FreePBX 2.5 and so much more. From the time you insert the CD 'til you have a functioning Asterisk PBX with all the bells and whistles imaginable... 15 minutes!
If you've been following along with our articles, you already know that we've identified what we believe to be the perfect Asterisk SIP phone, the Aastra 57i. But our previously anointed perfect small business/home computer on which to run a production Asterisk server for about 50 employees, the Everex gPC2 (aka "The WalMart Special"), is no more. So this build moves to a different platform and a very different performance level. You'll see about twice the performance on the SC440 compared to the WalMart Special. Today's build provides a preconfigured SC440 installation on a 2-disk ISO image backup of the whole system using Mondo. And, NO, it won't work with any other hardware! Once you download the ISO images and burn your CDs, it's a 15-minute No-Brainer to install the entire image onto your own SC440. Wait to install any add-on cards until after you complete the Orgasmatron install. You must have an SC440 configured as above, or this Mondo restore will not work. So accept no substitutes, or you may end up with an Electronic Brick instead of an Orgasmatron II.
We've preconfigured some extensions on your new system as well as outbound and incoming trunks from some terrific providers including our second homegrown entry for VoIP terminations. Joe Roper and his business partner in Spain now offer a terrific IAX VoIP termination service. You can choose penny a minute service in the U.S. and most of Canada, or you can opt for premium VoIP service at about 2¢ a minute in the U.S. International rates also are VERY reasonable! You literally can sign up for service, plug in your phones, and have a system in full operation in under an hour.
So... what do you get with this preconfigured build? In addition to all of the goodness of a stock PBX in a Flash 1.3 build including Asterisk 1.4.21.2 running under CentOS 5.2 with a version of Zaptel that actually works with legacy cards. You also get the brand new FreePBX 2.5 as well as the latest versions of Apache, MySQL, PHP, and SendMail. And you get a Baker's Dozen preconfigured Nerd Vittles applications. Complete documentation is available here.
- Inbound and Outbound VoIP Faxing Using nvFax... finally!
- FONmail for Asterisk to send voice messages to any email address on the planet
- AsteriDex RoboDialer and Telephone Directory
- Telephone Reminders with Support for Recurring Reminders and Web-based TTS Reminder Messages
- NewsClips for Asterisk featuring Dozens of Yahoo News Feeds (TTS)
- Weather Reports by Airport Code (TTS)
- Weather Reports by ZIP Code (TTS)
- Worldwide Weather Forecasts (TTS)
- xTide for Asterisk (TTS)
- MailCall for Asterisk: Get Your Email By Telephone (TTS)
- TeleYapper 4.0 Message Broadcasting System
- CallWho for Phone Lookup and Dialing of Entries in the AsteriDex Database (TTS)
- TFTP Server with preconfigured setups for 10 Aastra 57i SIP telephones
In addition, you get dozens of preconfigured telephony applications and functions that would take even an expert the better part of a year or two to build independently. And, unlike all of the other distributions, we build Asterisk from source so it's simple to modify and upgrade whenever you feel the need. Here's a short list of what you have to look forward to:
- Stealth AutoAttendant with Welcome and Application IVRs
- Key Telephone Support Using Park and Parking Lot
- Intercom/Paging Support
- Bluetooth Proximity Detection with Automatic Call Forwarding to Cell Phone
- DISA
- Blacklisting with Web and Telephony Interfaces
- CallerID Name Lookups from Numerous Providers
- Weekly Automated System Backups to a Flash Drive
- One Touch Day/Night Service
- Music on Hold
- Voicemail with Email Delivery of Messages and Pager Notification
- Voicemail Blasting
- Cell Phone Direct Dial
- Call Forward: All, Busy, No Answer
- Call Waiting
- Call Pickup
- Zap Barge
- Call Transfer: Attended and Blind
- Dictation Service with Email Delivery
- Do Not Disturb
- Gabcast
- Phonebook Dial by Name
- Speed Dial
- Flite Text to Speech (TTS)
- Windows Networking with SAMBA
- Linux Firewall and Fail2Ban with SSH, HTTP, and SIP/IAX login protection
- PBX in a Flash Software Update Service To Keep Your System Current
- One-Click Cepstral TTS Install with Allison... Just Type install-cepstral
Prerequisites. As mentioned, you'll need an SC440 configured with the specs outlined above including the 2GB RAM upgrade. We also recommend a 4GB USB flash drive on which to store automatic weekly backups of your new system. Just plug it into your new machine, log in as root, and type: /root/usbformat.sh. That's it! Every Sunday night, you'll get a new backup in ISO format on your flash drive. If something goes wrong on your system, copy the ISOs to CDs and reboot with Disk 1. It doesn't get any easier than that. And you can always check on the latest backup by issuing the command: /root/usbcheck.sh
Finally, you'll need to cough up a whopping $5 to download the two-disk ISO image for this build. And, yes, we eat our own dog food. The ISO images you'll be downloading were captured as a backup on the flash drive of one of our SC440 lab machines. We got 'em yesterday! If you use this special build, it seemed only fair that you cover the cost of the bandwidth to download it. As most of you know, we don't have the luxury of freeloading off SourceForge for our downloads. And we didn't want to impose upon our existing bandwidth providers to bring you this custom image. The good news is that, once you download the image from DreamHost, you are more than welcome to pass it along to one or more of your friends or business acquaintances at no charge. You can even do it electronically through the DreamHost Files Forever program. And, if you'd like to host this image for your fellow man at no cost, be our guest... and thank you! Bottom line: For under $250, you'll have the slickest, fastest, most reliable PBX and fax machines on the planet with rock-solid weekly backups and, of course, access to the one-of-a-kind PBX in a Flash Software Update Service!
Getting Started. Once you have your SC440 in hand, take it out of the box, plug it into your LAN with DHCP and DNS support and Internet connectivity. You'll need a USB keyboard for typing temporarily. We also strongly recommend that you always keep your system running behind a NAT-based firewall/router. We strongly recommend the dirt-cheap dLink WBR-2310 WiFi router which handles NAT issues with VoIP masterfully. Don't redirect any ports to the machine and don't turn the PC on just yet.
Download the two ISO images for the SC440 from here. If you don't know how to create a CD from an ISO image, read that section from our previous article. In fact, read the whole article. It'll help you immensely down the road. Once you have the two CDs in hand, turn on the SC440 and quickly insert Disk 1 into the CD drive and close the drive. If you don't see a Mondo Rescue screen within a minute or less, turn the machine off and then back on again. At the Mondo Rescue main screen, type nuke and press the Enter key. This will erase, repartition, and reformat your hard disk in case you didn't know. This is normal. If you get any kind of errors about incorrect drive or partition names, halt the install by pressing CTL-ALT-DEL and remove the CD. Otherwise, ignore the errors. You'll need to install PBX in a Flash using our standard ISO which is available here. Otherwise, go have a cup of coffee and come back in about 10 minutes. After fileset #87 is restored, you'll be prompted to insert Disk 2 and press Enter to finish the install. When the second CD finishes, eject it and wait for the prompt. Then type "exit" and press Enter. Your SC440 will reboot, and you're ready to go.
After the reboot finishes, type root at the login prompt for your username and password for your password. The IP address assigned by your DHCP server should appear near the top of the screen. Write it down. If there is no IP address, your machine does not have network connectivity or access to a DHCP server with an available IP address. Correct the problem and reboot. You can safely ignore the warning that Fail2Ban is OFFLINE. We've updated the Fail2Ban software to protect Asterisk SIP and IAX connections, and our status program isn't up to date as this article goes to press. update-fixes will get you a new version from our Software Update Service shortly.
Securing Passwords. We're going to change five passwords now. For the time being (until you've done some reading), think up one really difficult password (that you won't forget) and use it for all five passwords. At the root@pbx:~ $ command prompt, type the following commands and type in your new password when prompted. Don't forget your password or you'll get to put in your two CDs and start over.
passwd
passwd-maint
passwd-wwwadmin
passwd-meetme
/usr/libexec/webmin/changepass.pl /etc/webmin root yournewpasswordhere
Now, using a web browser, go to the IP address of your new PBX in a Flash server. Click the Admin tab and then choose the FreePBX Administration botton. Log in as maint with your new maint password. Before you do anything else, change ALL of the 10 extension passwords to something secure... as if your phone bill depended upon it! Click Setup, Extensions and then choose each extension, modify BOTH the device secret and Voicemail Password, and click Submit. When you finish all the extensions, then reload the dialplan to save your changes. Finally, change your DISA password to something very, very secure: Setup, DISA, DISAmain, PIN. Reload your dialplan once again to save your changes.
Regardless of what you may read elsewhere, the Orgasmatron II has all the very latest security patches as of today. If you want more security, take our advice and add a hardware-based firewall/router between your Internet connection and your new Orgasmatron II and don't expose port 80 (the web interface) to the Internet!
Permanently Setting the IP Address. There are different schools of thought on whether to use a fixed or dynamic IP address. Most hardware-based routers support DHCP IP address reservations. The simplest way to permanently secure the existing IP address for your server is to reserve it on your router. If you'd prefer to assign your own IP address, we have included the deprecated netconfig utility which can be run after logging into your server as root. Sometimes you will need to run it once, enter your settings, reboot, and then repeat the drill. Then you should be all set. Either way, you need a permanent IP address for your machine when all is said and done. Once you have a permanent IP address, hop on over to dyndns.org and sign up for your own fully-qualified domain name (FQDN), e.g. mypbx.dyndns.org. You're going to need it for a whole host of things with your new PBX, and dyndns.org is about the easiest way to do it. Once you have your FQDN and DynDNS username and password, log in as root and edit: /etc/ddclient/ddclient.conf. Search (Ctl-W) for ***. Fill in your username and password and uncomment those two lines. Then search for *** again, uncomment the next three lines and fill in your fully-qualified domain name. Save the file and service ddclient restart. To make sure everything worked, issue the following command: ddclient -force. Assuming there are no errors, issue the following command to start ddclient each time your server reboots: /sbin/chkconfig --add ddclient. Now the IP address of your Asterisk server will always resolve to your FQDN from DynDNS. And anyone can call you via SIP for free using the following SIP URI: mothership@yourFQDN.dyndns.org. You can take this a step further and sign up for a free incoming phone number at ipkall.com. For your account type, choose SIP. For your SIP phone number, enter: mothership. For your SIP proxy, enter the fully-qualified domain name (FQDN) for your server, e.g. mypbx.dyndns.org. Choose a password and enter your real email address, and they will beam you a Washington state phone number within a day or so. You can't beat the price!
