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The Most Versatile VoIP Provider: FREE PORTING
JUST RELEASED: Visit the Incredible PBX WikiTweaking Asterisk for Free Google Voice Calling
Now that the Asterisk® and Google Voice marriage is finally underway, we wanted to step back today and revise the original methodology a bit to take advantage of some of the terrific comments which were offered in response to our last article. First, the good news. U.S. calls through Google Voice using Asterisk work! They sound great, and they're free. The not so good news was that the MeetMe conferencing trick to join your outbound call with the Google Voice click-to-dial return call from your destination worked great so long as a real person answered the phone. But, if an answering machine picked up or no one answered the call at all, there were problems because these calls already had been transferred to the MeetMe conference and there was no simple way to disconnect them. And the need for two DIDs to support a single Google Voice interface just seemed a bit wasteful.
9/1/2010 Update: A good bit has changed with Google Voice since this article was first published. For the definitive guide and installation procedure, we highly recommend The Incredible PBX and accompanying article which can be found at this link. Google Voice (and much more) already is included in our new PBX which is literally Plug-and-Play. If you prefer to roll your own, be sure to also have a look at this excellent update on the Michigan Telephone Blog.
Today we want to try to eliminate these two quirks while stiill providing a seamless interface between Google Voice and Asterisk. We also appreciate that thousands of you already have implemented the previous approach. So we want your transition to the new way of doing things to be as painless as possible. On the other hand, for frequent readers, we hope you'll bear with us as we repeat some of what already has been covered in previous articles so new visitors don't have to jump around between articles to get the complete picture of what we're trying to accomplish.
The objective remains the same. We want a methodology that lets us make outbound calls from any Asterisk phone using the Google Voice service to take advantage of free calling in the United States and Canada. And we want calls to our Google Voice number delivered to our Asterisk system for transparent call processing. Yes, SIP is still on our wish list for both outbound and inbound calls with Google Voice, but we'll make do with PSTN calls particularly while Google is footing the bill for all of the calls.
Update: There's now a turnkey Asterisk solution that implements Google Voice calling without getting your hands dirty. Check out our new Orgasmatron V.
Tweaked Design. Here's the new design. You obviously still need a free Google Voice account. If you don't have one, you can request an invite here. At last report, it's only taking a few days from application to invite which is really great news. Don't use a space in your Google Voice password! Once you have a Google Voice account and phone number (Google has reserved a million of them so... not to worry!), then you'll need a DID that provides unlimited, free incoming calls. Once you get your DID set up on your Asterisk system, we'll set up a forwarding phone number for this DID in your Google Voice account so that Google Voice calls can be connected to your Asterisk server.
For outbound calls, we'll combine a little dialplan voodoo with pygooglevoice to instruct Asterisk to place a click-to-dial call using your Google Voice forwarding number. Then we'll stuff in the destination U.S. phone number. When you dial GV-678-1234567 from any of your Asterisk phones, Asterisk will park your initial call in a reserved parking lot slot and then join the called party to the originally parked call. The entire procedure is virtually transparent both to the caller and the callee. And, unlike the MeetMe conference, the parking lot fades out of the picture as soon as the call is connected. Thus, if either party hangs up, the active channel for the call is terminated on your Asterisk server.
For inbound calls from your Google Voice number, we'll tweak the dialplan so that it can distinguish between a RingBack call that Google Voice initiated and a true inbound call. We'll peel off the real inbound calls and route them to a separate Inbound Route in FreePBX for processing in any way you desire.
Finally, for those that implemented the methodology in our previous article, we'll walk you through the steps to revise your existing setup to take advantage of these new tweaks. You can skip over the initial installation process if you already have gone through the Google Voice setup from our earlier article. Just skip down to Tweaking Previous Setups.
Special Thanks. At the outset, we again want to express our sincere appreciation to Jacob Feisley and Paul Marks for their pioneering work on a Python interface to Google Voice. We also stumbled upon another Python development project, Google Voice for Python. While we originally had planned to rely upon Jacob and Paul's script, we ultimately decided to implement pygooglevoice because of the additional flexibility it provided for down the road. With pygooglevoice, you not only can make Google Voice calls, but you also can send SMS messages with no muss or fuss. Jacob Feisley has now joined that project as well. So, our special tip of the hat goes to the entire Google Voice for Python development team. It's a terrific product as you will see.
Prerequisites. Today's setup requires a CentOS-based Asterisk aggregation with a current version of FreePBX. Be aware that today's solution requires Python 2.4 or higher and reportedly will not work with Python 2.3 found in some Linux distributions. We've tested everything with PBX in a Flash and, on that platform, you're good to go. The install script should work equally well with the other CentOS-based Asterisk aggregations, but we haven't tested them. Be our guest, and let us know if you encounter any problems. Finally, a word of caution. We don't ordinarily distribute solutions using development tools we don't use. Our knowledge of Python wouldn't fill a thimble. We've made an exception today because of the extraordinary interest in Google Voice by the Asterisk community. But, if something comes unglued, we can't fix it. So have a backup plan in place just in case. 🙂
Today's Drill. To get everything working today, there are six steps: (1) obtaining and configuring a DID to manage calls between Google Voice and Asterisk, (2) configuring a Google Voice forwarding number for this DID to manage your outbound and inbound calls, (3) configuring FreePBX to route all outbound calls with a GV prefix to your special Google Voice dialplan context, (4) configuring an inbound route to manage incoming calls from your Google Voice number, (5) setting up a series of Parked Call extensions, one of which will be used to manage your outbound Google Voice calls, and (6) running our install script which adds the dialplan code for Google Voice calling with your credentials and puts the Python application into place on your server. It sounds more complicated than it is. So hang on to your hat. Here we go!
