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Free Asterisk Voicemail Transcription with IBM Watson STT

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There are many commercial voicemail transcription services for Asterisk® PBXs, but none hold a candle to the speech-to-text (STT) quality of the IBM Cloud offering known as Watson® STT, formerly known as Bluemix TTS. Despite a recent price increase that takes effect in December, the pricing remains competitive. On the Standard Pricing Plan, voicemail transcription is 2¢ per minute. Or you can try things out on the LITE plan which offers 100 minutes a month at no cost. When the messages are delivered by email, you get the voicemail recording in MP3 format AND transcribed text courtesy of Watson TTS. With IBM services, there no longer are username:password credentials. Instead, you will have a new apikey.

Those with existing configurations can update your credentials by inserting a new apikey using the following commands, or you can simply insert apikey as your $API_USERNAME and enter your actual APIkey as your $API_PASSWORD.

cd /usr/local/sbin
sed -i 's|$API_USERNAME:$API_PASSWORD|"apikey:x-yy-zzz"|' sendmailmp3
sed -i 's|$API_USERNAME:$API_PASSWORD|"apikey:x-yy-zzz"|' bluemix-test

IBM Cloud’s STT solution is a real game-changer for one simple reason. Their STT API performs more accurately than any speech recognition engine in the world. As an added bonus, you won’t have to worry about Google breaking our middleware every month. It’s worth noting that IBM doesn’t round up minutes. Transcribing two 30-second messages counts as one minute.


https://youtu.be/JWnLgZ58zsw

Overview. What we’ve done today is integrate the Watson STT API directly into existing Asterisk voicemail systems. We started with Nicolas Bernaerts’ terrific sendmailmp3 script. It works on both the Wazo and FreePBX® platforms. If you have deployed Incredible PBX®, then the setup takes a couple of minutes. For everyone else, there’s an additional configuration step using your favorite GUI. To get started, you’ll sign up for an IBM Cloud account and obtain your credentials. Next, you download today’s script for your platform and insert your credentials. Finally, you set up voicemail on the extensions desired and insert an email address for each voicemail account. On generic FreePBX systems, you’ll need to add the name of our script to manage your voicemail recordings. And, regardless of your PBX platform, you obviously need outgoing SMTP email working reliably.

Start by sending yourself a test email and get that working first:

echo "test" | mail -s testmessage yourname@your-email-domain.com

What About the Quality? Here’s the bottom line. Speech recognition isn’t all that useful if it fails miserably in recognizing everyday speech. The good news is that IBM Watson’s speech recognition engine is now the best in the business. If you want more details, read the article below which will walk you through IBM’s latest speech recognition breakthrough:


Obtaining IBM Cloud Speech to Text Credentials

Follow this link to set up your IBM account and obtain credentials for both Speech to Text (STT) and Text to Speech (TTS) services. Please note that your STT and TTS API keys will NOT be the same. So don’t accidentally use the wrong one.

 

Installing STT Engine with Incredible PBX for Wazo

1. After logging into your Incredible PBX for Wazo server as root using SSH/Putty:

cd /usr/sbin
wget http://incrediblepbx.com/sendmailibm.tar.gz
tar zxvf sendmailibm.tar.gz
rm -f sendmailibm.tar.gz

2. Edit sendmailibm and insert IBM STT API_KEY and URL.

3. Edit bluemix-test and insert IBM STT API_KEY and URL.

4. Apply the patch documented above if using LITE plan using sendmail filename instead of sendmailmp3.

5. Copy the updated sendmailibm file to sendmail:

cd /usr/sbin
cp -p sendmailibm sendmail

6. Test your Bluemix STT setup: bluemix-test

7. Result should be: please record your message after the beep

8. Set up voicemail account for a Wazo extension with your email address.

9. Place a test call to the extension and record a voicemail when prompted.

10. Your message will be transcribed and delivered via email.

 

Installing STT Engine with Incredible PBX for RasPi

1. After logging into your Raspberry Pi server as root using SSH/Putty:

cd /usr/sbin
wget http://incrediblepbx.com/sendmailibm-raspi.tar.gz
tar zxvf sendmailibm-raspi.tar.gz
rm -f sendmailibm-raspi.tar.gz

2. Edit sendmailmp3.ibm and insert your Bluemix STT API_KEY and URL. Save file.

3. Edit bluemix-test and insert your Bluemix STT API_KEY and URL. Save the file.

4. Copy the updated sendmailmp3.ibm file to sendmailmp3:

cd /usr/sbin
cp -p sendmailmp3.ibm sendmailmp3

5. Apply the patch documented above if using LITE plan.

6. Test your Bluemix STT setup: bluemix-test

7. Result should be: your dictation is now being processed and emailed please wait

8. Set up voicemail for a RasPi extension with your email address.

9. Place a test call to the extension and record a voicemail when prompted.

10. Your message will be transcribed and delivered via email.

 

Installing STT Engine with Incredible PBX 13-13

1. After logging into your Incredible PBX 13 server as root using SSH/Putty:

cd /usr/local/sbin
wget http://incrediblepbx.com/sendmailibm-13.tar.gz
tar zxvf sendmailibm-13.tar.gz
rm -f sendmailibm-13.tar.gz

2. Edit sendmailmp3.ibm and insert your IBM STT API_KEY and URL. Save file.

3. Edit bluemix-test and insert your IBM STT API_KEY and URL. Save the file.

4. Copy the updated sendmailmp3.ibm file to sendmailmp3:

cd /usr/local/sbin
cp -p sendmailmp3.ibm sendmailmp3

5. Test your Bluemix STT setup: bluemix-test

6. Result should be: we are now transferring you out of the company directory…

7. Set up voicemail for an extension and include your email address.

8. Place a test call to the extension and record a voicemail when prompted.

9. Your message will be transcribed and delivered via email.

 

Installing STT Engine with VitalPBX

For those using VitalPBX with or without Incredible PBX, we’ve written a new tutorial to walk you through the procedure to get voicemail transcription with IBM Watson STT up and running. Here’s the link.

Installing STT Engine with Legacy FreePBX Servers

1. Follow steps #1 through #8 from the Incredible PBX 13 tutorial above.

2. Choose Settings -> Voicemail Admin -> Settings in the GUI.

3. In the format field, insert: wav|wav49

4. In the mailcmd field, insert: /usr/local/sbin/sendmailmp3

5. Click Submit to save your settings and then Reload the FreePBX Dialplan.

6. Place a test call to the extension and record a voicemail when prompted.

7. Your message will be transcribed and delivered via email.

Update: Matt Darnell reports that, depending upon your existing setup, you may need to add the unix2dos and lame packages with legacy FreePBX servers to get MP3 messages delivered correctly.

 

Originally published: Monday, March 12, 2018  Updated: Monday, November 12, 2018




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Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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.

blankBOGO 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.

blankThe 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.

blankVitalPBX 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!
 

blankSpecial 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…

Double-NAT Blues: Tackling Asterisk’s Thorniest Problems

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Whether you’re new to VoIP technology or an Old Timer, nothing is quite as frustrating as wrestling with one-way audio and no audio on SIP calls either because of poorly designed NAT-based routers or poorly implemented SIP ALG solutions on low-end residential routers. To make matters worse, you get to deal with calls originating behind not one, but two, NAT-based routers neither of which complies with the basic SIP Rules of the Road. In a perfect world, SIP and RTP packets arriving from the Internet would have their public IP address translated into a private LAN address upon arrival at the NAT-based router. And the departing packets would have their private IP addresses translated into the public IP address of the router when leaving. If your PBX and SIP phone happen to be behind different NAT-based routers and hardware from the likes of Comcast, Spectrum, and AT&T, the odds of SIP calls working reliably are somewhere between slim and none. Perhaps it’s no coincidence that each of these providers also happens to offer competing (expensive) telephony service.