Adding Plain Old Phones. Before your new PBX will be of much use, you're going to need something to make and receive calls, i.e. a telephone. For today, you've got several choices: a POTS phone, a softphone, or a SIP phone (highly recommended). Option #1 and the best home solution is to use a Plain Old Telephone or your favorite cordless phone set (with 8-10 extensions) if you purchase a little device (the size of a pack of cigs) known as an SPA-2102. It's under $70. Be sure you specify that you want an unlocked device, meaning it doesn't force you to use a particular service provider. Once you get it, plug the device into your LAN, and then plug your phone instrument into the SPA-2102. Note that this adapter supports two-line cordless phones! Your router will hand out a private IP address for the SPA-2102 to talk on your network. You'll need the IP address of the SPA-2102 in order to configure it to work with Asterisk. After you connect the device to your network and a phone to the device, pick up the phone and dial ****. At the voice prompt, dial 110#. The device will tell you its DHCP-assigned IP address. Write it down and then access the configuration utility by pointing your web browser to that IP address.
Once the configuration utility displays in your web browser, click Admin Login and then Advanced in the upper right corner of the web page. When the page reloads, click the Line1 tab and then repeat this drill for the Line2 tab if you want to connect the device to two extensions on your Asterisk system. Scroll down the screen to the Proxy field in the Proxy and Registration section of the form. Type in the private IP address of your Asterisk system which you wrote down previously. Be sure the Register field is set to Yes and then move to the Subscriber Information section of the form. Assuming you're using the preconfigured extensions starting with 701, do the following. Enter House Phone as the Display Name. Enter 701 as the User ID. Enter your actual password for this extension in the Password field, and set Use Auth ID to No. Click the Submit All Changes button and wait for your Sipura to reset. In the Line 1 Status section of the Info tab, your device should show that it's Registered. You're done. Now repeat the drill for Line2 using extension 702. Pick up a phone and dial 1234# to test out BOTH extensions.
Downloading a Free Softphone. Unless you already have an IP phone, the easiest way to get started and make sure everything is working is to install an IP softphone. You can download a softphone for Windows, Mac, or Linux from CounterPath. Or download the pulver.Communicator. Here's another great SIP/IAX softphone for all platforms that's great, too, and it requires no installation: Zoiper 2.0 (formerly IDEfisk). All are free! Just install and then configure with the IP address of your PBX in a Flash server. For username and password, use one of the extension numbers and passwords which you set up with FreePBX. Once you make a few test calls, don't waste any more time. Buy a decent SIP telephone. We think the best phone out there is the Aastra 57i for under $200. Another $100 buys you the Aastra 57i CT with a cordless DECT phone.
Configuring Aastra 57i SIP Phones. Your new system comes preconfigured to automatically configure up to 15 Aastra 57i phones. Plug each phone into your network and wait for it to boot. Once it boots, press the Option button, then Phone Status (3), then IP & MAC Address (1). Write down each phone's IP address and MAC address. Then press Done to exit from the menus.
Next, we need to tell your phone to use your new Asterisk server as the TFTP server to obtain its setup. Press the Option button again, then Admin Menu (5). Type 22222 for the admin password and press Enter. Then choose Config Server (1), then TFTP Settings (2), then Primary TFTP (1), enter the IP address of your new server, and press Done a half dozen times.
Log back into your server as root. Switch to the TFTP directory: cd /tftpboot. You'll notice that there are config files for up to 15 phones. Simply choose the extension number you wish to use for each phone AND rename each file (filenames are 701.cfg to 715.cfg) to the MAC address of each phone.cfg. Do NOT use hyphens in the MAC address. One final step and you'll be ready to load up your phones. We need to set the correct IP address to tell each phone where your server is located. So... issue the following command using the IP address of your new server instead of 192.168.0.123. Leave the rest of the command as it is!
sed -i 's|192.168.0.0|192.168.0.123|g' /tftpboot/aastra.cfg
Now restart each phone by pressing the Option button and then Restart Phone (6) and then the Restart button. Once the phone reboots, you can make a test call by dialing 1-2-3-4. You can get the latest news by dialing 5-1-1. Or get a weather forecast by airport code (6-1-1) or zip code (Z-I-P).
A Word About Ports. For the techies out there that want to configure remote telephones or link to a server in another town, you'll need to know the ports to remap to your new server from your firewall. Here's a list of the ports available and used by PBX in a Flash. We don't recommend exposing UDP 5038 which is used to communicate with Asterisk via the Asterisk Manager.
TCP 80 - HTTP (needed to access the web sites on your server from the Internet... not recommended!!!)
TCP 22 - SSH (needed if you want remote SSH access)
TCP 9001 - WebMin (needed if you want remote WebMin access... not recommended!!!)
UDP 10000-62000 - RTP (needed for SIP communications)
UDP 5004-5037 - SIP (ditto)
UDP 5039-5082 - SIP (ditto)
UDP 4569 - IAX2 (needed for IAX communications typically between Asterisk servers)
Setting Up Trunks for Outgoing and Incoming Calls. If you want to communicate with the rest of the telephones in the world, then you'll need a way to route outbound calls (terminations) to their destination. And you'll need a phone number (DIDs) so that folks can call you. Unlike the Ma Bell world, you need not rely upon the same provider for both. And nothing prevents you from having multiple outbound and incoming trunks to your new PBX. At a minimum, however, you do need one outbound trunk and one inbound phone number unless you're merely planning to talk to other extensions set up on your system. We've actually put all the hooks in place to make it easy for you to interconnect to other Asterisk servers, but we'll save that for another day. For today, we want to get you a functioning system so that you can place outbound calls to anywhere in the world and can receive incoming calls from anywhere in the world.
For outbound calling, we recommend you establish accounts with several providers. We've included the necessary setups for Joe Roper's new service for PBX in a Flash as well as Vitelity and AOL. To register for the service, just visit the web site and register. To sign up to the service in the USA and be charged in US Dollars, please sign up here. To sign up for the European Service and be charged in Euros, sign up here.
In addition to being one of the least expensive providers, there's also the premium service option. You can prefix any number with 000 to try it out. Give it a try. We think you'll be pleased with the service AND the pricing. DIDs for inbound service are not yet available, but Vitelity has lots of them, and there's a link below to get you started.
Vitelity: One of the Best Providers on the Planet. If you're seeking the best flexibility in choosing an area code and phone number plus reasonable entry level pricing plus high quality calls, then Vitelity is a winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. For PBX in a Flash users, sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. You can't beat the price (except with us) and the call quality is excellent as well. We've tried just about everybody.
To sweeten the pot a bit more, we've preconfigured both inbound and outbound Vitelity trunks for you. For the vitel-inbound trunk, all you'll need to do is plug in your username, password, and host assigned by Vitelity and adjust the registration string to match your assigned username and password. In FreePBX, click Setup, Trunks, SIP/vitel-inbound and make the changes. Then adjust the vitel-outbound trunk to reflect your actual username in the fromuser and username entries, your real password in the secret entry, and the correct host provided by Vitelity for your outbound calls, and you're all set. In FreePBX, click Setup, Trunks, SIP/vitel-outbound and make the changes. The same setup drill will get you going the the PIAF VoIP service as well.
To test things out, pick up a phone configured on your system and dial an area code and number of someone in the United States or Canada. Now get someone to call you using your new number. Presto! You have inbound and outbound phone service. And, if you'd like to see just how good SIP service can be, pick up a phone on your system and dial D-E-M-O. This will connect you to the PBX in a Flash hosted demo applications server at Aretta Communications.
An Alternate Outbound Calling Solution. As we said, it costs you almost nothing to add an alternate outbound calling solution to your new system. As luck would have it, adding a third outbound calling provider is now a breeze because AOL just entered the SIP terminations market with a product called AIM Call Out. We wrote about it recently, and you can read the article here. All you need is an AOL or AIM account name and $5 to get you started. The system you've just installed is preconfigured to use AIM Call Out. All you have to do is plug in your username and password, and you can immediately make calls to anywhere in the United States for under 2¢ per minute. Adding international calling is as easy as inserting the correct dial string. If you never use it, it doesn't cost you a dime. So $5 is mighty cheap insurance in our book.
First things first. Sign up for the service at this link. Your username will look something like this: johndoe@aim.com. You also will be assigned a password. Using your web browser, open FreePBX by pointing to the IP address of your new server and choosing Administration, then FreePBX. Type in admin as your username and the password you assigned to your system. From the main FreePBX menu, choose Setup, Trunks, and click on SIP/AIM in the far right column. Scroll down to the Peer Details section of the form and replace yourAIMpassword with your new password. Then replace yourAIMaccountname with your actual AIM account name. Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.
Setting Up an Alternate DID for Incoming Calls. You also may want to consider a second phone number where people can call you. For example, if Grandma and Grandpa happen to be in another state and still have an old fashioned telephone, you might consider adding an additional DID to your system in their area code. They then can make a local call to reach you by dialing the local DID. On the les.net pay-as-you-go plan, it costs less than a dollar a month plus a penny a minute for the calls. Money well spent if we do say so... and you'll sleep better.