Dedicated DID. Before you can use Google Voice with Asterisk, you'll need a DID that can be dedicated to your Google Voice interface to Asterisk. We'd recommend a free IPkall or SIPgate DID. To get started, use one of the links above to obtain and configure the DID. Temporarily point the DID to an extension on your Asterisk system that can be used to verify your requests for the number. Since all of these calls are free, the area code of the DID really doesn't matter because you're never going to publish the fact that it exists.
The easiest method for setting up the DID is to first create a SIP URI for the DID on your Asterisk system. Next route the SIP URI to an Inbound Route in FreePBX where you can manage the destination for calls to that DID. Initially, you want the destination to be an extension on your Asterisk system that you can answer to verify both the DID setup and the GV setup below. Finally, point the DID you obtained to the SIP URI defined above.
HINT: The entry in extensions_override_freepbx.conf would look something like this for a SIP URI called ipkall-1:
exten => ipkall-1,1,Goto(from-trunk,${DID},1)
Then you would create an inbound route named ipkall-1 using FreePBX and designate some existing extension on your server as the destination for these inbound calls.
When you set up the SIP forwarding for the DID at ipkall.com, you'd specify the SIP URI as:
ipkall-1@ipaddress_of_your-Asterisk_server
We've previously covered in detail how to do this so read the article if you need a refresher course. To reiterate, the area code of this DID really doesn't matter because you're never going to give out the number. So use one of the free sources and save yourself some money. The real trick is you want to use a DID with unlimited, free inbound calls. Both IPkall and SIPgate provide that functionality at no cost.
Google Voice Setup. Log into your Google Voice account and click Settings, Phones, Add Another Phone. Add the area code and phone number of your DID. Be sure the DID is pointed to an extension on your PBX that you can answer since you have to go through Google's confirmation drill to successfully register the number. After the DID is confirmed, be sure there's a check mark beside this Google Voice destination so that incoming calls to your GV number will be routed to your Asterisk server.
While you're still in the Google Voice Setup, click on the General tab. Uncheck Enable Call Screening. Turn Call Presentation Off. And set CallerID to Display Caller's Number. Be aware that IPkall DIDs only forward your IPkall number as the CallerID number while SIPgate DIDs reportedly forward the actual number of the person calling you. If this matters to you, then you may prefer the SIPgate DID option. Finally, uncheck Do Not Disturb. Now click the Save Changes button.
Integrating Google Voice into Asterisk with FreePBX. Open FreePBX with a web browser and choose Setup, Trunks, Add Custom Trunk. Insert your GV number in the Outbound CallerID field and add the following Custom Dial String on the form and Submit Changes and reload the dialplan:
local/$OUTNUM$@custom-gv
Next, choose Setup, Outbound Routes, Add Route and fill in the following entries on the form:
Route Name: GoogleVoice
Dial Pattern: 48|NXXNXXXXXX
Trunk Seq: local/$OUTNUM$@custom-gv
Inbound Routes. Next, we need two Inbound Routes to get everything working. In setting up your DID with IPkall or SIPgate, you already should have created one inbound route for that provider. It already should be routing calls to an extension on your PBX. Now we need to create a Custom Destination for this inbound route and then reroute these calls there. In that way, your RingBack calls will be routed to some special dialplan code that drops these calls into a custom parking lot where the RingBack call is married up to the extension from which you placed the original call. Then we need to create another inbound route to manage normal incoming calls that are forwarded to your PBX whenever someone dials your Google Voice number.
To begin, choose Tools, Custom Destinations, Add Custom Destination and add an entry like this and then click the Submit Changes button:
Custom Destination: custom-park,s,1
Description: Custom GV-Park
Next choose Setup, Inbound Route and click on the inbound route you created previously for IPkall or SIPgate. Change the destination for these calls to Custom Destination: Custom GV-Park.
Now click on Add Incoming Route and create a new route for your incoming Google Voice calls. Give it any description you like but, for the DID number, it must be gv-incoming. You can leave most of the other defaults. Just be sure you set a destination for your incoming calls from Google Voice. It could be an extension, ring group, IVR, or whatever best meets your needs. The important entry here is gv-incoming for the DID number. Click the Submit button to save your entries. Ignore the warning that you've entered an oddball DID. We know what we're doing. 🙂
Setting Up the Parking Lot. While still in FreePBX, we need to create or adjust your existing settings in Setup, Parking Lot. The parking lot is used by FreePBX to simulate old key telephones where you could place a call on hold and then someone else in the office could pick up the call by clicking on the blinking key on their phone. The Asterisk equivalent is to press the flash hook and dial your Parking Lot Extension which then places the call in a Parking Lot space and tells you what the space number is. Someone else then can dial the number of that space to pick up the call. Our little trick today works like this. When you place an outbound call through Google Voice, your extension will be dumped into a reserved parking lot space. When Google Voice initiates the RingBack call before connecting the destination number you've dialed, that call will be sent to the same reserved parking lot space. The two calls then are joined, and you'll hear the parking lot number followed by ring tones as your call is connected by GV to its final destination. Our special thanks to Richard Bateman for his comment on the previous article and this terrific tip! He wins an Atomic Flash installer from Nerd Vittles. In addition, A. Godong wins an Atomic Flash installer for his tip on consolidating two DIDs into a single DID to manage both inbound and outbound GV calls. Just send us your addresses.