Today we’d like to offer some Asterisk® solutions that resolve these issues. First, if you are the subscriber to cable or DSL Internet service, you may have some success by talking to your provider and persuading them to set up their hardware in bridged mode so that you can install your own NAT-based router that properly handles SIP traffic. Second, it’s almost always a good idea to disable SIP ALG service on routers that you control. The reason is because of the poor ALG implementations on almost all low-cost routers. Third, configuring the Public and Private IP NAT Settings for your PBX using the FreePBX® GUI (Settings->Asterisk SIP Settings->NAT Settings) often solves the problems. Fourth, make sure NAT=yes is set in your extension and trunk settings.

If you happen to be traveling and have no control over the network architecture, the chances of the above recommendations resolving your SIP problems are not likely. This includes offerings in hotels, rental units, cruise ships, and WiFi HotSpots worldwide. In most of these locations, you would want to use a SIP phone to connect back to your home or office PBX so that you could receive incoming calls and place outbound calls just as if you were sitting at your desk at home. In these situations, we have a failsafe solution for you, but it requires a little advance planning because you need to configure your home or office Asterisk server to support the design.

The easiest way to eliminate NAT problems is to take NAT out of the equation when making and receiving SIP calls. With Asterisk, this is easy. What we typically do is interconnect the home or office Asterisk PBX with a local Asterisk PBX using an IAX2 trunk. Thus, no SIP traffic passes between your local PBX and your home or office PBX regardless of the number of layers of routers that are present between the two servers. If you can make SIP calls through a provider while sitting at home, you have solved the SIP connectivity issues at the home/office end. If your local PBX and SIP phone or softphone are on the same local LAN whether wired or wireless, then there is no SIP connectivity issue locally either. So how?

Rule #1: Always travel with a notebook computer that includes VirtualBox and a reliable SIP softphone. We’re big fans of all of the Mac notebooks, any of them will suffice. Windows and Linux notebooks work as well. Steer clear of Chromebooks which lack a crucial Linux kernel driver required by VirtualBox. There’s a solution, but it’s painful. On the Mac platform, you can’t beat the free Telephone app for your SIP phone.

Rule #2: Set up a NeoRouter VPN to provide secure interconnectivity between your home or office PBX and your local PBX. With Incredible PBX platforms, the NeoRouter client is included. You’ll just need to install the NeoRouter server component on some server with a public IP address. Complete details are here. To obtain a NeoRouter private IP address on each PBX, run this command after logging in as root: nrclientcmd.

Configuring IAX Trunk on Home/Office Server. You’ll need the NeoRouter IP address and a secure password to set up the trunk that will interconnect your Home-PBX with your local PBX. We’re going to refer to the two servers as Home-PBX (10.0.0.1) and Travel-PBX (10.0.0.2) to keep things simple. On the Home-PBX, create an IAX trunk using the FreePBX GUI with a Trunk Name of Travel-PBX. The PEER Details should look like the following using a very secure password that will be used on the trunk at the other end as well:

type=friend
secret=very-secure-password
host=dynamic
context=from-internal
requirecalltoken=no
deny=0.0.0.0/0.0.0.0
permit=0.0.0.0/0.0.0.0

The Registration String would look like the following where very-secure-password is your actual shared secret for the two trunks and 10.0.0.2 is the actual VirtualBox IP address of the Travel-PBX: Home-PBX:very-secure-password@10.0.0.2

Configuring IAX Trunk on Travel-PBX Server. You’ll need the NeoRouter IP address and a secure password to set up the trunk that will interconnect your Travel-PBX server with your Home-PBX. On the Travel-PBX, create an IAX trunk using the FreePBX GUI with a Trunk Name of Home-PBX. The PEER Details should look like the following using a very secure password that will be used on the trunk at the other end as well:

type=friend
secret=very-secure-password
host=dynamic
context=from-internal
requirecalltoken=no
deny=0.0.0.0/0.0.0.0
permit=0.0.0.0/0.0.0.0

The Registration String would look like the following where very-secure-password is your actual shared secret for the two trunks and 10.0.0.1 is the actual VirtualBox IP address of the Home-PBX: Travel-PBX:very-secure-password@10.0.0.1

Once you get this far, log into both servers as root and start up the Asterisk CLI. On each server, issue the following command to be sure the two trunks are registered with each other: iax2 show registry

Routing Calls from Home-PBX to Travel-PBX. What follows is one scenario for call routing. We’re assuming calls to your Home-PBX are routed to a Ring Group consisting of various extensions in your home or office. We’re also assuming you want to now add an extension on Travel-PBX to that Ring Group so that incoming calls to your Home-PBX will also ring the softphone connected to an extension on your Travel-PBX. In the Asterisk/FreePBX world, we accomplish this by adding an Outbound Route for the Travel-PBX extension and then adding this number to the Ring Group with a # prefix to tell FreePBX that it’s a trunk call rather than a local extension. In our example, we’re assuming the softphone extension on Travel-PBX is 701, but we’re also assuming there is a different extension 701 on Home-PBX. To avoid confusing the Home-PBX, we’ll add a 7 prefix for the Travel-PBX extension and then strip it off before passing the call to Travel-PBX.

First, create an Outbound Route called Travel-PBX-Out. For the Dial Pattern, enter a Prefix of 7 and a Match Pattern of 701. For the Trunk Sequence, choose Travel-PBX. Move the Outbound Route near the top of your route list to assure that it gets processed before any other 4-digit extensions. Second, edit your Ring Group and add 7701# to the existing list.

Routing Calls from Travel-PBX to Home-PBX. On the Travel-PBX, we’re assuming you’d like calls placed from your softphone to be processed exactly as if you were calling from a local extension on Home-PBX. Create an Outbound Route called Home-PBX-Out. For the Dial Patterns, add one for 10-digit calls: NXXNXXXXXX. If you want to be able to reach 3-digit extensions on Home-PBX, add a second dial pattern with a 9 prefix and XXX for the Match Pattern so it doesn’t conflict with local extensions. For Trunk Sequence, choose Home-PBX.

Originally published: Monday, August 20, 2018


blankSupport Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.


blank
Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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.

blankBOGO 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.

blankThe 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.

blankVitalPBX 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!
 

blankSpecial 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 the GPL Toolkit for FreePBX and Incredible PBX


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We’ve been scratching our head for a good way to commemorate Micro$oft’s $7.5 billion purchase of GitHub which has served as the linchpin of the open source development community for many years. We’ll leave it to others and history to judge whether this was a good idea or not. What we came up with was a GPL Toolkit for Incredible PBX 13-13 that makes it child’s play to upgrade FreePBX® GPL modules in our Incredible PBX® 13-13 offerings for CentOS/SL, Ubuntu, and Raspbian using the FreePBX GitHub repository. Because Incredible PBX platforms don’t use module signature checking, it was fairly simple to design an upgrade methodology that leverages the FreePBX 13 module offerings posted on GitHub. While these modules would cause all sorts of module signature alarms to go off if used with other distributions, with Incredible PBX, implementation is straight-forward and painless.