If this setup looks a bit complicated, don't be intimidated. Remember, we're connecting your PBX to the rest of the world so people can call you! With les.net, you have a choice of rate plans for most DIDs. You either can pay $3.99 a month for unlimited inbound calls with two concurrent channels or 99¢ per month and 1.1¢ per minute with four concurrent channels. Just visit their site and click Signup to register. Once you are registered, click Login and then Order DIDs. Pick a phone number. Then click Peers/Trunks and Create New Peer. Write down the Peer Name as you will need it in a minute to set up your connection. Choose SIP for Peer Technology, RFC2833 for DTMF Mode, G.711 for Codecs, Registration for Peer Type, enter the public IP address of your server for Peer Address, make up a secure password and write it down also, specify an Outbound CallerID for your calls, and check the 10-digit dialing box. Leave voicemail unchecked since you'll handle this on your end. Save your changes.
Now choose Your DIDs and click on the one you just ordered. We now need to tie the phone number to the Peer setup you just created above. Click on the DID and select the Route to Peer which you just created. Check the Send DID Prefix box and leave everything else blank. Click Save Changes and you're finished at the les.net end. Now let's set up your inbound DID trunk in Asterisk using FreePBX.
Log into FreePBX using a web browser. Click Setup, Trunks and then Add SIP Trunk. Fill in the CallerID and then drop down to the Outgoing Settings section of the form. For Trunk Name, use the Peer Name that you created above and wrote down. It ought to look something like this: 1092832198. For Peer Details, enter the following using the Peer Name and Password you assigned at les.net:
canreinvite=no
context=from-trunk
fromuser=1092832198
host=did.voip.les.net
insecure=port,invite
nat=yes
secret=yourpassword
type=peer
username=1092832198
For Incoming Settings, use from-pstn for the User Context and enter the following User Details:
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
insecure=port,invite
nat=yes
type=user
For the registration string, enter a string like the following using your Peer Name and Password:
1092832198:yourpassword@did.voip.les.net/1092832198
Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.
Choosing a VoIP Provider That Supports Faxing. We've included a reliable fax solution in this build, and we'll cover all the details soon. We do want to give you a head start if you plan to use your new machine to handle inbound faxes. To test your machine, you can connect a real fax machine to one of the lines on an SPA-2102. Then send a fax to extension 329 (F-A-X). But first you must configure your email address in two places using FreePBX: Setup, General Settings, Email address to have faxes emailed to AND Setup, Inbound Routes, any DID / any CID, fax Email. Once you've saved your settings, send the fax and see if it's delivered to your email address. If it works reliably, then the fax and email applications on your machine are configured correctly. Unfortunately, that's only half the battle. To receive faxes from outside your system, you'll also need a DID from a provider that supports faxing. And then it's still only about a 90% proposition... on a good day. We've tested this with many, many VoIP providers. Some work. Many don't. Some, such as Vitelity, offer a faxing service for a fee. Guess what? Their regular VoIP setup doesn't support faxing. Our old friends at Telasip.com still support faxing. We've also had good luck with Future-Nine and Teliax. You can read the beginnings of our fax dissertation here for more details. With the exception of the trunk setup covered in the article, all of the remaining setup steps already have been completed on your new server!
Interconnecting Two Asterisk Servers. We've preconfigured this build to support an IAX interconnect to a second PBX in a Flash system. The trunk setup for the second machine to match the setup on this build can be printed out. The filename is /root/MainPeerTrunkSetup.gif.
Choosing a Preferred Provider. Finally, you'll need to decide whether to use PIAF-USA or AOL or Vitelity as your primary terminations provider. HINT: We're the cheapest! So we've set things up this way. This is handled in FreePBX in the Outbound Routes tab under the Default entry. You can adjust easily these in any way you like by adding trunks or moving entries up and down the list to change their priority. Just be sure to leave ENUM at the top of the list since ENUM calls are always free. If a free call isn't possible, your server will automatically drop down to the next trunk in the priority list. Don't add Vitelity to the list unless you have actually created a Vitelity account since they handle unsuccessful connections in a non-standard way which will cause FreePBX not to drop down to the next trunk to attempt a connection.
Activating the Stealth AutoAttendant for Inbound Calls. By default, all incoming calls are routed to the Day/Night Code 1 context which allows you to toggle calls between a Day setting and a Night setting by pressing *281. For builds before Rev. D, the Day setting for Code 1 is set to Ring Group 700 which rings all of the extensions on your system. If you'd prefer our Stealth Autoattendant which plays a brief greeting during which you can choose other options or direct dial extensions on your system before the call is passed to Ring Group 700, then edit Day/Night Code 1 and set the Day option to IVR: MainIVR.
A Word About Mondo Rescue. We would be remiss if we didn't mention what a fantastic open source product Mondo Rescue is. It's the sole reason that today's build was possible. Our special thanks go to the development team: Bruno Cornec, Andree Leidenfrost, and Hugo Rabson. It is the first (and only) backup software for Linux builds that actually works reliably. The best way to prove that for yourself is to download this build and try it for yourself on your Dell, dude. It has much more flexibility than what you will experience, but that would take another dozen pages to explain. We'll save that for another day. In the meantime, if you'd like more information, visit the Mondo Rescue web site.
What you need to know today is that the device name for your USB flash drive may differ from the setting of /dev/sdb1 that is preconfigured depending upon the Dude that built your Dell. If you have the Rev. D build (shown at the bottom of the DreamHost download site), simply log into your server as root and type: /root/usbdevice.sh. You're all set. With prior builds, to find out the identity of your USB stick, plug it into one of the front USB ports, log in as root, and type dmesg. Included in the output will be a section that looks something like this:
USB Mass Storage support registered.
Vendor: VBTM Model: Store 'n' Go Rev: 5.00
Type: Direct-Access ANSI SCSI revision: 00
SCSI device sdb: 2013184 512-byte hdwr sectors (1031 MB)
sdb: Write Protect is off
sdb: Mode Sense: 23 00 00 00
sdb: assuming drive cache: write through
SCSI device sdb: 2013184 512-byte hdwr sectors (1031 MB)
sdb: Write Protect is off
sdb: Mode Sense: 23 00 00 00
sdb: assuming drive cache: write through
sdb: sdb4
If the entry in bold above does not say "sdb1" then you have a little work to do. First, edit /root/usbcheck.sh and change sdb1 on the mount line to the sdb# entry shown above in bold. Save your change: Ctrl-X, Y, then Enter. Now edit /root/usbformat.sh and make the same change in the fdisk AND mkdosfs lines of the script. Save your changes. Finally edit /etc/asterisk/disk-backup.conf. Press Ctl-W and search for sdb1. Change the entry to the device name in bold above. Save your change. Now restart Asterisk with the command: amportal restart. Finally, format your new flash drive and you're ready to go: /root/usbformat.sh. Be sure to check your flash drive periodically to make certain you're getting backups: /root/usbcheck.sh.
Installing Cepstral on Your New Server. If you want real text-to-speech with Allison's familiar voice, then you'll need to buy Cepstral. It's dirt cheap for single, non-commercial use. To install it, there's still a problem with the script on your new machine unfortunately. Something has happened to Darren Sessions' archives, but luckily we still have backups. This has been fixed in Rev. D. Otherwise, to point to the uncorrupted version of the software, log in to your server as root and issue the following two commands:
sed -i 's|www.darrensessions.com/pub|pbxinaflash.net/source|' /root/install-cepstral
sed -i 's|www.darrensessions.com/pub|pbxinaflash.net/source|' /usr/local/sbin/install-cepstral
Then run install-cepstral from the command prompt. At one point you'll be asked whether to create a missing directory for the Cepstral installation. Be sure to type y at the prompt rather than just pressing the Enter key. Instructions for registering your copy of Cepstral are displayed when the install completes. For complete documentation, read our previous tutorial.
Addendum: Enabling Parking Lot. As configured, FreePBX gets confused by the 700 ring group thinking it is your default parking lot. To fix the problem, simply enable the Parking Lot feature from the FreePBX Setup tab. Click on the Enable checkbox, leave the default 70 extension to place calls in the parking lot, and choose a default location to which to send orphaned parking lot calls. Then everything works normally.
Where To Go From Here. Well, we've covered a good bit of territory today. When you're ready, move on to the second part of this article at the link below. In the meantime, you have a new phone system that works. And there are a number of PDF documents in the /root folder on your new system which are worth a read. Better yet, you can browse through all of the documentation which is available for PBX in a Flash by going here. You also can dial D-E-M-O on your new system and see just how powerful direct SIP connections can be to other Asterisk hosts (in this case, ours!)... at no cost. Finally, you can log into your server and type help-pbx for access to a treasure trove of additional features. Enjoy!
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...
The Asterisk Mother Lode: Introducing the Orgasmatron II for the $199 Everex gPC2
Well, okay. Today's creation still doesn't quite measure up to the legendary Orgasmatron... but, we're getting closer. It's been several months since we released our first Orgasmatron for Asterisk®. Much has changed both in Asterisk and in the hardware and software environment since then. So today, to celebrate the release of PBX in a Flash 1.3 and FreePBX 2.5, we're taking another stab at building the Ultimate Kitchen Sink. From the time you insert the CD 'til you have a functioning Asterisk PBX with all the bells and whistles imaginable... 15 minutes! There's now a custom build for the Dell SC440 as well. Here's the link.
Our approach today is refined a bit since the last time around. The processing overhead of CentOS 5.2 continues to make VMware problematic. Luckily, the price of hardware continues its downward spiral. So today we're comfortable recommending the best phone, the best value PC, and our own new entry in the VoIP provider sweepstakes. But, you'd better hurry, there's only one retailer still carrying the Everex Green PC: good old WalMart. And now you can even get free shipping of the unit to the WalMart store of your choice.