Now, where were we? Most FreePBX systems have a default setup for the Parking Lot. What we need to do is be sure you have reserved one more space in the parking lot than you actually need for day to day operation of your PBX. We'll use the last parking lot space number to manage outbound calling through Google Voice. Our entries look like the following:
Enable Parking Lot Feature: checked
Parking Lot Extension: 70
Number of Slots: 5
Parking Timeout: 30 seconds
Parking Lot Context: parkedcallsDestination for Orphaned Calls: Terminate Call: Hangup
If you use our setup above, the Magic Number is 75 which is the fifth slot in the Parking Lot. If you use a different Parking Lot extension or number of slots, here's how to calculate the Magic Number. Start counting the slots beginning with one more than the Parking Lot Extension. When you get to the last slot in the number of slots you've specified, that's your Parking Lot Magic Number. Write it down. You'll need it in a second when you run our GV installation script.
Save your entries and reload the Asterisk dialplan when prompted.
Integrating pygooglevoice. Now we're ready to complete the setup by running our revised script which loads pygooglevoice and sets up your dialplan in extensions_custom.conf. You'll need 5 pieces of information to run the script so write them down before you begin:
1. Your 10-digit Google Voice phone number
2. Your Google Voice email address
3. Your Google Voice password (no spaces!)
4. Your 11-digit RingBack DID (16781234567)
5. Your Parking Lot Magic Number
A word of caution: If you used a gMail address to set up your Google Voice account, it's possible to have different gMail and Google Voice passwords. For this to work, you'll need to enter your gMail password, not your Google Voice password (assuming they're different).
Now log into your Asterisk server as root and issue the following commands:
cd /root
wget http://bestof.nerdvittles.com/applications/gv/install-gv-new
chmod +x install-gv-new
./install-gv-new
Google Voice Speed Dials. For frequently called numbers, you can add speed dials by inserting entries in the [from-internal-custom] context of extensions_custom.conf that look like the example below where 333 is the speed dial number and 6781234567 is the area code and number to call. Be sure to reload your Asterisk dialplan to activate them.
exten => 333,1,Dial(local/6781234567@custom-gv,300)
Congratulations! You now have what we hope will be flawless and free U.S. calling on your Asterisk system using Google Voice. No gimmicks, no strings, no cost. Enjoy!
Finally, one additional word of caution. Both Google Voice and this call design are set up for a single call at a time. There are no safeguards to prevent multiple calls, but that may violate the Google Voice terms of service.
Asterisk 1.6 Solution. Several readers now have documented the procedure for implementing the Asterisk 1.6 bridge technology to make outbound Google Voice calls. You can read all about it here.
Tweaking Previous Setups. If you installed pygooglevoice using our previous tutorial, here's what you need to do. First, log into your Asterisk server as root and issue the following commands:
cd /etc/asterisk
nano -w extensions_custom.conf
Scroll to the bottom of the file by pressing Ctrl-W then Ctrl-V. Move up the file using up arrow until you reach [custom-gv]. Press Ctrl-K repeatedly to delete all of the lines in the [custom-gv] context. If you get to another line that starts with a label in brackets like [this], STOP deleting. Once you've deleted all of the lines in the [custom-gv] context, save the file: Ctrl-X, Y, and press Enter.
Now continue reading this article by jumping up to the Google Voice Setup topic. The Custom Trunk entry and the GoogleVoice outbound route will already be in your FreePBX system so there's no need to repeat those two steps. You will need to perform the remaining FreePBX steps beginning at the Inbound Routes topic and continuing on with Setting Up the Parking Lot. Finally, when you run the new installation script, it will detect that pygooglevoice is already on your system and will skip that step but will install the new custom contexts in extensions_custom.conf using your new settings. Enjoy!
Thought for the Day. Which is more arbitrary: (1) Apple snubs Google Voice or (2) Google Voice snubs SIP? Pays to look in the mirror occasionally.
Best Read of the Week. Memo to Steve Jobs and Apple: Stop Being A Jerk!
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
whos.amung.us If you're wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what's happening. It's a terrific resource both for us and for you.
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...
Asterisk CallerID Perfected: CallerID Superfecta 2.0
If you’re new to VoIP and Internet Telephony, it may come as a shock when you discover that incoming calls no longer display the names of your callers unlike your POTS CallerID service. The problem is that most telephone providers deliver only a CallerID number when sending calls. Your DID service provider is responsible for looking up the incoming number in their directory and supplying a matching name. Simply stated, CallerID numbers are pushed to recipients, but CallerID names must be pulled from in-house databases. With Ma Bell and siblings, this was easy because they had a monopoly on all of the records. They’ve done their best to preserve that advantage. For example, Google provides very good CallerID information for people in the traditional phone book but provides no facility for adding numbers of those that don’t have a Ma Bell line. Coincidence? Probably not. With Internet Telephony Hosting Providers, CNAM lookups are few and far between. A few now will provide CallerID lookups but only on a per call fee basis.
A similar problem arises for outbound VoIP calls that you place. Even though you may provide a CallerID name and number, most telephone companies throw the supplied CallerID name in the bit bucket and do their own lookups. If you’re not in their directory, your number or nothing will be supplied instead of your actual name. Bottom line: This is one major piece of the telephone monopoly that Judge Greene forgot to address. And it’s only going to get worse now that more people own cellphones than Ma Bell landlines.
With Asterisk®, the simplest solution is to do your own CNAM lookups. And today we’re delighted to introduce CallerID Superfecta 2.0, a FreePBX module with incredible flexibility. It also serves as a living testament to how open source development actually should work.