To get started, log into your Incredible PBX 13-13 server as root and issue the following commands to download the GPL Toolkit:

cd /root
wget http://incrediblepbx.com/gpl-toolkit.tar.gz
tar zxvf gpl-toolkit.tar.gz
rm -f gpl-toolkit.tar.gz

Once the install finishes, you’ll end up with 7 scripts: gpl-module-list-full, gpl-module-list-fpbx, gpl-module-list-contrib, gpl-download-fpbx, gpl-download-contrib, gpl-install-fpbx, and gpl-install-contrib. As the names suggest, the gpl-module-list-full provides a simple way to list ALL FreePBX GPL modules (Sangoma-produced and contributor-produced) which are available for download from GitHub. And that will be your starting point whenever you wish to install or upgrade a FreePBX module in Incredible PBX 13-13. You will note that modules fall into one of two categories: fpbx or contrib. The reason is because Sangoma has chosen to store the modules in two separate user accounts on GitHub. So you first must decipher which repo houses the module you wish to download or install. Once you’ve figured that out, you can choose to either manually download and install the module (gpl-download) or automatically download and install the module (gpl-install). The syntax is simple. Use either /root/gpl-install-fpbx modulename for Sangoma-produced modules or /root/gpl-install-contrib modulename for contributor-produced modules.

Cautionary Notes. Updating a few FreePBX modules may cause problems with Incredible PBX 13-13 because of modifications that were made when the distribution was initially developed. So steer clear of the modules blocked by the install script. Aside from those few exceptions, the remaining modules should work well without causing any problems. Be advised that you should always update the framework module before attempting to update the core module. We’ve tested all of the Sangoma releases shown below, but that’s not to say something can’t come unglued down the road. With contributor-produced modules, Sangoma doesn’t vouch for them, and neither do we. HINT: It is ALWAYS a good idea to make a good backup of your server before venturing into uncharted territory.


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Adding a few FreePBX modules may reset the Module Signature Checking flag. If this should happen to you, simply reset the Enable Module Signature Checking flag to NO in Settings: Advanced Settings. If this doesn’t fix the issue, issue the following command after logging into your server as root: /root/sig-fix

Taking the GPL Toolkit for a Spin. There have been some reported bugs in a few of the default Sangoma-produced modules in the Incredible PBX 13-13 build so let’s tackle those to demonstrate how easy this new upgrade process actually is. Using the gpl-module-list-full script, we would have deciphered the names of the modules we wanted to upgrade as superfecta, bulkhandler, and phonebook. So, after logging into your server as root, issue these gplinstall commands:

/root/gpl-install-fpbx superfecta
/root/gpl-install-fpbx bulkhandler
/root/gpl-install-fpbx phonebook

Where To Go From Here? The next step in your adventure should be to learn a bit about each of the available GPL modules for FreePBX including when each module was last updated. You can do that by visiting Sangoma’s FreePBX repo on GitHub and the FreePBX contributor’s repo on GitHub. We’ve always found it’s a smart idea to build a second Incredible PBX 13-13 server with either VirtualBox or an inexpensive cloud facility to use as a sandbox for experimentation. There are lots of Nerd Vittles articles to show you how. Then you don’t have to worry about damaging your production server until you first have verified that the upgrades don’t introduce problems of their own. Enjoy!

Originally published: Thursday, June 14, 2018  Updated: Friday, August 17, 2018


blankSupport Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.


blank
Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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.

blankBOGO 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.

blankThe 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.

blankVitalPBX 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!
 

blankSpecial 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…

VoIP 101: Developing a Cost-Effective SIP Strategy

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In the lead up to the demise of Google Voice XMPP service next week, we wanted to offer what we have found to be a cost-effective SIP strategy which takes advantage of the best of all worlds. We would divide SIP offerings into five broad categories: business-class unlimited SIP trunks, Old Faithful SIP providers, Mom-and-Pop SIP services, dirt-cheap termination services, and Gee Whiz SIP providers. As we have said many times, the beauty of setting up an Asterisk® PBX such as Incredible PBX® is you don’t have to put all your VoIP eggs in one basket. In our particular case, that has included a mix of Google Voice trunks plus all five of the SIP categories above. Today we want to document why we’ve personally made the selections we’ve made and hope that it provides a roadmap for your own VoIP setup while encouraging you to venture out of your safe zone and try some new VoIP options.

The all-you-can-eat business plans, which we previously have covered, make little sense for most home and small business users. Then there are the rock-solid, long term pay-as-you-go providers such as Vitelity and CallCentric that make perfect sense as your primary DID and SIP provider. While they may not always be the cheapest VoIP providers, the tradeoff is dependability and long-term reliability for your VoIP platform. In the case of Vitelity, it turns out the Nerd Vittles DID special (detailed below) from our Platinum Sponsor is perhaps one of the best VoIP deals on the planet.

The third category of SIP providers and our personal favorite is what we would call the mom-and-pop providers. These are typically one or two-person operations that offer incredible deals on all-you-can-eat VoIP plans for home users. Included in this category are Vestalink (available to existing customers only), Future-Nine and CircleNet. VestaLink originally began as OBiVoice and morphed over trademark issues. While the service is no longer available to new customers, it remains the best bargain at $72 for two years of unlimited inbound and outbound residential calling services. A close second goes to Future-Nine and their "Future 5 Grey" plan which provides 1,500 inbound and 1,500 outbound minutes a month for only $5. You can sign up here. Be sure to read the Terms of Services carefully, especially item #18. The New Kid on the Block is CircleNet. In addition to very attractive pay-by-the-minute offerings of $.005 per minute to most of the U.S. and Canada, they also have an $8 a month all-you-can-eat plan for residential customers that includes a very reasonable 5,000 minutes a month for calls to the following countries: United States, Canada, Australia, Bangladesh, Belgium, Brazil, Chile, Cyprus, Denmark, Finland, France, Germany, Greece , Guam, Hungary, India,Ireland, Italy, Japan, Latvia, Mexico, Netherlands, New Zealand, Norway, Poland, Puerto Rico, Singapore, Spain, Sweden, Taiwan, Thailand, United Kingdom, and Vatican City. Just let them know that you plan to use it with an Asterisk-based PBX. CircleNet also is offering Nerd Vittles readers a free month of the $8/month service to kick the tires. Simply send an email to sales@circlenet.us with your valid email address to take advantage of the offer. One free trial per customer/email address. CircleNet also offers a $15 a month business plan with even more minutes.

A fourth class of VoIP providers is the dirt-cheap termination services including Anveo Direct, TelecomsXchange, V1VoIP and the Betamax companies for low-cost international calling. These providers make terrific additions for supplementing your other VoIP services. TelecomsXchange is our personal favorite because of the special deal they have extended to Incredible PBX users. You get access to 300 VoIP wholesalers and can read about their services in this Nerd Vittles article. V1VoIP also has some terrific deals with 15¢/mo. DIDs from 13,000 Rate Centers and incoming and outgoing U.S. call pricing as low as $.003 per minute (not a typo!). Anveo Direct was perhaps the first provider to offer wholesale pricing to consumers, and they remain a terrific service both for DID and origination services with T.38 fax support as well as many of the lowest cost SIP terminations worldwide featuring user-configurable least-cost routing. Check out their pricing and rates here.

Finally, there are the SIP providers such as VoIP.ms that offer a rich collection of special features that you won’t find in many places and certainly not under the same roof. These features include SMS messaging, SIP URI proxying and iNUM for free worldwide calling, and fax support. Every one of these features is free when you sign up for an account at VoIP.ms. We encourage you to take advantage of these little known free services to enhance your PBX.