If you've been following along with our articles, you already know that we've identified what we believe to be the perfect Asterisk SIP phone, the Aastra 57i, and we've also identified a perfect small business/home computer on which to run a production Asterisk server for about 50 employees, the Everex gPC2 (aka "The WalMart Special"). So this build provides a preconfigured gPC2 installation on a 2-disk ISO image backup of the whole system using Mondo. And, NO, it won't work with any other hardware! Once you download the ISO images and burn your CDs, it's a 15-minute No-Brainer to install the entire image onto your own Everex gPC2. But you must have a gPC2 so accept no substitutes, or you may end up with an Electronic Brick instead of an Orgasmatron II. Once again for the reading impaired, the $199 gPC2 systems are only available from WalMart.
We've preconfigured some extensions on your new system as well as outbound and incoming trunks from some terrific providers including our own new entry for VoIP terminations. Ours is dirt cheap, of course, at just over a penny a minute in the U.S. and about half that in many parts of Canada. You literally can sign up for service, plug in your phones, and have a system in full operation in under an hour.
So... what do you get with this preconfigured build? In addition to all of the goodness of a stock PBX in a Flash 1.3 build including Asterisk 1.4.21.2 running under CentOS 5.2, you also get the brand new FreePBX 2.5 as well as the latest versions of Apache, MySQL, PHP, and SendMail. And you get a Baker's Dozen preconfigured Nerd Vittles applications. Complete documentation is available here.
- Inbound and Outbound VoIP Faxing Using nvFax... finally!
- FONmail for Asterisk to send voice messages to any email address on the planet
- AsteriDex RoboDialer and Telephone Directory
- Telephone Reminders with Support for Recurring Reminders and Web-based TTS Reminder Messages
- NewsClips for Asterisk featuring Dozens of Yahoo News Feeds (TTS)
- Weather Reports by Airport Code (TTS)
- Weather Reports by ZIP Code (TTS)
- Worldwide Weather Forecasts (TTS)
- xTide for Asterisk (TTS)
- MailCall for Asterisk: Get Your Email By Telephone (TTS)
- TeleYapper 4.0 Message Broadcasting System
- CallWho for Phone Lookup and Dialing of Entries in the AsteriDex Database (TTS)
- TFTP Server with preconfigured setups for 15 Aastra 57i SIP telephones
In addition, you get dozens of preconfigured telephony applications and functions that would take even an expert the better part of a year or two to build independently. And, unlike all of the other distributions, we build Asterisk from source so it's simple to modify and upgrade whenever you feel the need. Here's a short list of what you have to look forward to:
- Stealth AutoAttendant with Welcome and Application IVRs
- Key Telephone Support Using Park and Parking Lot
- Intercom/Paging Support
- Bluetooth Proximity Detection with Automatic Call Forwarding to Cell Phone
- DISA
- Blacklisting with Web and Telephony Interfaces
- CallerID Name Lookups from 8 Providers
- Weekly Automated System Backups to a Flash Drive
- One Touch Day/Night Service
- Music on Hold
- Voicemail with Email Delivery of Messages and Pager Notification
- Voicemail Blasting
- Cell Phone Direct Dial
- Call Forward: All, Busy, No Answer
- Call Waiting
- Call Pickup
- Zap Barge
- Call Transfer: Attended and Blind
- Dictation Service with Email Delivery
- Do Not Disturb
- Gabcast
- Phonebook Dial by Name
- Speed Dial
- Flite Text to Speech (TTS)
- Windows Networking with SAMBA
- Linux Firewall
- PBX in a Flash Software Update Service To Keep Your System Current
- One-Click Cepstral TTS Install with Allison... Just Type install-cepstral
Prerequisites. As mentioned, you'll need a $199 Everex gPC2 (aka The WalMart Special) to use this build. We also recommend an additional $25 gig of RAM for anything other than home use. We also recommend a 4GB USB flash drive on which to store automatic weekly backups of your new system. Just plug it into your new machine, log in as root, and type: /root/usbformat.sh. That's it! Every Sunday night, you'll get a new backup in ISO format on your flash drive. If something goes wrong on your system, copy the ISOs to CDs and reboot with Disk 1. It doesn't get any easier than that. And you can always check on the latest backup by issuing the command: /root/usbcheck.sh
Finally, you'll need to cough up a whopping $5 to download the two-disk ISO image for this build. And, yes, we eat our own dog food. The ISO images you'll be downloading were captured as a backup on the flash drive of our gPC2 lab machine. If you use this special build, it seemed only fair that you cover the cost of the bandwidth to download it. As most of you know, we don't have the luxury of freeloading off SourceForge for our downloads. And we didn't want to impose upon our existing bandwidth providers to bring you this custom image. The good news is that, once you download the image from DreamHost, you are more than welcome to pass it along to one or more of your friends or business acquaintances at no charge. You can even do it electronically through the DreamHost Files Forever program. And, if you'd like to host this image for your fellow man at no cost, be our guest... and thank you! Bottom line: For about $250, you'll have the slickest, most reliable PBX and fax machine on the planet with rock-solid weekly backups and, of course, access to the one-of-a-kind PBX in a Flash Software Update Service!
Getting Started. Once you have purchased your Everex gPC2, take it out of the box, plug it into your LAN with DHCP and DNS support and Internet connectivity. Having said that, we strongly recommend that you always keep your system running behind a NAT-based firewall/router. We strongly recommend the dirt-cheap dLink WBR-2310 WiFi router which handles NAT issues with VoIP masterfully. Don't redirect any ports to the machine and don't turn the PC on just yet.
Download the two ISO images for the gPC2 from here. If you don't know how to create a CD from an ISO image, read that section from our previous article. In fact, read the whole article. It'll help you immensely down the road. Once you have the two CDs in hand, turn on the gPC2 and quickly insert Disk 1 into the CD/DVD drive and close the drive. If you don't see a Mondo Rescue screen within a minute or less, turn the machine off and then back on again. At the Mondo Rescue main screen, type nuke and press the Enter key. This will erase, repartition, and reformat your hard disk in case you didn't know. This is normal. If you get any kind of errors about incorrect drive or partition names, halt the install by pressing CTL-ALT-DEL and remove the CD. You'll need to install PBX in a Flash using our standard ISO which is available here. Otherwise, go have a cup of coffee and come back in about 12 minutes. When prompted, insert Disk 2 and press the Enter key to finish the install. When the CD ejects, remove it and your gPC2 will reboot after you perform the three-finger salute (Ctl-Alt-Del).
After the reboot finishes, type root at the login prompt for your username and password for your password. The IP address assigned by your DHCP server should appear near the top of the screen. Write it down. If there is no IP address, your machine does not have network connectivity or access to a DHCP server with an available IP address. Correct the problem and reboot.
Securing Passwords. We're going to change five passwords now. For the time being (until you've done some reading), think up one really difficult password (that you won't forget) and use it for all five passwords. At the root@pbx:~ $ command prompt, type the following commands and type in your new password when prompted. Don't forget your password or you'll get to put in your two CDs and start over.
passwd
passwd-maint
passwd-wwwadmin
passwd-meetme
/usr/libexec/webmin/changepass.pl /etc/webmin root yournewpasswordhere
Now, using a web browser, go to the IP address of your new PBX in a Flash server. Click the Admin tab and then choose the FreePBX Administration botton. Log in as maint with your new maint password. Before you do anything else, change ALL of the 16 extension passwords to something secure... as if your phone bill depended upon it! Click Setup, Extensions and then choose each extension, modify BOTH the device secret and Voicemail Password, and click Submit. When you finish all the extensions, then reload the dialplan to save your changes. Finally, change your DISA password to something very, very secure: Setup, DISA, DISAmain, PIN. Reload your dialplan once again to save your changes.
Regardless of what you may read elsewhere, the Orgasmatron II has all the very latest security patches as of October 1. If you want more security, take our advice and add a hardware-based firewall/router between your Internet connection and your new Orgasmatron II and don't expose port 80 (the web interface) to the Internet!
Permanently Setting the IP Address. There are different schools of thought on whether to use a fixed or dynamic IP address. Most hardware-based routers support DHCP IP address reservations. The simplest way to permanently secure the existing IP address for your server is to reserve it on your router. If you'd prefer to assign your own IP address, we have included the deprecated netconfig utility which can be run after logging into your server as root. Sometimes you will need to run it once, enter your settings, reboot, and then repeat the drill. Then you should be all set. Either way, you need a permanent IP address for your machine when all is said and done. Once you have a permanent IP address, hop on over to dyndns.org and sign up for your own fully-qualified domain name (FQDN), e.g. mypbx.dyndns.org. You're going to need it for a whole host of things with your new PBX, and dyndns.org is about the easiest way to do it. Once you have your FQDN and DynDNS username and password, log in as root and edit: /etc/ddclient/ddclient.conf. Search (Ctl-W) for ***. Fill in your username and password and uncomment those two lines. Then search for *** again, uncomment the next three lines and fill in your fully-qualified domain name. Save the file and service ddclient restart. To make sure everything worked, issue the following command: ddclient -force. Assuming there are no errors, issue the following command to start ddclient each time your server reboots: /sbin/chkconfig --add ddclient. Now the IP address of your Asterisk server will always resolve to your FQDN from DynDNS. And anyone can call you via SIP for free using the following SIP URI: mothership@yourFQDN.dyndns.org. You can take this a step further and sign up for a free incoming phone number at ipkall.com. For your account type, choose SIP. For your SIP phone number, enter: mothership. For your SIP proxy, enter the fully-qualified domain name (FQDN) for your server, e.g. mypbx.dyndns.org. Choose a password and enter your real email address, and they will beam you a Washington state phone number within a day or so. You can't beat the price!