The History Lesson. We originally wrote an AGI script for Asterisk@Home known as the CallerID Trifecta. As sources of CNAM lookups expanded, a number of other individuals contributed code to support those lookups. The original application morphed into the CallerID Superfecta when these new sources were added, but all the code still resided in a difficult-to-maintain PHP/AGI script. Tony Shiffer picked up the ball last year and converted the CallerID Trifecta into a FreePBX module which made it incredibly easy to install the application and configure the providers you wished to use. Fast forward to today, and now Jeremy Jacobs, with help from other members of the PBX in a Flash Forum community, has taken the original work of a number of PBX in a Flash Forum participants and released a real masterpiece, CallerID Superfecta 2.0. It’s not only easy to install and configure, but now it’s incredibly flexible and simple to expand as more and more people begin contributing lookup routines to this open source project. And Tony has put the icing on the cake with a new SVN development site to track bugs and feature requests for this project and others as we move forward.
The CallerID Superfecta Design. As originally implemented, CallerID Superfecta let you choose one or more lookup sources for incoming CallerID numbers. When an inbound call arrived, the sources were queried in a specified order, and the first source that provided a matching CallerID name won. The CNAM search result was returned to Asterisk for display on your phone instruments, and the lookup procedure ended. The problem with the original design was that newer and better lookup providers continued to appear, the hard-coded search order wasn’t necessarily ideal for every user or organization, and the providers kept changing formats to make lookups more challenging. CallerID Superfecta 2.0 fixes all of that!
What’s New. For openers there are a number of new lookup sources as well as some tweaks to older sources that stamped out a few bugs from our previous, sloppy code. 🙂 The web user interface (UI) for FreePBX also has been totally reworked. Now you can prioritize the lookups in the order that best meets your needs. And, as new CNAM lookup sources become available, they are automatically added to the new UI. The code also has been modularized with separate templates for each provider. This makes the addition of new lookup sources incredibly simple to implement. There’s also a new debug function built directly into the web user interface. By entering a telephone number in the new debug field and pressing the debug button on the form, the results from your selected lookup sources together with the latency of each enabled data source are displayed on the form for you to review. This new debug function greatly enhances troubleshooting while serving as a terrific tool to assist in fine tuning which providers to actually enable and in what order. Providers who can’t be reached, or who perform too slowly, or who provide lousy results can be turned off completely or moved to the bottom of the search order. Finally, CallerID Superfecta 2.0 introduces prefix code hooks. This gives developers the ability to trigger an additional outside process when the Caller ID function is initiated. For example, this feature might be used in a call center to allow the system to automatically perform an ODBC query and bring up a customer record for use by a customer service representative.
Installing CallerID Superfecta 2.0. Installation and setup should be a snap on any of the FreePBX-based Asterisk aggregations including PBX in a Flash. First, using a browser on your desktop PC, download CallerID Superfecta 2.0 from the Superfecta repository. Do not decompress the .tgz archive. Second, open FreePBX with your browser and choose Admin, Module Admin, Upload Module. Browse and select the superfecta-2.x.x.tgz module from your desktop and click the Upload button. When the upload completes, click local module administration. Scroll down and click CID Superfecta. Click the Install or Upgrade radio button depending upon whether you have previously installed the Superfecta FreePBX module. Click Process, Confirm, Return to install the new module. Reload the Asterisk dialplan when prompted.
Configuring CallerID Superfecta 2.0. There really are only two steps to bring CallerID Superfecta on line. First , we’ll configure the lookup sources and search order of the lookups. And then, for each inbound route on your Asterisk system, we’ll tell FreePBX to use CallerID Superfecta as the CallerID lookup source.
To configure CallerID Superfecta, click Admin, Setup, CID Superfecta in FreePBX. If you’re using PBX in a Flash or trixbox, be sure to insert the UserName maint and your FreePBX maint password in the fields provided under General Options. Then choose the Services you’d like to use for queries by clicking on the corresponding Enabled buttons. If you’re unfamiliar with previous versions of the product, we’d recommend you start with Addresses, White Pages, Yellow Pages, Any Who, and Telco Data. If you use the Asterisk Phonebook, AsteriDex, or SugarCRM, enable those options as well. Our rule of thumb in prioritizing the searches is to move your personal directories (Asterisk Phonebook, AsteriDex, and SugarCRM) to the top of the list. For the remaining choices, we recommend you start with the following search order: Addresses, White Pages, Yellow Pages, Any Who, and then Telco Data. Telco Data normally returns only the city and state of the caller, not the caller’s name. Who Called requires registration. Once you get everything squared away, click the Save button. Then key in a few known phone numbers in the Debug section of the form and click the Debug button to make sure everything is working as you expected. Take note of the retrieval times and the results and adjust the search order to meet your needs. Remember, the first match on a name using the search sources from top to bottom wins. The other search sources are never consulted for this number.
For additional configuration options and tips on configuring SugarCRM, see this thread on the PBX in a Flash Forum.
Once you’re satisfied with your lookup sources and the search order, the only remaining step is to designate CID Superfecta as the CallerID lookup source in your inbound routes. For each inbound route on which you want CallerID lookups performed, click Admin, Setup, Inbound Routes and choose the desired route from the column of routes on the right margin. Scroll to the CID Lookup Source section of the form and choose CID Superfecta from the dropdown box. Click Submit, Apply Config Changes, Continue to save your entry. Enjoy!