Putting It All Together. Now that we’ve covered the options, let’s go over how we would actually implement this. For the inbound trunk and primary DID, we’d recommend a SIP trunk from either Vitelity, VoIP.ms, or CallCentric. If you have multiple, simultaneous inbound calls, then the Nerd Vittles Vitelity special below can’t be beat because it provides four call paths. In addition, you get SMS support on the same trunk. Many people now assume your primary number supports SMS. We actually get dozens of unsolicited SMS messages on our home number from schools, churches, and political groups. If incoming call volume isn’t an issue, then VoIP.ms and CallCentric also offer a free iNUM number for your account. And VoIP.ms throws in a SIP URI as well.

For outbound calling for home and SOHO deployments, we recommend at least one of the mom-and-pop, all-you-can-eat providers: Future-Nine or CircleNet. If international calling is a requirement, you can’t beat the CircleNet offering. In addition to using your primary incoming provider, we also recommend you set up SIP accounts with a couple of the dirt-cheap termination providers. These don’t cost you anything other than a modest deposit unless you actually use them to place calls. And, when your primary outbound service has an outage, your PBX will never miss a beat.

The icing on the cake always has been several Google Voice trunks which work well for IVRs, Stealth AutoAttendants with DISA support, and faxing. While this may change with the demise of XMPP support, it appears that Bill Simon’s SIP Gateway to Google Voice will live on. With the Nerd Vittles sign-up link, you can migrate your existing Google Voice XMPP connections to the Simonics gateway for $4.99 each should the need arise. Enjoy!

Originally published: Monday, June 11, 2018


CircleNet SIP Setup for FreePBX/IncrediblePBX/VitalPBX/Issabel:

username=acct-id
type=friend
trustrpid=yes
sendrpid=yes
secret=acct-pword
qualify=yes
nat=yes
insecure=port,invite
host=sip.circlenet.biz
fromuser=acct-id
context=from-trunk
disallow=all
allow=ulaw

Registration String: acct-id:acct-pword@sip.circlenet.biz:5060/did-num

Future-Nine SIP Setup for FreePBX/IncrediblePBX/VitalPBX/Issabel:

username=acct-num
type=friend
trustrpid=yes
sendrpid=yes
secret=acct-pword
qualify=yes
nat=yes
insecure=port,invite
host=incoming.future-nine.com
fromuser=acct-num
context=from-trunk
canreinvite=no
disallow=all
allow=ulaw

Registration String: acct-num:acct-pword@incoming.future-nine.com/acct-num

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Need help with Asterisk? Visit the PIAF Forum.


 

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.

blankBOGO 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.

blankThe 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.

blankVitalPBX 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!
 

blankSpecial 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…

Some Fresh CallerID Magic for Incredible PBX 13-13

blank It’s been more than 10 years since we first introduced CallerID Trifecta for Asterisk® and the FreePBX® platform. A few months later it morphed into CallerID Superfecta and, as they say, the rest is history. Today CallerID Superfecta is used by over a million people around the globe to obtain CallerID Name (CNAM) information from over 70 different lookup sources. WOW! Just call me the Proud Papa. What a journey it has been, and our special thanks to the dozens of contributors.

Unfortunately, for those in the United States, most of the lookup sources now are commercial enterprises much like the original Bell Sisters that monopolized CallerID information by keeping it proprietary instead of forwarding CallerID Name data together with the CallerID number of incoming calls. In fact, for many users, the cost of CallerID lookups now is more expensive than the cost of the VoIP trunks carrying the calls. While we can’t solve the problem, we can make the cost of these CallerID lookup services considerably cheaper using a neat little trick with CallerID Superfecta.


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Here’s the problem. Many cell phone users don’t subscribe to any service that keeps track of the caller’s identity. The same is true for those using services such as Google Voice. When you receive a call from one of these users, the CNAM provided for the caller will either show something like "Mobile Phone User" or the city and state of the caller’s cellphone provider. This doesn’t keep the commercial CNAM providers from dinging you every time you receive one of these calls. And if your girlfriend calls you a hundred times a month, the commercial CNAM providers charge you for a hundred lookups even when the calls are from the same caller. That ends today. Once you implement this new CallerID add-on, you’ll only get billed for a single CNAM lookup regardless of how many times that person calls your PBX. Better yet, with a little effort, you can quickly replace the CNAM entries of the mystery cellphone and Google Voice callers by inserting their actual names into your Asterisk phonebook using either the FreePBX GUI or the Asterisk CLI. We’ll show you how.

Here’s the trick. CallerID Superfecta lets you prioritize source lookups when incoming calls hit your PBX. For example, you can lookup the CallerID number in AsteriDex and the Asterisk Phonebook in a fraction of a second at no cost. Once there’s a match on the CallerID number of the caller, additional lookups stop. So what we want to do is prioritize the lookups so that the free sources are used before any commercial lookup service is triggered. For the commercial CNAM source in the U.S., we’re going to use OpenCNAM in our examples because their service is one of the easiest and least expensive to implement. All OpenCNAM lookup options are less than a half a cent per lookup. By placing OpenCNAM third in the Default Superfecta lookup sequence, it will only be used when there is no match in either AsteriDex or the Asterisk Phonebook.


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Getting Started. It would obviously be a hassle to have to add every caller to one of these phonebooks every time a call arrives on your PBX. So the other half of today’s magic is to automatically add an entry to the Asterisk Phonebook every time a call hits your PBX. Then, the next time you receive a call from that same number, CallerID Superfecta will be able to find it in your free databases instead of querying OpenCNAM for yet another commercial lookup. As a general rule, in the United States, phonebook data is not copyrightable. However, you should consult an attorney for questions regarding the legality of this methodology as it applies to any particular CNAM lookup service.

We’re aware that CallerID Superfecta has had a caching option which provides similar functionality; however, users have reported mixed results and data corruption with that feature so we’ll be implementing a slightly different solution that relies on Asterisk’s extremely reliable SQLite3 database. This also provides the flexibility to customize the entries after completion of a call for more accurate CNAM information on subsequent calls. Unfortunately, a CNAM entry of ATLANTA, GA doesn’t tell you much about the caller even though you paid for the lookup. By inserting these call entries in the Asterisk Phonebook, you then have the option of easily customizing them in multiple ways. Not only will this provide better CNAM data on subsequent inbound calls, but the phonebook can also be used to make outgoing calls and to produce an alphabetized phonebook listing as well.

cd /var/lib/asterisk
sqlite3 astdb.sqlite3 "select key,value from astdb where key LIKE '%cidname/%' order by value"

Upgrading Asterisk Phonebook Module. Before we get too deep in the weeds, let’s get your Asterisk Phonebook module in FreePBX upgraded to resolve a bug in the version that ships with Incredible PBX which won’t let you edit phonebook entries without also adding a speed dial number for each entry. For generic FreePBX platforms, you can use Module Admin to upgrade the Phonebook module. For Incredible PBX platforms, log into your server as root and issue these commands:

cd /root
./gpl-install-fpbx phonebook

Activating CallerID Superfecta. There are three steps in activating CallerID Superfecta. First, configure the Default Scheme for CallerID Superfecta in Admin:CID Superfecta using the settings we previously displayed. Use the Arrow keys to properly order the three lookup sources: AsteriDex, Asterisk Phonebook, and OpenCNAM.


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Be advised that there is a credentials mismatch in the Default setup for AsteriDex. To correct this, open the Default configuration for Superfecta and click on the Tool icon to the left of AsteriDex. The username must be root, and the password is passw0rd with a zero. Save your settings.