Adding Plain Old Phones. Before your new PBX will be of much use, you're going to need something to make and receive calls, i.e. a telephone. For today, you've got several choices: a POTS phone, a softphone, or a SIP phone (highly recommended). Option #1 and the best home solution is to use a Plain Old Telephone or your favorite cordless phone set (with 8-10 extensions) if you purchase a little device (the size of a pack of cigs) known as an SPA-2102. It's under $70. Be sure you specify that you want an unlocked device, meaning it doesn't force you to use a particular service provider. Once you get it, plug the device into your LAN, and then plug your phone instrument into the SPA-2102. Note that this adapter supports two-line cordless phones! Your router will hand out a private IP address for the SPA-2102 to talk on your network. You'll need the IP address of the SPA-2102 in order to configure it to work with Asterisk. After you connect the device to your network and a phone to the device, pick up the phone and dial ****. At the voice prompt, dial 110#. The device will tell you its DHCP-assigned IP address. Write it down and then access the configuration utility by pointing your web browser to that IP address.
Once the configuration utility displays in your web browser, click Admin Login and then Advanced in the upper right corner of the web page. When the page reloads, click the Line1 tab and then repeat this drill for the Line2 tab if you want to connect the device to two extensions on your Asterisk system. Scroll down the screen to the Proxy field in the Proxy and Registration section of the form. Type in the private IP address of your Asterisk system which you wrote down previously. Be sure the Register field is set to Yes and then move to the Subscriber Information section of the form. Assuming you're using the preconfigured extensions starting with 701, do the following. Enter House Phone as the Display Name. Enter 701 as the User ID. Enter your actual password for this extension in the Password field, and set Use Auth ID to No. Click the Submit All Changes button and wait for your Sipura to reset. In the Line 1 Status section of the Info tab, your device should show that it's Registered. You're done. Now repeat the drill for Line2 using extension 702. Pick up a phone and dial 1234# to test out BOTH extensions.
Downloading a Free Softphone. Unless you already have an IP phone, the easiest way to get started and make sure everything is working is to install an IP softphone. You can download a softphone for Windows, Mac, or Linux from CounterPath. Or download the pulver.Communicator. Here's another great SIP/IAX softphone for all platforms that's great, too, and it requires no installation: Zoiper 2.0 (formerly IDEfisk). All are free! Just install and then configure with the IP address of your PBX in a Flash server. For username and password, use one of the extension numbers and passwords which you set up with FreePBX. Once you make a few test calls, don't waste any more time. Buy a decent SIP telephone. We think the best phone out there is the Aastra 57i for under $200. Another $100 buys you the Aastra 57i CT with a cordless DECT phone.
Configuring Aastra 57i SIP Phones. Your new system comes preconfigured to automatically configure up to 15 Aastra 57i phones. Plug each phone into your network and wait for it to boot. Once it boots, press the Option button, then Phone Status (3), then IP & MAC Address (1). Write down each phone's IP address and MAC address. Then press Done to exit from the menus.
Next, we need to tell your phone to use your new Asterisk server as the TFTP server to obtain its setup. Press the Option button again, then Admin Menu (5). Type 22222 for the admin password and press Enter. Then choose Config Server (1), then TFTP Settings (2), then Primary TFTP (1), enter the IP address of your new server, and press Done a half dozen times.
Log back into your server as root. Switch to the TFTP directory: cd /tftpboot. You'll notice that there are config files for up to 15 phones. Simply choose the extension number you wish to use for each phone AND rename each file (filenames are 701.cfg to 715.cfg) to the MAC address of each phone.cfg. Do NOT use hyphens in the MAC address. One final step and you'll be ready to load up your phones. We need to set the correct IP address to tell each phone where your server is located. So... issue the following command using the IP address of your new server instead of 192.168.0.123. Leave the rest of the command as it is!
sed -i 's|192.168.0.0|192.168.0.123|g' /tftpboot/aastra.cfg
Now restart each phone by pressing the Option button and then Restart Phone (6) and then the Restart button. Once the phone reboots, you can make a test call by dialing 1-2-3-4. You can get the latest news by dialing 5-1-1. Or get a weather forecast by airport code (6-1-1) or zip code (Z-I-P).
A Word About Ports. For the techies out there that want to configure remote telephones or link to a server in another town, you'll need to know the ports to remap to your new server from your firewall. Here's a list of the ports available and used by PBX in a Flash. We don't recommend exposing UDP 5038 which is used to communicate with Asterisk via the Asterisk Manager.
TCP 80 - HTTP (needed to access the web sites on your server from the Internet... not recommended!!!)
TCP 22 - SSH (needed if you want remote SSH access)
TCP 9001 - WebMin (needed if you want remote WebMin access... not recommended!!!)
UDP 10000-62000 - RTP (needed for SIP communications)
UDP 5004-5037 - SIP (ditto)
UDP 5039-5082 - SIP (ditto)
UDP 4569 - IAX2 (needed for IAX communications typically between Asterisk servers)
Setting Up Trunks for Outgoing and Incoming Calls. If you want to communicate with the rest of the telephones in the world, then you'll need a way to route outbound calls (terminations) to their destination. And you'll need a phone number (DIDs) so that folks can call you. Unlike the Ma Bell world, you need not rely upon the same provider for both. And nothing prevents you from having multiple outbound and incoming trunks to your new PBX. At a minimum, however, you do need one outbound trunk and one inbound phone number unless you're merely planning to talk to other extensions set up on your system. We've actually put all the hooks in place to make it easy for you to interconnect to other Asterisk servers, but we'll save that for another day. For today, we want to get you a functioning system so that you can place outbound calls to anywhere in the world and can receive incoming calls from anywhere in the world.
For outbound calling, we recommend you establish accounts with several providers. We've included the necessary setups for our own service as well as Vitelity and AOL. To register for our service, just dial any 10-digit phone number from a phone on your system before you set up any other trunks. We're one of the least expensive providers, but you know the old saying about that. Give us a try and, if you don't like the call quality, do some more shopping. We think it's pretty good quality actually, but we don't sell DIDs for inbound service... yet.
Vitelity: One of the Best Providers on the Planet. If you're seeking the best flexibility in choosing an area code and phone number plus reasonable entry level pricing plus high quality calls, then Vitelity is a winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. For PBX in a Flash users, sign up before October 15, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. You can't beat the price (except with us) and the call quality is excellent as well. We've tried just about everybody.
To sweeten the pot a bit more, we've preconfigured both inbound and outbound Vitelity trunks for you. For the vitel-inbound trunk, all you'll need to do is plug in your username, password, and host assigned by Vitelity and adjust the registration string to match your assigned username and password. In FreePBX, click Setup, Trunks, SIP/vitel-inbound and make the changes. Then adjust the vitel-outbound trunk to reflect your actual username in the fromuser and username entries, your real password in the secret entry, and the correct host provided by Vitelity for your outbound calls, and you're all set. In FreePBX, click Setup, Trunks, SIP/vitel-outbound and make the changes. The same setup drill will get you going the the PIAF VoIP service as well, and you have your choice of the following POPs: Houston, Dallas, LAX, NYC, London, Montreal, and Toronto. The POP addresses are entered in the following format: sip.lax.pbxinaflash.net or sip.london.pbxinaflash.net.
To test things out, pick up a phone configured on your system and dial an area code and number of someone in the United States or Canada. Now get someone to call you using your new number. Presto! You have inbound and outbound phone service. And, if you'd like to see just how good SIP service can be, pick up a phone on your system and dial D-E-M-O. This will connect you to the PBX in a Flash hosted demo applications server at Aretta Communications.
An Alternate Outbound Calling Solution. As we said, it costs you almost nothing to add an alternate outbound calling solution to your new system. As luck would have it, adding a third outbound calling provider is now a breeze because AOL just entered the SIP terminations market with a product called AIM Call Out. We wrote about it recently, and you can read the article here. All you need is an AOL or AIM account name and $5 to get you started. The system you've just installed is preconfigured to use AIM Call Out. All you have to do is plug in your username and password, and you can immediately make calls to anywhere in the United States for under 2¢ per minute. Adding international calling is as easy as inserting the correct dial string. If you never use it, it doesn't cost you a dime. So $5 is mighty cheap insurance in our book.
First things first. Sign up for the service at this link. Your username will look something like this: johndoe@aim.com. You also will be assigned a password. Using your web browser, open FreePBX by pointing to the IP address of your new server and choosing Administration, then FreePBX. Type in admin as your username and the password you assigned to your system. From the main FreePBX menu, choose Setup, Trunks, and click on SIP/AIM in the far right column. Scroll down to the Peer Details section of the form and replace yourAIMpassword with your new password. Then replace yourAIMaccountname with your actual AIM account name. Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.
Setting Up an Alternate DID for Incoming Calls. You also may want to consider a second phone number where people can call you. For example, if Grandma and Grandpa happen to be in another state and still have an old fashioned telephone, you might consider adding an additional DID to your system in their area code. They then can make a local call to reach you by dialing the local DID. On the les.net pay-as-you-go plan, it costs less than a dollar a month plus a penny a minute for the calls. Money well spent if we do say so... and you'll sleep better.
If this setup looks a bit complicated, don't be intimidated. Remember, we're connecting your PBX to the rest of the world so people can call you! With les.net, you have a choice of rate plans for most DIDs. You either can pay $3.99 a month for unlimited inbound calls with two concurrent channels or 99¢ per month and 1.1¢ per minute with four concurrent channels. Just visit their site and click Signup to register. Once you are registered, click Login and then Order DIDs. Pick a phone number. Then click Peers/Trunks and Create New Peer. Write down the Peer Name as you will need it in a minute to set up your connection. Choose SIP for Peer Technology, RFC2833 for DTMF Mode, G.711 for Codecs, Registration for Peer Type, enter the public IP address of your server for Peer Address, make up a secure password and write it down also, specify an Outbound CallerID for your calls, and check the 10-digit dialing box. Leave voicemail unchecked since you'll handle this on your end. Save your changes.