Good Things Are Coming. Sometimes you can just sense that something really terrific is about to happen with a project that’s been properly nurtured. So it is with PBX in a Flash. Many of us invested countless hours in the Asterisk@Home project, but it just never quite took off in terms of fostering independent development. And that is what open source is all about. When we started the PBX in a Flash project with Tom King and Joe Roper, we wanted something different, a participatory platform that could get folks excited about Internet telephony and open source development. As long time readers of this column know, we’ve pumped lots of applications into PBX in a Flash in hopes of encouraging others to catch the spark. Well, we’ve made it. CallerID Superfecta 2.0 is a perfect example of a creation from a developer who hasn’t previously contributed to any open source project. And there now are many more in the works. So we’re excited for the Asterisk community. Seeing your baby grow up is quite a thrill. We’re quickly building a network of application developers and contributors of many varieties, and it is starting to transform a really good telephony toolbox into a really spectacular application platform. Have a look at this new project to get a glimpse of what lies ahead.
This isn’t a private club. If you’d like to participate in a big or a small way, please visit our new, non-denominational Development Site and kick the tires. The objective is to create applications that can run on any of the FreePBX-based Asterisk distributions. So, while we’re a little biased, you don’t have to be to join in on the fun. Just click the Register link at the top of the form. Welcome!!!
Other Coming Attractions. Well, we got distracted again this week because CallerID Superfecta 2.0 was such as terrific new product. We’ll wrap up the Orgasmatron Installer tutorial in our next column. Stay tuned!
Security Alert. Secunia has published some security vulnerabilities in FreePBX. Upgrade all of your FreePBX modules at your earliest convenience.
Twitter Magic. If you haven’t noticed the right margin of Nerd Vittles lately, we’ve added a new link to our Twitter feed. If you explore a little, you’ll discover that the user interface now brings you instant access to every Twitter feed from the convenience of the Nerd Vittles desktop. Enjoy!
whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.
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…
Why Wait? Build Your Own Skype Gateway to Asterisk
As the world awaits the much-hyped Digium® commercial offering of a Skype for Asterisk® gateway, we began exploring existing alternatives last week that provide individuals1 the same functionality at what will no doubt be considerably less cost. The Gizmo5 offering is darn-near perfect for those that need a quick solution and don't mind spending $20 a year to let someone else wrestle with the technical complexities that invariably accompany maintenance of cross-platform gateways such as this one.
NOTE: Article has been updated. For the latest edition, follow this link.
We, of course, like the technical challenge that comes with the territory so today we turn our attention to Greg Dorfuss' SipToSis software which forms the lynchpin of Gizmo's offering and which lets any Asterisk user create much the same gateway at no cost other than the expense of any Skype Out calls you may choose to make. When we're finished, you'll be able to call any Skype user in the world from any extension on your Asterisk server by entering either a Skype username or any 10-digit telephone number preceded by an 8 to take advantage of SkypeOut calling rates. You'll also be able to receive incoming calls from any Skype user on any extension of your Asterisk system. In short, what you get is a transparent interface to several hundred million Skype users from your Asterisk server. And the time to set this up is less than 15 minutes assuming you already have one of the Nerd Vittles Orgasmatron Builds in operation. These are available for Dell PowerEdge servers, Everex gPC desktop systems and minis, and most of the new Atom-based systems. If you have a garden-variety FreePBX-based Asterisk server such as PBX in a Flash, trixbox, or Elastix, just add another 5 minutes to reconfigure a few things.
Prerequisites. For today's project, we're assuming you have an existing FreePBX-based Asterisk server with either CentOS 5.2 or the Fedora 10 Remix featured in our Atomic Flash build. You'll need both a keyboard and mouse! For inbound Skype calling to work with other implementations including generic PBX in a Flash systems, you'll need to create a SIP URI for your Asterisk server: mothership@127.0.0.1. We've previously explained how to set one up in this article. The Atomic Flash installer, VPN in a Flash build, and the Orgasmatron II and III builds include this SIP URI functionality out of the box. You'll also need Java 1.5. To see if it's included in your distribution, issue the following command: rpm -q jdk. Finally, we're assuming you already have an existing Skype account. If not, download the Skype software for your Mac or Windows PC, and sign up. Try out a demo from any Skype phone. Just call nerdvittles.
Installing JAVA. If your particular Asterisk distribution doesn't have JAVA 1.5 or higher installed (rpm -q jdk), here's how to do it. Go to Sun's Java SE Development Kit 6u12 website, choose Linux for the platform, and agree to the license agreement. Then click Continue. Download jdk-6u12-linux-i586-rpm.bin and copy it to your Asterisk server. Make the file executable (chmod +x jdk-6u12-linux-i586-rpm.bin) and then run it. Scroll down the wordy license agreement AGAIN and type yes. Java 1.6 then will be installed on your system. Whew!
Basic Installation. Now we're ready to get started. Log into your Asterisk server as root and issue the following commands.
cd /root
mkdir skype
cd skype
wget http://www.skype.com/go/getskype-linux-centos
#Atomic Flash builds including VPN in a Flash
#should skip the next 3 yum commands.
#Then pick up again after the next comment line.
yum install xorg-x11-server-Xvfb
yum install qt4
yum install xterm
#everyone continues on from here
yum install libXScrnSaver.i386
wget http://pbxinaflash.net/source/skype/siptosis.tgz
rpm -ivh skype*
cd /
tar zxvf /root/skype/siptosis.tgz
FreePBX Design. The FreePBX setup that we recommend goes something like this. For outbound Skype calls, you have two choices.