Next, you need to create and fund an account at OpenCNAM.com. Once your account is established, copy your credentials and insert them into the OpenCNAM lookup source by clicking on OpenCNAM’s tool icon blank:


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Once you have saved the Default Scheme, then you need to enable CallerID Superfecta for each of your Inbound Routes under the Other tab in Connectivity:Inbound Routes:


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Adding Hook for Inbound Call Processing. One of the decade-old shortcomings of FreePBX has always been the inability to insert dialplan code into the Inbound Call process without forever damaging your ability to add additional DIDs moving forward. While extensions_override_freepbx.conf lets you extract and manipulate large chunks of FreePBX code, the drawback is that once you do that, the flexibility provided by the GUI goes up in smoke unless you remember to always move the newly generated dialplan code back into the override context.

We much prefer a simpler way of inserting custom dialplan code into the Inbound Call processing routine, but it requires a change in the FreePBX Core module. We would caution you about using this except on Incredible PBX 13-13 servers running under CentOS 6 or 7 or Ubuntu 18.04 with Signature Checking Disabled. It will trigger nasty signature checking errors on the FreePBX Dashboard. See the next section for a workaround on generic FreePBX 13 platforms. On Incredible PBX 13 servers, login as root and issue the following commands making sure that you expand the last two lines into a single command before executing it:

cd /var/www/html/admin/modules/core
sed -i "s|new ext_noop('CallerID Entry Point')|new ext_gosub('1','s','sub-log-caller')|" \\
functions.inc.php
mysql -u root -ppassw0rd asterisk -e "update admin set value='true' \\
where variable='need_reload'"
amportal a r

Next, we need to add some supplemental dialplan code to extensions_custom.conf. This code actually populates the Asterisk Phonebook with the new call entries each time a call is received by your PBX:

cd /tmp
wget http://incrediblepbx.com/sub-log-caller.tar.gz
tar zxvf sub-log-caller.tar.gz
rm -f sub-log-caller.tar.gz
cd /etc/asterisk
cat /tmp/sub-log-caller.txt >> extensions_custom.conf
fwconsole reload
amportal restart

Disabling Module Signature Checking. If you’re running FreePBX 13 behind a secure firewall that blocks web access to FreePBX for anonymous users, then here’s a simple way that you can apply the patch above and take advantage of the sub-log-caller functionality without having to endure module signature errors within FreePBX. Simply disable Module Signature Checking. Here’s how:

cd /var/www/html/admin/libraries
sed -i 's|"SIGNATURECHECK", true|"SIGNATURECHECK", false|' modulefunctions.class.php
mysql -u root -ppassw0rd asterisk -e "UPDATE notifications SET candelete =  '1' WHERE  true;"
mysql -u root -ppassw0rd asterisk -e "DELETE FROM notifications WHERE true;"
mysql -u root -ppassw0rd asterisk -e "UPDATE freepbx_settings SET value='0' WHERE \\
keyword='SIGNATURECHECK';"
amportal restart

Adding CNAM data to CDR Listings. One of our concerns with the current CDR implementation in FreePBX is the lack of display of CNAM data for most entries in CDR listings. Currently, the only way to decipher the CNAM information is to use your mouse to hover over each of the CallerID numbers. We much prefer displaying both CallerID Name and Number in the CallerID column of CDR Reports. Here’s how to fix it:

cd /var/www/html/admin/modules/cdr
sed -i "s|cdr_formatSrc($row\['src'], $row\['clid']);|cdr_formatSrc($row\['clid'], \\
$row\['clid']);|" page.cdr.php
amportal a r

Managing Your Asterisk Phonebook. As we mentioned, there are a couple of ways to display and/or update entries in your Asterisk Phonebook that have been populated with OpenCNAM lookups. From the Asterisk CLI, you can display entries like this: database show cidname.

You also can add or update entries using the Asterisk CLI like this:

database put cidname 8431234567 "Mundy, Ward"

The simplest method to manage your Asterisk Phonebook is from within the FreePBX GUI. Simply navigate to Admin:Asterisk Phonebook. There you can edit individual entries and replace generic CNAM entries such as Atlanta, GA within something more descriptive of the caller’d identity. You also have the ability to export the data and even create speed dial entries for frequently called parties. Once Speed Dial numbers have been created, you can reach the party from any phone connected to your PBX by dialing *0 plus the speed dial number.

Finally, you need to complete some additional steps to implement the Spam Call Blocker as documented in this tutorial. Enjoy!

Originally published: Wednesday, May 30, 2018  Updated: Monday, November 26, 2018


blankSupport Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.


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Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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.

blankBOGO 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.

blankThe 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.

blankVitalPBX 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!
 

blankSpecial 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…

Dare to Compare: The Best (free) VoIP Offerings for 2018


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Last week we showed you how to get 10 months of free hosting for your Incredible PBX® in the Cloud. And today we present our semi-annual survey of the latest and greatest VoIP offerings for 2018. The beauty of the cloud platform is you can try all of them for less than a penny an hour and decide for yourself which free offering best meets your needs. This year we’ve ushered in new Asterisk® 13 LTS releases of Incredible PBX® on the CentOS, Ubuntu, and Raspberry Pi platforms as well as new versions for Issabel 4 and VitalPBX. To sweeten the pot even further, we nailed down a new Cloud-based offering for $10 a year that makes a perfect VOIP sandbox for our CentOS platform. For 2018, we also secured new (free) DID offerings in the U.S. and announced a Nerd Vittles exclusive providing access to 300+ VoIP providers worldwide, all at wholesale prices. And, last but not least, we introduced Digium’s newest IP phones for Asterisk including a $59 model that makes a perfect VoIP companion.


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Choosing the Best VoIP Platform for Your Needs

Choosing a VoIP platform is partially a subjective decision, but there also are some glaring red flags to consider. We suggest you begin by deciding whether your preferences include any must-have’s. Do your requirements mandate an open source solution? Do you need text-to-speech and voice recognition? Does the operating system have to be Linux-based and, if so, must it be CentOS, Debian, or Ubuntu? If you’ll be using SIP phones, must the platform include phone provisioning software for your phones, or is the ability to purchase it as an add-on sufficient? Is paid support important in making your platform decision and how much are you prepared to pay? Are automatic or pain-free software updates critical in making your selection? Is migration from an existing platform a factor? Does a preconfigured, secure firewall matter, or are you prepared to do it yourself or take your chances? Before choosing to ignore security, read this RIPS analysis of FreePBX®. Here’s a snippet from the article. Read it carefully. It’s your phone bill.

Since FreePBX is written completely in PHP, we decided to throw it into our code analysis tool RIPS. The results were more than surprising and should tell you why a rock-solid firewall is absolutely essential.

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The total amount of detected vulnerabilities is very high. Luckily, the majority of the detected vulnerabilities are inside the administration control panel, such that attackers either need to steal a valid account or they have to trick an administrator into visiting a malicious website that triggers one of the critical vulnerabilities. For example, a remote command execution vulnerability could be triggered by a less critical cross-site scripting vulnerability. By chaining both vulnerabilities, the severity is increased drastically and can lead to full server compromise.

In choosing which platforms to include today, we eliminated platforms which we considered too complicated for the average new user to configure. We also eliminated any platform that did not offer at least a free tier of service with a reasonably complete feature set as part of their offering. So here’s our Pick of the Litter.

We must confess that we are partial to the Incredible PBX offerings because they provide a turnkey GPL platform with minimal configuration required on your part. Regardless of platform, all come standard with a preconfigured firewall and about three dozen applications for Asterisk that will help you learn everything there is to know about VoIP telephony.