Now choose Your DIDs and click on the one you just ordered. We now need to tie the phone number to the Peer setup you just created above. Click on the DID and select the Route to Peer which you just created. Check the Send DID Prefix box and leave everything else blank. Click Save Changes and you're finished at the les.net end. Now let's set up your inbound DID trunk in Asterisk using FreePBX.
Log into FreePBX using a web browser. Click Setup, Trunks and then Add SIP Trunk. Fill in the CallerID and then drop down to the Outgoing Settings section of the form. For Trunk Name, use the Peer Name that you created above and wrote down. It ought to look something like this: 1092832198. For Peer Details, enter the following using the Peer Name and Password you assigned at les.net:
canreinvite=no
context=from-trunk
fromuser=1092832198
host=did.voip.les.net
insecure=port,invite
nat=yes
secret=yourpassword
type=peer
username=1092832198
For Incoming Settings, use from-pstn for the User Context and enter the following User Details:
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
insecure=port,invite
nat=yes
type=user
For the registration string, enter a string like the following using your Peer Name and Password:
1092832198:yourpassword@did.voip.les.net/1092832198
Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.
Choosing a VoIP Provider That Supports Faxing. We've included a reliable fax solution in this build, and we'll cover all the details next week. We do want to give you a head start if you plan to use your new machine to handle inbound faxes. To test your machine, you can connect a real fax machine to one of the lines on an SPA-2102. Then send a fax to extension 329 (F-A-X). But first you must configure your email address in two places using FreePBX: Setup, General Settings, Email address to have faxes emailed to AND Setup, Inbound Routes, any DID / any CID, fax Email. Once you've saved your settings, send the fax and see if it's delivered to your email address. If it works reliably, then the fax and email applications on your machine are configured correctly. Unfortunately, that's only half the battle. To receive faxes from outside your system, you'll also need a DID from a provider that supports faxing. And then it's still only about a 90% proposition... on a good day. We've tested this with many, many VoIP providers. Some work. Many don't. Some, such as Vitelity, offer a faxing service for a fee. Guess what? Their regular VoIP setup doesn't support faxing. Our old friends at Telasip.com still support faxing. We've also had good luck with Future-Nine and Teliax. You can read the beginnings of our fax dissertation here for more details. With the exception of the trunk setup covered in the article, all of the remaining setup steps already have been completed on your new server!
Choosing a Preferred Provider. Finally, you'll need to decide whether to use us or AOL or Vitelity as your primary terminations provider. HINT: We're the cheapest! So we've set things up with us and then AOL. This is handled in FreePBX in the Outbound Routes tab under the Default entry. You can adjust easily these in any way you like by adding trunks or moving entries up and down the list to change their priority. Just be sure to leave ENUM at the top of the list since ENUM calls are always free. If a free call isn't possible, your server will automatically drop down to the next trunk in the priority list. Don't add Vitelity to the list unless you have actually created a Vitelity account since they handle unsuccessful connections in a non-standard way which will cause FreePBX not to drop down to the next trunk to attempt a connection.
A Word About Mondo Rescue. We would be remiss if we didn't mention what a fantastic open source product Mondo Rescue is. It's the sole reason that today's build was possible. Our special thanks go to the development team: Bruno Cornec, Andree Leidenfrost, and Hugo Rabson. It is the first (and only) backup software for Linux builds that actually works reliably. The best way to prove that for yourself is to download this build and try it for yourself on your Everex gPC2. It has much more flexibility than what you will experience, but that would take another dozen pages to explain. We'll save that for another day. In the meantime, if you'd like more information, visit the Mondo Rescue web site.
Where To Go From Here. Well, we've covered a good bit of territory today so we're going to save the really fun stuff for our next installment. In the meantime, you have a new phone system that works. And there are a number of PDF documents in the /root folder on your new system which are worth a read. Better yet, you can browse through all of the documentation which is available for PBX in a Flash by going here. You also can dial D-E-M-O on your new system and see just how powerful direct SIP connections can be to other Asterisk hosts (in this case, ours!)... at no cost. Finally, you can log into your server and type help-pbx for access to a treasure trove of additional features. Enjoy!
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...
Using Asterisk and Gizmo5 to Transform Your Nokia N95 Cellphone into the Ultimate Free SIP Phone
We're wrapping up our Gizmo5 series with what we believe is the real silver lining in the Gizmo Project. Here's our YouTube demo to prove it. We kicked things off by showing you how to set up a Gizmo5 account to make free calls with Asterisk® using Backdoor Dialing and ENUM. Then last week we added an Asterisk dialplan script to actually test whether an outbound call would be free through Gizmo5 before choosing a final route to terminate outgoing calls from your Asterisk server. Today we're going to use your Gizmo5 account to transform a standard Nokia N95 cellphone into a SIM-free, carrier-free WiFi SIP telephone which can perform a whole bag of tricks at absolutely no cost... once you own the unlocked phone. It's the perfect back-to-college gift if your wallet matters. Without too much hype, suffice it to say the N95 earned PC Magazine's Editor's Choice. In a word, the call quality is phenomenal. If you like Snickers candy bars, then you'll love the size of this phone. And WiFi works all day with Nokia's Symbian OS even though you're lucky to make it an hour with Windows Mobile devices. Maybe it's not the WiFi that's the problem after all, Bill. Ooops. He retired. Sorry. Anyway, any unlocked Nokia N95 will do. Just not the N96! And you'd better get one while they last. Nokia apparently has had a change of heart on SIP telephony support, and it's quickly disappearing from their newer models. Dumb move!
SPECIAL NOTE: We have one, gently used N95 for sale. It actually was used to prepare this article. Make us an offer, or we'll make you a deal you can't refuse. If you're interested, contact us.
When we're finished, you'll have a better appreciation for why AT&T and the other cellphone carriers hate Nokia phones and why Comcast would prefer to limit your bandwidth and charge you $40 (extra) per month for their VoIP service and boatloads more for their pay-per-view movies. This isn't about greedy bandwidth abusers. It's about a greedy service provider. Comcast could easily rein in bandwidth abusers with a letter threatening to terminate service. What they can't control with nastygrams are the Blockbuster's and Walmart's of the world that want to deliver pay-per-view movies to your doorstep via the Internet. So this looks more like restraint of trade to us than protection of scarce resources from the Napster generation as Comcast would have you believe. You don't make the whole class stay after school because one kid chewed gum... if your motives are as pure as the Comcast TV barons would have you believe. Now where were we? </rant>
For openers today, you can place SIP calls at no cost to any SIP phone or Asterisk server in the world. See our previous tutorial to learn how to set this up on your Asterisk server. Second, you can place "regular" phone calls to any phone in the world using a Gizmo5 account at Gizmo5's discounted calling rates (2¢ a minute or less for calls to U.S., U.K., most of Europe, China, and Australia for example) rather than cellphone subscription plus stratospheric long distance charges. Third, you can place free calls to almost every non-AT&T cellphone in the U.S. by dialing 0101 and the number. Fourth, you can place free calls back to and through your Asterisk server to just about anywhere on the globe (except resort areas surrounded by water) for almost nothing. And, finally, you can receive free calls on your N95 cellphone whenever anyone dials your free IPkall-assigned DID in the Seattle area or your SIP number through Gizmo5. And, did we mention that all of this magic occurs with no connection to AT&T or any other cellphone carrier. In fact, we don't even have a SIM chip in our N95. Well... not all the time anyway. All you really need is a WiFi connection to make all of this work. And even the Asterisk server is optional. So let's get started.
Enabling WiFi on Your Nokia N95. Before we can use the N95 as a free SIP telephone, we've first got to get a WiFi connection enabled on the phone. Pressing Menu, Tools, WLAN Wizard will get you started. You can test your connection by opening the web browser for a trial run after you have your WiFi connection set up. Once it's working, be sure to disable the WiFi Access Point scanning feature by choosing Menu, Tools, Settings, Connection, Wireless LAN and set Show WLAN availability to Never.
Installing the Gizmo5 Application. Now the tricky part, and it's really not that difficult. It just happens that there's lots of conflicting information posted around the web, and this makes the drill more confusing than it needs to be. First, if you already have automatic registration of your Gizmo5 account on another device or an Asterisk server, disable the automatic registration. You can't have the same account registered in two places simultaneously. Just open a second account if you need it. There are two components that need to be installed on your Nokia N95, and they're in different places. First, install Nokia Internet Services Support Package to the device memory (not to the memory card). Here's Nokia's download link. Next, install the Gizmo5-Nokia PlugIn from gizmovoip.com. Here's the download link for that one. Finally, we had one little gotcha with getting everything to work once it was installed. On your phone go to Menu, Tools, Settings, Connection, SIP Settings, Options, Edit SIP Profile and set the Service Profile to Nokia 3GPP. Next, go to Menu, Tools, Internet Tel and activate Gizmo after choosing your default WiFi network. You'll be prompted for your Gizmo account name and password. Once it's registered, you should be able to dial 0101 plus an area code and phone number to test out the free calling feature. Or you can dial an area code and number, and route your outbound call as a pay-by-the-minute Gizmo5 Internet Call under the Options button. To call a sip phone directly, simply create a new Contact and insert an Internet telephone entry in the SIP URI format: sipname@FQDN.com. Once you have saved the entry, simply choose it from your Contacts to place the free SIP call. In Nokia-speak, it's referred to as an Internet Call.
If the above procedure doesn't work for you, repeat the drill and set the Service Profile to IETF instead of Nokia 3GPP. Not sure why but one setting works some of the time, and the other one works the rest of the time. If you can't connect, this is usually the problem... assuming you've gotten your Gizmo5 username and password entered correctly.