1. To place a call to a regular phone number using SkypeOut (which costs you money), you'll simply dial 8 plus the area code and number. Our foreign friends will have to adjust their dialplans and /siptosis/SkypeOutDialingRules.props accordingly. Today's setup assumes 10-digit phone numbers!
2. To place a call to a Skype username using a softphone that supports SIP URI dialing such as X-Lite, you simply precede the Skype username with an asterisk, e.g. *echo123 will connect you to the Skype Call Testing Service.
For incoming Skype calls, the default setup routes those calls to a SIP URI: mothership@127.0.0.1. Whether you point this URI to an extension, ring group, or IVR is up to you. In the default Orgasmatron and Atomic Flash builds, the mothership URI is pointed to the Stealth AutoAttendant, an IVR that plays a welcoming message and then transfers the call to a ring group if no digit is pressed by the caller.
Configuring FreePBX. To put this setup in place, use a web browser to access FreePBX on your Asterisk server. You'll need to create a Custom Trunk and then an Outbound Route.
1. Choose Setup, Add Trunk, Add Custom Trunk. Fill in the form so that it looks like the following using your own CallerID number obviously:
When you're finished, click the Submit Changes button and then reload the dialplan when prompted.
2. Next choose Setup, Outbound Routes, Add Route. Fill in the form so that it looks like this:
When you're finished, click the Submit Changes button. Be sure to move this new OutSkype route to the top position in your Outbound Routes listing in the right margin! Then reload the dialplan when prompted.
3. If you're not using one of our custom Asterisk builds, add a new DayNight Control 1 option while you're still in FreePBX. Just specify where you want calls routed for Day mode and Night mode. Then, here's the easy way to activate SIP URI support on your Asterisk/FreePBX server. Copy the [from-sip-external] context from the extensions.conf file in /etc/asterisk. Now copy the content into extensions_override_freepbx.conf. Be sure to preserve the context name in brackets! Now make it look like the following. The additions we're making are shown in bold below:
[from-sip-external]
;give external sip users congestion and hangup
; Yes. This is _really_ meant to be _. - I know asterisk whinges about it, but
; I do know what I'm doing. This is correct.
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)
exten => s,1,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?from-trunk,${DID},1)
exten => mothership,1,Goto(app-daynight,1,1)
exten => pbxinaflash,1,Goto(app-daynight,1,1)
exten => e164,1,Goto(from-trunk,e164,1)
exten => fax,1,Goto(from-trunk,fax,1)
exten => s,n,Set(TIMEOUT(absolute)=15)
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,Playtones(congestion)
exten => s,n,Congestion(5)
exten => h,1,NoOp(Hangup)
exten => i,1,NoOp(Invalid)
exten => t,1,NoOp(Timeout)
Finally, reload your Asterisk dialplan, and we're finished with Asterisk and FreePBX setup:
asterisk -rx "dialplan reload"
Activating Your Skype Gateway. Now we're ready to place your Skype gateway in production. You'll need to perform these steps from the console on your Asterisk server since we have to run Skype in graphics mode. This may look a bit complicated. It's really not. It's just a bit tedious to figure out the sequence of steps, but we've done that part for you.
For those on a CentOS-based system, here are the steps:
1. Start up X-Windows: xinit
NOTE: If xinit won't start on your particular machine, you may need to create /etc/X11/xorg.conf. Here's a generic config file that should work fine for our purposes:
Section "ServerLayout"
Identifier "X.org Configured"
Screen 0 "Screen0" 0 0
EndSectionSection "Device"
Identifier "Card0"
Driver "vesa"
EndSectionSection "Screen"
Identifier "Screen0"
Device "Card0"
SubSection "Display"
Viewport 0 0
Depth 16
Modes "800x600"
EndSubSection
SubSection "Display"
Viewport 0 0
Depth 16
Modes "800x600"
EndSubSection
EndSection
2. Next we need to start up Skype, log in, set it to autologin whenever we start it, and then in the Skype configuration, set it to run minimized: skype
NOTE: Place a Skype Test Call (echo123) to be sure your audio settings are set correctly. If a test call fails with a bad audio message, go into Options, Sound Devices and reconfigure your Audio settings until you can place the test call successfully. Otherwise, none of the rest will work! If you're using a Dell server such as the PowerEdge T100, you're probably S.O.L. without an advanced degree in Rocket Science.
HINT: Intel Atom-based motherboards are a piece o' cake!
3. Once you've got Skype working and all of the Skype settings configured above, shut down Skype.
4. Now start Skype again in background mode: skype &
5. Be sure to write down the PID for Skype in case you need to kill the job if something goes wrong. 🙂
6. You should have a command prompt back now. So issue these commands:
cd /siptosis
./SipToSis_linux
7. A message from Skype should pop up asking if you want to authorize external use of Skype: yes. Also be sure to mark the Checkbox to save this setting for future connections!
8. Now go to a softphone connected to an extension on your Asterisk server and dial *echo123
9. Go to any extension connected to your Asterisk server and dial 8 + your home phone number. This will place the outbound call through SkypeOut at 2¢ a minute.
Finally, here are a few navigation tips for managing your Asterisk console on CentOS systems:
1. Ctrl-Alt-F2 gets you a new login prompt for your server
2. Ctrl-Alt-F7 gets you back to the SipToSis/Skype session. You can kill SipToSis by holding down Ctrl-C for several seconds. To kill Skype: kill pid# (that you wrote down). To restart Skype: skype & and to restart SipToSis, just issue the command again: ./SipToSis_Linux
3. Ctrl-Alt-F9 gets you to the Asterisk CLI.
Automating the Skype Gateway Startup. Once you're positive everything is working reliably and you've rebooted and tried it again just to make certain there are no prompts, here's how to fire up your Skype gateway whenever you reboot your server.