VoIP Platform Feature Summary

Aggregation: Incredible PBX 13-13 for CentOS/SL
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 13 GPL modules
O/S: CentOS/SL 6.9 or 7
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic Update Utility included
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall Whitelist
Security Rating (as delivered): Secure
Comments: Lean & Mean or Whole Enchilada installers as well as ISO available

Aggregation: Incredible PBX 13-13 for Raspbian
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 13 GPL modules
O/S: Raspbian 7
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic Update Utility included
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall Whitelist
Security Rating (as delivered): Secure

Aggregation: Incredible PBX 13-13 for Ubuntu
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 13 GPL modules
O/S: Ubuntu 18.04
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic Update Utility included
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall Whitelist
Security Rating (as delivered): Secure
Comments: Lean & Mean or Whole Enchilada installers

Aggregation: VitalPBX
License: Closed Source
VoIP Platform: Asterisk 13
GUI: Free and Commercial modules
O/S: CentOS 7
Phone Provisioning: Free
Text-to-Speech/Voice Recognition: Optional/Optional
Software Updates: Automatic
Migration Tools: Yes
Security: Fail2Ban + User-Configurable Firewall
Security Rating (as delivered): Insecure
Comments: Incredible PBX add-on now available including TM3 firewall.

Aggregation: Incredible PBX for Issabel 4
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 11 GPL modules
O/S: CentOS 7
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: No/No
Software Updates: Semi-Automatic
Migration Tools: No
Security: Fail2Ban + Unconfigured Firewall
Security Rating (as delivered): Secure with Incredible PBX add-on
Comments: Incredible PBX add-on provides secure platform

Aggregation: FusionPBX for FreeSWITCH
License: Open Source MPL 1.1
VoIP Platform: FreeSWITCH 1.6
GUI: FusionPBX
O/S: Debian 8
Phone Provisioning: Free
Text-to-Speech/Voice Recognition: Optional/Optional
Software Updates: Automatic
Security: Fail2Ban + User-Configurable Firewall
Security Rating (as delivered): Secure with mods below
Comments: Incredible PBX firewall add-on now available .

Aggregation: Incredible PBX for Wazo
License: GPL3 Open Source
VoIP Platform: Asterisk 15 RealTime
GUI: Wazo GPL3 modules
O/S: Debian 9
Phone Provisioning: Extensive Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic or 2-minute Manual
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall
Security Rating (as delivered): Secure WhiteList with Incredible PBX add-on
Comments: High Availability & Call Center GPL3 Modules

Aggregation: FreePBX Distro a.k.a. AsteriskNOW
License: Closed Source
VoIP Platform: Asterisk 13/14/15
GUI: FreePBX GPL and Commercial modules
O/S: Closed-source CentOS fork
Phone Provisioning: Open Source (minimal) or Commercial
Text-to-Speech/Voice Recognition: Optional/No
Software Updates: Manual from Hidden Repo
Migration Tools: Yes
Security: Fail2Ban + User-Configurable Firewall
Security Rating (as delivered): Insecure
Comments: Extensive commercial NagWare preinstalled

 

Deploying a Local Server vs. Cloud Platform

We’ve always been big fans of local servers because you have almost total control of your own destiny. This was especially true when the Raspberry Pi came along to take the financial pain out of the server equation. But the price of Cloud-based servers has continued to plummet. For 2018, you can run any of our favorites on the least expensive platform at Vultr or Digital Ocean for $2.50 a month. And, if you hurry, your first 10 months are free at Vultr. Spending another 50 cents buys you automatic backups.1 And, for the Incredible PBX 13-13 build with CentOS 6.9 (64-bit), we’ve found a deal at HiFormance that offers a high-performance OpenVZ platform at an annual cost of just $10. The technical specs are impressive (even better if you sign up for 3 years), and we don’t think you’ll find a comparable deal with anything near comparable performance and specs anywhere, period. You get your choice of hosting sites including New York, Chicago, Los Angeles, Buffalo, Atlanta, and Dallas. Complete tutorial available here.

NOTE: OpenVZ/SolusVM platforms not suitable for CentOS 7, Debian 9, or Ubuntu 18 implementations, and some providers do not yet support Ubuntu 18.04 platform although Vultr and Digital Ocean both do.

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Available Free Trunks for VoIP Servers

For many years, we’ve offered free Google Voice connectivity with our VoIP platforms. And that remains true at least for a few more weeks. On all of the Incredible PBX platforms, Google Voice trunks can be set up to make free calls in the U.S. and Canada provided you have a U.S. residence and a U.S. cellphone number to verify that you are who you say you are. There’s even a ray of hope that the Simonics gateway may allow you to continue using Google Voice after Google Voice’s mid-June drop-dead date for XMPP. Details here. But what about the rest of the world. For 2018, we solved the problem by offering free DID trunks for inbound calls and a collection of 300 wholesale VoIP carriers worldwide to make outbound calls at the same wholesale rates offered to the very largest resellers. Simply pay a 13% surcharge in lieu of the $650 annual fee, and TelecomsXchange (TCXC) will provide you access to their entire suite of wholesale carriers together with state-of-the-art tools to manage all of the services.2 The Nerd Vittles setup tutorial is available here. Enjoy!

Published: Monday, March 5, 2018  Updated: Sunday, May 27, 2018


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Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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.

blankBOGO 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.

blankThe 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.

blankVitalPBX 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!
 

blankSpecial 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…

  1. On the Vultr and Digital Ocean $2.50 platforms, be sure to (1) create a 1GB swapfile once you’ve chosen your operating system. (2) Then, for Vultr, issue the following command before beginning the Incredible PBX install: apt-get install cloud-init.
    (3) Now complete the steps outlined in your preferred Nerd Vittles tutorial, and you’ll be all set in about 15 minutes. []
  2. Our special thanks to TelecomsXchange. They have generously offered to contribute a portion of the wholesale surcharge to support the Incredible PBX open source project. []

Creating an OBi200 Google Voice Trunk to Use with Asterisk

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Since Asterisk® will no longer be able to "talk" to Google Voice after June 17, we promised to hold our nose and document how to salvage your Google Voice trunks. Our exercise for today is to show you how to deploy an OBi 200-series device which can speak the new Google Voice language and use it as a traditional SIP bridge between Google Voice’s proprietary SIP platform and your Asterisk server. We will skip the editorializing on why Google is making a terrible mistake by discarding XMPP and forcing users to a proprietary solution necessitating a hardware purchase without first offering an open standards solution as Google’s Community Manager promised here. Promises, of course, don’t keep your phones ringing. For the whole story, see our article from last Saturday. For today, you’ll need to shell out $50 for an OBi 200 device. Once you have it in hand, feel free to read on and we’ll get you back in business. For security reasons, it should be noted that today’s setup assumes you are running an Incredible PBX® server and OBi device locally behind a NAT-based router. This will work equally well with the Incredible PBX-enhanced versions of Issabel and VitalPBX. We’ll leave it to the FreePBX® folks to figure out a solution for their proprietary distro.

Everything we’re covering below will work just as well using any of the OBi 200-series devices. We’ve simply chosen to use an OBi202 in our examples today because it supports an extra phone port. But an OBi200 works just as well if you will only be deploying Google Voice trunks (up to 3 and perhaps more) for your PBX. They retail for approximately $50 and are readily available at Amazon through the link in the right column which also provides a few shekels for Nerd Vittles to keep the lights on. As mentioned last week, Obihai crippled the OBi 110-series devices which will no longer work with the new Google Voice setup. Such a fine company that we once praised for producing our Device of the Year. And don’t worry. If you ever visit their forum, you can expect a cheery reception from the Obihai forum moderator. Here’s the response we got1 when raising concerns about the demise of Google Voice XMPP:


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Registering Your OBi2x Device with OBiTALK

A Quick Start Guide accompanies your OBi hardware. Following along in the tutorial will get your OBi set up using a free (so far) OBiTALK account. When you get to Step 5, you’ll be ready to set up your Google Voice account by clicking the Google Voice Set-Up button.