You also can use your Asterisk server to forward outbound SIP calls from your N95 to other phones. For example, if there are 10 close friends that you call frequently, assign each of them a SIP URI on your Asterisk server. We covered the setup process in this article. In a nutshell, create an Incoming Route in FreePBX named tom and point it to the phone number you wish to call. For destination phones outside your PBX, first create a Miscellaneous Destination called Tom-home that includes the home phone number. Then use this destination in your Incoming Route for tom. Save your entry and reload your FreePBX dialplan. Finally get your own fully-qualified domain name from a service such as dyndns.org. Assuming your FQDN was pbx.dyndns.org, then your Internet telephone entry for Tom in your N95 contacts would be this SIP URI: tom@pbx.dyndns.org. Otherwise, you'll need a SIP URI with the IP address of your Asterisk server, e.g. tom@36.24.36.1.
Adding a Free DID for Inbound Calling to Your Nokia N95. One of the world's best kept secrets continues to be the availability of free DIDs from ipkall.com in Seattle. This saves you $35 a year over the current Gizmo5 DID rate, and IPkall will give you a free phone number in one of several available area codes to use with the SIP device of your choice. Your Nokia N95 qualifies! Just be sure to place at least one call a month to the number, or it's automatically recycled to someone else. To register for a free IPkall account, go to this link and sign up. Your SIP Phone Number is your 11-digit Gizmo5 phone number starting with a 1. Your SIP Proxy for Gizmo5 is proxy01.sipphone.com. Now plug in a valid email address and create a password for your account. Your new phone number will be delivered to this email address. Once it arrives, you should be able to dial the number from any phone, and your Nokia N95 should start ringing. Answer the call just as you would any other cellphone call. The only difference is that you can talk as long as you like... for free. For other free DIDs and some great tips including ATA setup, go here.
Using Asterisk to Add the Missing Pieces. There are a number of ways you can use Asterisk to enhance your SIM-free Nokia experience. By enabling DISA, you can place a SIP call to your Asterisk server, obtain dial tone, and call anywhere using your existing Asterisk trunks. Here's the way we set this up. Edit /etc/asterisk/extensions_custom.conf and add a [custom-disa] context at the end of the file that looks like the following code. Be sure to set a VERY secure password in line s,7 by replacing 1234. It's your phone bill! Then set your IPkall DID number as the CallerID in s,13. By changing 701 in s,12 you can call any extension on your Asterisk server just by dialing 0 when you're using DISA. For our foreign friends, be sure to adjust the dial string length (10) in s,9 to meet your local needs.
[custom-disa]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Set(TIMEOUT(digit)=7)
exten => s,4,Set(TIMEOUT(response)=10)
exten => s,5,Background(enter-password)
exten => s,6,Read(MYCODE,beep,7)
exten => s,7,GotoIf($["${MYCODE}" = "1234"]?8:15)
exten => s,8,Set(TIMEOUT(absolute)=9000)
exten => s,9,Read(NUM2CALL,pls-entr-num-uwish2-call,10)
exten => s,10,Playback(pls-wait-connect-call)
exten => s,11,GotoIf($["${NUM2CALL}" = "0"]?12:13)
exten => s,12,Dial(Local/701@from-internal)
exten => s,13,Set(CALLERID(number)=4251234567)
exten => s,14,Goto(outbound-allroutes,${NUM2CALL},1)
exten => s,15,Hangup
Next, add an Incoming Route using FreePBX. For the DID Number, use a SIP name that is not easily guessed, e.g. DISA2375. This gives you an extra layer of password protection since anyone can try to guess your SIP URI's once they know the IP address of your Asterisk server. Leave all of the other entries at their defaults and, for the Destination, choose Custom Route: custom-disa,s,1. Save your settings and reload your dialplan. Ignore the warning that you're doing something odd. We know what we're doing.
Finally, on your Nokia N95, add a new Contact called DISA with an Internet telephone number to match the name of your incoming route above with the fully-qualified domain name of your Asterisk server, e.g. DISA2375@pbx.dyndns.org. Now you're ready to dial away by simply selecting this contact on your N95. Enter your DISA password when prompted and then enter a 10-digit phone number to call.
The WiFi HotSpot Two-Step (and a few more steps). Now that everything is working swimmingly, we're ready to take your Nokia N95 on the road. Here's the failsafe step-by-step to get connected in a WiFi HotSpot of your choice.
- Turn off the phone
- Arrive at HotSpot
- Turn on the phone
- Menu, Tools, WLAN Wiz., Pick Your HotSpot, and Start Web Browsing, Create WLAN While OffLine=Yes
- Using Browser, log into the HotSpot with your account name and password
- Leave the browser open so it'll be easy to log out when you're finished
- Menu, Tools, Internet Tel., Pick Your HotSpot AP
- Once Connected, Dial Away As Usual
- When finished, Hold Down Menu Button and Choose Browser App, Log Out of the HotSpot
- Turn off the phone
Cellphone Options. But what if you really do want to use the Nokia N95 in all its glory with the 5 megapixel camera and the multimedia goodies and even a cellphone provider? Well, it works great for just about anything you need. In fact, you can even take the SIM chip from your iPhone (even a First Generation iPhone) and plug it in. Phone calls work, voicemail works (even though you get two text messages when a new voicemail arrives... which is a lot better than Cingular in the old days when you typically got zero), and email and web browsing work great, too. Just select MediaNet as the access point when you open your Internet connection, and you'll be off to the races. Of course, all the cheapo, pay-as-you-go SIM cards work as well. Both Oxygen and Airvoice packages including free minutes and a SIM card are available at this eBay store for under $10. And there are lots of other options as well. Enjoy!
VPN in a Flash Update! We've had over 100 reservations for our new VPN in a Flash system. We're very close to having a manufacturer in place so hopefully we'll have more good news in a week or two. We have begun the documentation for the new product, and we encourage you to take a look and offer any questions or comments you may have on our forums. The documentation is in the new Google Knol format and can be reviewed here. It's not too late to get in the queue and place a reservation for a system. Just send us a note, and we'll keep you posted as the release date approaches. It'll hold your place in line with absolutely no obligation to purchase.
Coming Attractions. We're very close to signing on a new VoIP provider for PBX in a Flash users that will provide penny-a-minute calls in the U.S. and Canada. And a new version of AsteriDex with Outlook synchronization and a TTS dialer for AsteriDex queries from any connected Asterisk phone is just around the corner. Stay tuned!
Hosting Provider Mega Deal. 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.
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...
Free Asterisk Calls to Zillions of Phones with ENUM and Gizmo5’s Backdoor Dialing
It’s been a while since there’s been much to cheer about in the free calls department with Asterisk®. But today, to kick off the new school year, we have lots of good news and some simple tricks to add zillions of free phone numbers to your Asterisk repertoire. In fact, you’ll be able to call almost any non-AT&T cellphone or landline in the United States at no cost. Remember that when you buy your next cellphone! Special thanks to Cliff on the PBX in a Flash Forums for heads up.
Some early readers of Nerd Vittles may remember sipphone.com which morphed into Gizmo5.com. In January of this year, Gizmo5 struck peering deals with a number of telephone providers that already routed their calls over the Internet. And it’s a pretty impressive list that includes more than 10% of the phones and cellphones in the United States according to Gizmo5’s bean counters. There’s Access One, Airadigm, Allegiance, Alltel, Cablevision Lightpath, Cat Communications, Cbeyond, Cellcom, Cellular Properties, Centennial Wireless, Choice One, Cincinnati Bell Wireless, Cinergy Communications, Cingular, CityNet, Cleveland Unlimited, Comcast Digital Voice, Commpartners, Conversent Communications, Cox Communications, CP Telecom, CTC Communications, Dobson Cell, Eureka, Globalcom, Heartland Communications, Illinois Valley, ITC Deltacom, LDMI, McLeod, Metro PCS, Mpower, Nationsline, Nextel, Nextera Communications, Paetec, RCN, Sprint PCS, Talk America, Telnet Worldwide, T-Mobile, US Cellular, Verizon Wireless, and XO. Whew! And the program is constantly being expanded. Toll-free numbers and Gizmo5-to-Gizmo5 calls also are free using Gizmo5. You can check whether your frequently called numbers are free calls by simply entering the phone numbers at this link.
Thus was born what Gizmo5 calls Backdoor Dialing. Just dial 0101 and the 10-digit number of your choice. If it’s free, the call goes through. If not, you get a message that the number is not yet supported and click. The beauty of the program is that your total investment to use the free service with Asterisk is a one-time fee of $10 for a bucket of CallOut minutes to activate your account. Sometimes this takes a day for the credit to appear, particularly if you use PayPal to cover the cost. The good news is you can spend most of the $10 making calls to any phone in the world, many for under 2¢ per minute, using just about any computer on the planet. Just leave a few cents in the pot to keep your free Backdoor Dialing service enabled. From our testing, we’d rate the Gizmo5 call quality as excellent on both the free and the pay-per-minute calls! Complete rate tables are available here.
Gizmo5 provides free softphones for Windows, Macs, and Linux as well as numerous cell phones and mobile devices including Treo, Nokia, and many more (not the iPhone… yet!). All of the softphones make it extremely easy to place SIP calls, e.g. joeschmo@mypbx.dyndns.org. And you can place these calls all day long at no cost. See our tutorial for step-by-step instructions on setting up your own SIP addresses on your Asterisk server. The softphones also include Conferencing, SMS, and Instant Messaging with AIM, Yahoo, MSN, Google, and MySpace.
As with many of these services, they weren’t designed for Asterisk, but nothing in their fine print precludes Asterisk use so today we’ll show you how. Will the program last forever? Who knows, but it’s free for now. And the cost of admission is too good to resist. You’re obviously not going to dial every number you frequently call twice just to see if the call is free. That’s why you’ll want to use a robodialer such as AsteriDex for your outbound calling. Then it’s easy to adjust the phone numbers of your friends with Sprint, T-Mobile, or Verizon cellphones so that you never have to pay for those calls again. Just add a prefix of 0101 to the numbers, and you’re done. And they can call you on your Gizmo5 CallIn number through Asterisk if you’ve enabled the CallIn Service and chosen a number. It’s under $3 a month with an annual subscription. Or the calls can be returned using the CallerID number displayed by Gizmo5 when you call your friends. Toll charges may apply in this case due to the Gizmo5 area code.