Log into your server as root and issue the following commands:
cd /root
wget http://pbxinaflash.net/source/skype/start-skype
chmod +x start-skype
echo "/root/start-skype" >> /etc/rc.d/rc.local
reboot
For those using one of our Fedora 10 builds such as VPN in a Flash or the Atomic Flash installer, these systems have a full implementation of X-Windows and KDE. Just start the system in mode5 (graphics mode), log in, run Skype in one window and start up SipToSis in a terminal window using the commands in Step 6 above. Authorize external use of Skype when prompted.
Setting Up Speed Dials for Skype Friends. One of the wrinkles with Skype is that Skype uses names for its users rather than numbers. If you don't have a SIP URI-capable softphone, there's still an easy way to place calls to your Skype friends using FreePBX. Just add a Speed Dial number to your FreePBX dialplan. Choose Extension, then select the Custom type, provide an Extension Number which is the Speed Dial number (this could actually spell your friend's name using a TouchTone phone), enter a Display Name for your friend, and add an optional SIP Alias. Then insert the following in the dial field replacing joeschmo with your friend's actual Skype name. Save your entries and reload the dialplan when prompted.
SIP/joeschmo@127.0.0.1:5070
Security Warning. One final note of caution. Do NOT expose UDP port 5070 to the Internet unless you first secure this port with a username and password to avoid Internet intruders using your gateway as a free Skype dialing platform! You do not need 5070 exposed to the Internet to implement today's gateway solution for inbound or outbound Skype calling from your Asterisk server so we recommend you keep it securely behind at least one firewall.
Where To Go From Here. Well, those are the basics. You now can make one outbound Skype call at a time from your Asterisk server, and you can receive an inbound Skype call on any Asterisk extension when Skype users call your regular Skype name. If you want to start up your own business (like Gizmo5), then you'll need to do some tweaking. What you'll need is the STS Trunk Builder toolkit which is free, but proprietary. Enjoy!
Want a Bootable PBX in a Flash Drive? Our Atomic Flash bootable USB flash installer for PBX in a Flash has been quite the hit. Special thanks to all of our generous contributors! Atomic Flash provides all of the goodies in the VPN in a Flash system featured last month on Nerd Vittles. You can build a complete turnkey system using almost any current generation PC with a SATA drive and this USB flash installer in less than 15 minutes!
If you'd like to put your name in the hat for a chance to win a free one delivered to your door, just post a comment with your best PBX in a Flash story.2
Be sure to include your real email address which will not be posted. The winner will be chosen by drawing an email address out of a hat (the old fashioned way!) from all of the comments posted over the next several weeks.
And it still isn't too late to make a contribution of $50 or more to the PBX in a Flash project and get a free Atomic Flash installer delivered to your door as our special thank you gift. See this Nerd Vittles article for details.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest...
- Skype and this suggested implementation are intended for individual use. Your use is, of course, governed by the Skype Terms of Service. [↩]
- This offer does not extend to those in jurisdictions in which our offer or your participation may be regulated or prohibited by statute or regulation. [↩]
Adding Post-Dial Processing to Asterisk and FreePBX Dialplans
Last week we introduced a couple of new free calling options for Asterisk®: ENUM and Gizmo5's Backdoor Dialing. But one of the limitations of the Gizmo5 service in particular was the need for a 0101 prefix in order to trigger a free call as opposed to a pay per minute call to the same number. This highlighted a pretty serious limitation in the way FreePBX processes most outbound calls. As we indicated, the process goes something like this. After a caller dials a number, FreePBX searches through its Outbound Routes (from top to bottom) looking for a match on the dial string. Once it finds one, FreePBX then initiates calls beginning with the first trunk in the trunk priorities list for that outbound route. If the call is completed, no further call processing takes place. And a completed call includes a call that is either answered or rings busy. If a call is not completed, FreePBX continues to drop down the available trunks list and repeats the process until a call is either completed or the trunk list is exhausted. The one exception to this scenario was support for ENUM. In that situation, a lookup occurs after a call is dialed to see if it can be placed as a free SIP call. We'd like to do the same thing with Gizmo5's Backdoor Dialing. What we want to do is query the Gizmo5 database to determine whether a number to be called is a free call. If it is, then we want to modify the route for processing the outbound call to take advantage of Gizmo5's free calling option. And we'll also need to change the phone number by adding a 0101 prefix.
Since our last column, another serious limitation in FreePBX post-call processing was mentioned on the PBX in a Flash Forums. With a number of commercial PBXs, it's possible to specify post-dial processing for emergency calls. For example, in an office environment, if an employee dialed 911, it would be helpful to alert a receptionist in some way so that immediate first aid could be attempted and also to give the receptionist a heads up so that he or she could direct emergency responders to the appropriate location in a building. As written, FreePBX doesn't provide an easy way to handle this.
So our objective today is to provide a couple applications which address these limitations. And the apps also will document a methodology for overcoming other post-dial processing issues which may arise using the existing FreePBX framework.