Before you begin the Google Voice setup, we strongly recommend that you plug a POTS phone into your OBi device and dial ***6 to update your firmware to the latest release. Depending upon where you purchased your device, it may or may not have the latest firmware which is required to communicate with Google Voice on or after June 17.

We also recommend that you dial ***1 and obtain the DHCP-assigned IP address for your OBi. You’ll need this in a few minutes. And, while you’re at it, be sure to set the OBi up behind a NAT-based router to protect it from intrusion. Once someone gains access to your OBi, they’ve essentially got the keys to your telecom castle. So always deploy an OBi behind a hardware-based firewall that is on the same private LAN as your Asterisk PBX. Finally, on your router, be sure to reserve the DHCP-assigned IP address of your OBi for permanent use by the OBi hardware. Otherwise, the IP address of your OBi may change, and this will break the SIP gateway connection to your Asterisk server.

Finally, a word about the new OBi setup. All of your settings are now stored and managed in the OBiTALK cloud. Obihai then pushes the configuration to your OBi device. To put it charitably, this usually works but sometimes it doesn’t, and you end up with a quirky OBi setup that looks correct in the cloud but simply doesn’t work. We’ve found the simplest solution is to unplug the device and then restart it. Then check all of your cloud-based settings when the OBi device comes back to life to be sure none of your settings disappeared. Sometimes they do! In the old days, you had the option of configuring your OBi device locally; however, Obihai (now Polycom) has disabled that functionality with the new Google Voice setup presumably to disguise what they are doing under the covers to connect to Google.

Configuring a Google Voice Trunk on OBi200

To give credit where credit is due, configuring a Google Voice trunk on the OBi 200-series devices is dead simple. Login into your OBiTALK account, click on your OBi device, and then click the Google Voice Set-Up button.

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Enter your Google Voice credentials when prompted, give Obihai permission to control your Google Voice account, and you’re done. Within a few seconds, the connections dialog box should show Google Voice connected on service provider SP1.

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If you haven’t already done so, plug a POTS phone into your OBi device and place a call to somebody by dialing a 10-digit number. Then use another phone and call the Google Voice number you assigned to your OBi device. The POTS phone should ring. Don’t continue until you get these calls working in both directions. You’d be wasting your time.

Now we need to adjust the destination for incoming calls to your OBi device and redirect them from the POTS phone to the SP3 trunk we’ll be using to connect to your Asterisk server. We’ll leave SP2 unoccupied in case you wish to add another Google Voice trunk down the road.

To make this change, click the OBi Expert Configuration button at the bottom of the Device Configuration window. Then click OK to confirm that you know what you’re doing. Next click the Enter OBi Expert button at the top of the next form. In the left column, click Voice Services and then SP1 Service. The fifth parameter is called X_InboundCallRoute. Beside it, uncheck both the OBiTALK Settings and Device Default checkboxes. Now enter sp3(6781234567) in the Value field for X_InboundCallRoute where 6781234567 is your actual Google Voice phone number (DID). Scroll to the bottom and click the Submit button.

Finally, at the top of the left column of the form, click Return to OBi Dashboard.

Configuring OBi SIP Trunk for Asterisk

1. Login to your OBi Dashboard using a web browser . After signing up for an account and registering your OBi device, click on the OBi 200 device in the My OBi Devices list.

2. In the Device Configuration dialog, click OBi Expert Configuration button. When prompted whether you’re sure, click OK.

3. In the OBi Expert Configuration Menu, click Enter OBi Expert button.

4. In the Production Information (left) column, click Service Providers.

5. In the Service Providers listing, click ITSP Profile C General.

6. For each of these fields, uncheck OBiTALK Settings and then uncheck Device Default:

  • General:Name
  • Service Provider Info:Name
  • Service Provider Info:URL

7. Fill in the ! field Values as shown below using the private IP address of your PBX:


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8. Click Submit button after checking your entries carefully.

9. In the Service Providers listing on the left, click ITSP Profile C SIP.

10. In the ITSP Profile, enter the private IP address of your PBX in the Proxy Server, Registrar Server, and Outbound Proxy fields after first unchecking both the OBiTALK Settings and Device Default checkboxes.

11. Scroll down the form to X_SpoofCallerID and uncheck both the OBiTALK Settings and Device Default checkboxes. Then check the Value field for X_SpoofCallerID.

12. Scroll down the form to X_DiscoverPublicAddress and uncheck both the OBiTALK Settings and Device Default checkboxes. Then uncheck the Value field for X_Discover PublicAddress.

13. Click Submit button after checking your entries beside the 5 red exclamation points.

14. In the Production Information (left) column, click Voice Services

15. In the Voice Services listing on the left, click SP3 Service.

16. In the SP3 Service Profile, fill in the 5 fields in which the OBiTALK Settings checkbox is unchecked. The AuthUsername and AuthPassword entries will be used to authenticate to your PBX so be sure to choose a very secure password. It’s your phone bill. The URI field actually makes the trunk connection to your PBX so replace the 192.168.0.82 entry shown with the actual IP address of your PBX.

17. In the SIP Credentials section of the form, make certain that X_EnforceRequestUserID is unchecked. If not, uncheck both the OBiTALK Settings and Device Default checkboxes and then uncheck X_EnforceRequestUserID.

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18. If you do not want to pass the CallerID number with your calls, in the Calling Features section of the form, be sure to check AnonymousCallEnable after unchecking both the OBiTALK Settings and Device Default checkboxes.

19. In the Service Providers listing on the left, click ITSP Profile A SIP.

20. Be sure X_SpoofCallerID is checked.

21. Click Submit button after checking your entries carefully.

Configuring Incredible PBX GUI for an OBi200

On the Incredible PBX side, log into the GUI using a web browser. We’ll be adding a SIP trunk, an outbound route, and an inbound route to process calls to and from the OBi device.

Add a SIP Trunk with a Trunk Name matching whatever you used in your OBi SIP credentials, e.g. obi200 or obi202. Plug in your Outbound CallerID to match your Google Voice phone number. In the Dialed Number Manipulation Rules tab, add a Match Pattern of NXXNXXXXXX. In the SIP Settings tab for Outgoing, the Trunk Name should match whatever you used on the OBi side, e.g. obi200 or obi202. In the PEER DETAILS, enter the following using the default username and password you assigned on the OBi side. Normally, port 5061 is the default port assigned on the OBi side. If you get a failed registration, try 5060 and then 5062 and 5063. Click Submit and reload your dialplan when finished.

type=friend
defaultuser=obi200
secret=your-password
qualify=yes
port=5061
nat=yes
host=dynamic
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
insecure=port,invite

For Outbound Call Routing, we recommend an Outbound Route using the 624 (OBI) prefix and 10-digit numbers. For example, if a user dials 624-888-1234567, your Incredible PBX server would place a call using the OBi’s Google Voice trunk to 1-888-1234567. When your Outbound Route setup looks like the following, click Submit and reload your dialplan.