So let’s get started. Step 1 is to download and install a free softphone of your choice and follow the prompts to sign up for your account. There’s really no reason not to install a Gizmo5 softphone on every computer you own. If you don’t use it, there’s no cost. If you ever need it, it’ll be there for you. Step 2 is to make a $10 purchase of CallOut minutes. While you’re waiting on the credit to appear (and it usually takes less than a day), let’s set up Asterisk. You’ll need your new account name, password, and phone number from Gizmo5 to get started.
Setting Up a FreePBX Trunk for Gizmo5. If you’re using a product such as PBX in a Flash that includes FreePBX, then open FreePBX in your browser and choose Setup->Trunks->Add SIP Trunk. Leave the General Settings blank. For the Dialing Rules, if you just want free calling through your Gizmo5 trunk, plug in values below. For regular calls as well, add 1NXXNXXXXXX or an entry that is suitable for each country you wish to call.
1800NXXXXXX
1822NXXXXXX
1833NXXXXXX
1844NXXXXXX
1855NXXXXXX
1866NXXXXXX
1877NXXXXXX
1888NXXXXXX
800NXXXXXX
822NXXXXXX
833NXXXXXX
844NXXXXXX
855NXXXXXX
866NXXXXXX
877NXXXXXX
888NXXXXXX
0101+NXXNXXXXXX
0101NXXNXXXXXX
Name the Trunk: Gizmo5. Make the following entries in Outgoing Settings Peer Details:
disallow=all
allow=ulaw
auth=md5
authuser=youracctnameNOTyourphonenumber
canreinvite=no
context=from-trunk
dtmfmode=auto
fromdomain=proxy01.sipphone.com
fromuser=youracctnameNOTyourphonenumber
host=proxy01.sipphone.com
insecure=very
nat=yes
qualify=yes
secret=yourpassword
type=peer
username=youracctnameNOTyourphonenumber
Clear out the Incoming Settings and use the following syntax for the Registration String. Then Save your setup and Reload Your Dialplan. NOTE: Don’t use any registration string unless you want incoming call support. By not registering, you can use your softphones whenever you need it to also make outbound calls. If you register with Gizmo5 using a registration string, then it knocks out use of a softphone since you can’t have two simultaneous registrations to the same account. But registering allows those you call with this service to call you back conveniently… although not necessarily for free from the caller’s phone.
youracctname:yourpassword@proxy01.sipphone.com/yourphonenumber
Setting Up a FreePBX Outbound Route for Gizmo5. While still in FreePBX, choose Setup->Outbound Routes->Add Route. Name the route: OutGizmo5. Then enter the following Dial Pattern: 0101NXXNXXXXXX. Choose SIP/Gizmo5 as your Trunk Sequence. Then click Submit Changes and Reload Your Dialplan.
Setting Up a FreePBX Inbound Route for Gizmo5. While still in FreePBX, choose Setup->Inbound Routes->Add Incoming Route. Name the route: Gizmo5 and plug in your 10-digit DID number in the appropriate field. Then Set a Destination for the incoming calls. That’s it. Save your entries by clicking the Submit button and then Reload Your Dialplan.
Making a Free Call with Gizmo5. Once your DialOut credit appears on your softphone or in your Gizmo5 web account, you’re ready to start making calls. From any phone connected to your Asterisk server, just dial 0101 plus the 10-digit phone number. On the Asterisk CLI, you should see the call routed out through your SIP/Gizmo5 trunk. If you get a congestion tone and you’re sure your DialOut credit has been posted to your account, then check your username and password entries in your Trunk setup. Be sure to use your account name and NOT your Gizmo5 phone number for your username, authuser, and fromuser entries. But, if that doesn’t work, try using your Gizmo5 phone number instead of your assigned user name. Some have reported quirks in which actually works. For us, the assigned user name did the trick. Also make certain that the disallow all entry is above the allow=ulaw in versions of FreePBX after 2.3, or no calls will ever be successful.
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.
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.
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…
The Digium Conundrum: Will Asterisk Be Just Another Asterisk on the VoIP Radar
It’s been several months since we last addressed the "Asterisk® Problem" and much has been written and spoken on the subject since then. In a nutshell, the problem is the code changes made in each new version of Asterisk which break existing business applications. We’ve come to a better appreciation of the point of view of some of the Asterisk developers. But I’m sorry to say they haven’t budged. The good news is that much has changed for the better in spite of the Asterisk developers. And today we wanted to share some of those developments with you.
The Developer Mentality. Suffice it to say that the Asterisk development community is quite small and mostly driven (like most programmers) by a fierce sense of independence. The puzzling part is that most of these guys (and it is an all-male club) work for companies that make their living in the Asterisk marketplace, either manufacturing or selling hardware for Asterisk-based telephony systems. For the most part, however, these companies are hardware peddlers rather than system integrators. One fact of life has become crystal clear. New versions and new beta releases NEVER break existing hardware. Why? Because these are the companies that feed these guys. "Whose bread I eat, his song I sing" goes the old adage. So hardware that was purchased in the Asterisk 1.2 days still works equally well with the latest 1.6 beta releases. Thank you very much.
Business application software for Asterisk is an altogether different equation. Here the developer mantra goes something like this. You’re using our code for free, and we’ll improve it in any way we think is best. If it breaks your application code, too damn bad! You can either fix it, stop using it, or go elsewhere. And we really don’t care which option you choose. The sad part of that mentality is a total lack of appreciation for the fact that, once demand for Asterisk systems in the business community dries up, the demand for Asterisk hardware will also take a nosedive.
The types of business applications that have been broken are major, not organization-specific. For example, Asterisk 1.4 broke the open source fax application. And Asterisk 1.6-beta broke both the open source and commercial text-to-speech (TTS) engines. The sad part is that the applications were broken by trivial code changes in Asterisk that just as easily could have been accomplished without breaking any application code.
This development approach, of course, keeps Asterisk out of most major corporations and government organizations even though it is an almost perfect fit for many of them. Why? It’s pretty simple. Business application software in most major organizations isn’t written in house. It’s developed by outside contractors who typically bid on a project, win the bid, develop the software application, and move on. Three to five years later, they usually are not around to rebuild something that the Asterisk developers have broken with their "improvements." Since phone systems usually are measured in decades and Asterisk releases are measured in years, it’s a pretty terrible fit for most major corporations and government organizations. Can you imagine a WalMart or a Hilton Hotel replacing their telephony applications every couple of years because all of their fax capability suddenly vanished? The same is equally true in the medical and legal communities as well as in major real estate and construction companies. Earth-to-Digium®: Companies have more to do than babysit their phone systems.
What we said four months ago is equally true today. When we began the PBX in a Flash project last November, our emphasis was radically different than some of the other Asterisk aggregations. First and foremost, we wanted a product that was stable. Of equal importance was our own Big Easy: easy to use, easy to enhance, and easy to upgrade. We didn’t want users or VARs having to reinvent the wheel each time a security patch or new enhancement was released. To look at it from the customer side, no business (that wants to stay in business) will tolerate a phone system that is routinely out of service for upgrades much less one that takes away features that the business depends upon. Whether it’s Caller ID, or Text-to-Speech, or Screen Pops, or Conferencing, or Phone Blasting, or even a Call Center really doesn’t matter. It does no good to tell a customer that they lost critical functionality but now they have the latest version of Asterisk. You can add your own customer expletive here if you’ve ever tried this approach in the real world.
In the good old days when there wasn’t much of a feature set and when no business would stake their livelihood on Asterisk, it really didn’t much matter when a new version of Asterisk was released. To put it charitably, things could only get better. But, businesses now rely upon Asterisk. So the dynamics are quite different. It’s no longer acceptable to trash big chunks of code without making certain that you didn’t break something that was already working. It’s no longer acceptable to invent new verbs in the programming language while deleting commands that used to work.
The Good News. There really is a silver lining to this story. There’s a new game in town: FreeSwitch. It may take a year, but this is an all new technology with a team of developers with an all new attitude about software development. This is a product that is being developed from the ground up to meet business needs. It employs modern, business interfaces with which most major organizations are already familiar. Can it do what Asterisk can do? For the 60% of Asterisk functions that already work, FreeSwitch not only gets a check mark but the performance improvement is staggering. And for the 60% of FreeSwitch functions that Asterisk can’t do at all… well, you’ll just have to try it. Give them six to twelve more months, and we predict the trickle of Asterisk defections is going to turn into a stampede. Both a Windows implementation for your desktop and a turnkey Linux install via ISO are now available. What’s still missing is a tool as simple as FreePBX to actually configure everything, but rumor has it that there are several GUI interfaces in the pipeline. And for the short term, nothing could be much simpler than the XML code that makes FreeSwitch tick. Indeed, dozens and dozens of sample XML scripts are already available which mirror most major Asterisk functions and dialplan applications.
And More Good News. The problem with breaking generic business applications is that the developers who initially wrote the open source apps are no longer around or interested in Asterisk. Wonder why? In any case, thanks to Antonio Gallo, open source faxing for Asterisk 1.4 is back. And, thanks to Darren Sessions, open source text-to-speech with Flite for Asterisk 1.4 and 1.6 is a reality once again. Installation on CentOS systems still is a bit hairy. So we will include Flite for both Asterisk 1.4 and 1.6 in our new PBX in a Flash 1.3 release next week. And faxing will return for Asterisk 1.4 in our first SUSHI update shortly thereafter. Enjoy!
Another Good Read: Open Source VoIP: Asterisk or FreeSwitch by David Greenfield
And Another: Asterisk vs. FreeSwitch by Anders Brownworth
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…
Free Asterisk Solutions