The trick to adding today's hooks into the Asterisk dialplan is to understand that Asterisk loads identically named dialplan contexts only once. Taking advantage of this, FreePBX provides a mechanism for users to insert custom code to replace default FreePBX contexts. All of these configuration files are stored in /etc/asterisk. For today, the context we want to modify is [macro-dialout-trunk]. This is the FreePBX macro that does the heavy lifting once a call has been placed and a trunk route has been selected to handle the call. With FreePBX 2.3, the macro is in extensions.conf. In FreePBX 2.4 and 2.5, the context is in extensions_additional.conf. In both cases, what we want to do is copy the entire contents of the existing context into the bottom of extensions_override_freepbx.conf. If you're using an editor to cut-and-paste the code, be sure you get the code that is located outside the left and right margins of your editor. And the context ends on the line before the next context begins. In the case of FreePBX 2.3, the next context is [macro-agent-add]. In the case of FreePBX 2.4, the next context is [macro-dialout-dundi]. And, in 2.4, there is now a comment which indicates where each context ends: ; end of [macro-dialout-trunk].
What we want to do is insert a line or two of custom code in this context which you've copied into extensions_override_freepbx.conf. The purpose is to run our custom code after the number to dial and trunk ID have been passed to this macro. Then, in the case of the Gizmo5 application, we'll run out to the Internet to determine if this call should be handled as a free call. If so, we'll change the trunk ID number to match your Gizmo5 trunk, and we'll change the number to dial by prefixing the existing number with 0101. The only gotcha with the Gizmo5 Backdoor Dialing is that every number must be tested at least once by someone (not necessarily you) in order to populate the Gizmo5 free calling database. You can check as many numbers as you like at this link. In the case of our 911 emergency application, we'll check to see if the number being dialed is 911. If so, we'll send an email or text message to an address that you define with an alert that extension 1234 just placed a call for emergency assistance to 911.
If you're using FreePBX 2.3, the custom code below should be inserted after the third "exten" line in the context, i.e. after the following line of code:
exten => s,n,Set(ROUTE_PASSWD=${ARG3})
If you're using FreePBX 2.4, the custom code below should be inserted after the first "exten" line, i.e. after the following code:
exten => s,1,Set(DIAL_TRUNK=${ARG1})
And the code to be inserted looks like this for Asterisk 1.4:
exten => s,n,AGI(nv-outbound.php|${ARG2}|${ARG1})
exten => s,n,AGI(nv-gizmo.php|${ARG2}|${ARG1})
For Asterisk 1.6, it should look like this:
exten => s,n,AGI(nv-outbound.php,${ARG2},${ARG1})
exten => s,n,AGI(nv-gizmo.php,${ARG2},${ARG1})
Now we need to add a couple of PHP scripts to your system and set a few configuration options, and you'll be ready to go. While logged into your server as root, issue the following commands:
cd /var/lib/asterisk/agi-bin
wget http://pbxinaflash.net/source/gizmo/nv-gizmo.zip
unzip nv-gizmo.zip
rm nv-gizmo.zip
wget http://pbxinaflash.net/source/gizmo/nv-outbound.zip
unzip nv-outbound.zip
rm nv-outbound.zip
chown asterisk:asterisk *.php
chmod +x *.php
asterisk -rx "dialplan reload"
grep OUT_ /etc/asterisk/extensions_add* | awk '/ = / { print $0 }'
The last line of code above is used to decipher the trunk numbers associated with each of your trunks. What we need to know is the trunk number for the Gizmo5 trunk that you set up in last week's tutorial. Write it down and then edit nv-gizmo.php: nano -w nv-gizmo.php. Look down the screen about 5 or 6 lines for the line that reads $GIZMO_TRUNK = "21" ; and replace 21 with the number you wrote down for your actual Gizmo5 trunk. In the next two lines, insert your actual Gizmo5 username and password between the quotation marks. Don't change anything else. Save your changes: Ctrl-X, Y, and then press the Enter key.
With the other application, nv-outbound.php, we need to be sure it's working with your phone system before you actually need it. And we don't place test calls to 911. So here's the drill. Edit the file: nano -w nv-outbound.php and insert your email address or text message address in the $email variable between the quotes. Then move to the next line and insert a telephone number with the area code that you can dial from a phone on your system to test that the notification is working. For example, put in your cell phone number. Once you save your changes, pick up a phone on your system and call your cellphone. You should receive an email notification within a few seconds. Once it's working, edit the application again and change the $number2monitor to "911" and you're all set. Enjoy!
VPN in a Flash Update! We've had over 100 reservations for our new VPN in a Flash system. We're very close to having a manufacturer in place so hopefully we'll have more good news in a week or two. We have begun the documentation for the new product, and we encourage you to take a look and offer any questions or comments you may have on our forums. The documentation is in the new Google Knol format and can be reviewed here. It's not too late to get in the queue and place a reservation for a system. Just send us a note, and we'll keep you posted as the release date approaches. It'll hold your place in line with absolutely no obligation to purchase.
Coming Attractions. We're very close to signing on a new VoIP provider for PBX in a Flash users that will provide penny-a-minute calls in the U.S. and Canada as well as all-you-can-eat plans for just over $10 a month with an annual contract. We're also only a week or two away from a new version of AsteriDex with Outlook synchronization and a TTS dialer for AsteriDex queries from any connected Asterisk phone. Stay tuned!
Hosting Provider Deal of the Century. Just an FYI that the Nerd Vittles hosting provider, BlueHost, has raised the bar again on hosting services. For $6.95 a month, you can host unlimited domains with unlimited web hosting disk storage and unlimited monthly bandwidth. Free domain registration is included for as long as you have an account. It really doesn't get any better than that. And their hosting services are flawless! Just use our link. You get a terrific hosting service, and we get a little lunch money.
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