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For Inbound Call Routing, create an Inbound Route specifying a DID Number to match your Google Voice number. Choose a Call Destination to meet your own requirements, e.g. an extension, ring group, or IVR. Then click Submit and reload your dialplan.

Now you’re ready to test an outgoing call by dialing the OBi prefix (624) plus a 10-digit number. Then place a call to your Google Voice number using your cellphone and be sure Asterisk routes it to the destination you specified in your inbound route above.

Configuring VitalPBX to Use an OBi200

Truth be told, we weren’t bright enough to figure out how to configure the VitalPBX Trunk using credentials so we simply set up the SIP trunk using IP address authentication with the IP address of the OBi device. It works just as well and just goes to prove there’s always more than one way to skin a cat. So here’s the Trunk configuration on the VitalPBX side. The only entry you will need to change is the Host IP address for your OBi device. If you don’t know it, plug a phone into the OBi and dial ***1.

NOTE: For the Username and Description fields below, be sure to match what you used on the OBi side (above) for your SIP credentials, i.e. obi200 or obi202. If they don’t match on both devices, you won’t get a successful connection. Our apologies for mixing apples and oranges in the screenshots.


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For Outbound Call Routing, we recommend an Outbound Route using the 624 (OBI) prefix and 10-digit numbers. For example, if a user dials 624-888-1234567, the VitalPBX server would place a call using the OBi’s Google Voice trunk to 1-888-1234567. Here’s the Outbound Route setup to make that happen:


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For Inbound Call Routing, go to PBX:External:Inbound Routes and add an inbound route and destination for calls from your 10-digit Google Voice number. Or you can use the Default Inbound Route which we explained in our previous VitalPBX tutorial. Basically, you set up an Inbound Route with a Description and Routing Method of Default. All the other fields should be left as is except for the Inbound Destination. For the destination, you can choose an IVR, Extension, Ring Group, etc. to meet your own requirements.

Originally published: Monday, May 14, 2018


blankSupport Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.


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SPECIAL TREAT: If you could use one or more free DIDs in the U.S. with unlimited inbound calls and unlimited simultaneous channels, then today’s your lucky day. TelecomsXChange and Bluebird Communications have a few hundred thousand DIDs to give away so you better hurry. You have your choice of DID locations including New York, New Jersey, California, Texas, and Iowa. The DIDs support Voice, Fax, Video, and even Text Messaging (by request). The only requirement at your end is a dedicated IP address for your VoIP server. Once you receive your welcome email with your number, be sure to whitelist the provider’s IP address in your firewall. For Incredible PBX servers, use add-ip to whitelist the UDP SIP port, 5060, using the IP address provided in your welcoming email.

Here’s the link to order your DIDs.

Your DID Trunk Setup in your favorite GUI should look like this:

Trunk Name: IPC
Peer Details:
type=friend
qualify=yes
host={IP address provided in welcome email}
context=from-trunk

Your Inbound Route should specify the 10-digit DID. Enjoy!


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Need help with Asterisk? Visit the PBX in a Flash Forum.


 

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.

blankBOGO 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.

blankThe 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.

blankVitalPBX 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!
 

blankSpecial 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.
 



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  1. You can always find a little humor in insults if you dig deep enough. Ironically and unbeknownst to our pal, Steve, it was Sherman Scholten and his OBi development team that were among the first Google Voice "freeloaders." Only years later after Google Voice was integrated into FreeSwitch did Josh Culp at Digium perfect a clean way to integrate Google Voice into the Asterisk platform. []

Introducing Digium’s Awesome SIP Phones for Asterisk


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If you’ve been waiting for a low-cost, feature-rich SIP phone that meshes perfectly with your Asterisk® PBX, your prayers have been answered. Digium has just released not one, but four, new SIP phones with prices starting at $59. No, that’s not a typo. Digium gave us a couple of early models to play with, and today we’ll walk you through the incredibly simple setup. We would begin by noting that, despite the pricing, these phones are configured with nothing resembling a bargain basement feature set. All four models have color displays, HD Voice, POE for use without the $15 power adapter, and at least two lines. The phones can be configured using the phones themselves, or through a slick web interface, or with auto-provisioning by MAC address. Beginning with the $89 A22, the top three models support gigabit Ethernet. With the $119 A25, you get four line registrations as well as a second LCD supporting six Rapid Dial keys or up to 30 BLF entries. The top-of-the-line $169 A30 supports six line registrations and an LED setup that closely matches our previous VoIP Phone of the Year, Yealink’s T46G. While the phones were not designed for use with Switchvox®, we found them to be plug-and-play with 3CX® which is probably also true with Switchvox even though we have not tested them on that platform. We have been using our A22 phone with one line connected to Incredible PBX® for the Raspberry Pi and the second connected to VitalBox. We’ve had zero issues with the phone, and sound quality is excellent.


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Connecting Digium’s A-Series IP Phone

To get started, you’ll need a power source for the phone which can be either a POE network connection or a power adapter. You’ll also need to connect to a network that can provide DHCP or VLAN configuration data. Once the phone boots up, press the checkmark button (✓) twice to display the IP address assigned to the phone. Using a desktop browser, navigate to that IP address and enter admin:789 as the default login credentials.

Configuring a SIP Extension on Your IP Phone

Once you’re logged in, click on the Line tab and fill in the blanks for the SIP1 account using the desired extension number, extension password, and IP address of your Asterisk server. Be sure Activate is checked. It should look something like the following. Then click Apply.

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This one-minute setup is all that’s required to put your new phone into production with Asterisk. You’re ready to make and receive calls. The L1 button on the A20 or A22 phone (pictured above) should now be lit. To light up the L2 button, add a second SIP connection by repeating the drill after choosing the SIP2 Line from the pull-down menu. If you have redundant PBXs, fill in the IP address of the Backup server, and the phone will automatically failover when the primary PBX goes down. It doesn’t get any easier than that.

With 3CX extensions, the setup is virtually identical except the phone’s Authentication Name field should reflect the Authentication Name chosen when setting up the 3CX extension.

Customizing Your SIP Phone Settings

VoiceMail Setup. The voicemail button can be activated for one or both SIP lines in the Advanced Settings tab under each of the SIP connections. Check the Subscribe to Voice Message box and enter the Voice Message Number to retrieve your voicemails, e.g. *98701 for extension 701 on an Asterisk PBX or 999 for a 3CX extension’s voicemail.

Customizing Phone Display. If you’d like to customize the branding and background image on your phone, navigate to Phone Settings and click the Advanced tab. Here’s a link to download one of our favorite beach scenes (pictured above), or you can use your own 320×240 BMP image on the A20 and A22. The high end phones use a 480×272 BMP image. The Asterisk label at the top of the phone’s display can also be adjusted in the Greeting Words field. We’re Enchilada fans personally. 🙂

Changing Passwords and PINs. You also can adjust the passwords and PINs for the phone device itself under the Phone Settings:Advanced tab. The default is 789. To modify the admin credentials for the browser interface or to add new accounts, go to System and click on the Account tab. Because the phone can be configured using either the phone itself or the browser interface, you’ll need to change both sets of passwords to secure your phone.

Adjusting Codecs. Depending upon your PBX setup, you may need to adjust or reorder the codecs for one or both of your SIP lines. Simply navigate to Line:SIP1:Codec Settings and make any necessary changes. HINT: You’ll rarely have a problem if you make G.711U (U.S.) or G.711A (elsewhere) your primary codec although G.722 is what you’ll want for HD Voice. This is especially important if you’re using Google Voice trunks or 3CX client software.

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Auto-Provisioning Your A20, A22, and A25 Phones

Let’s get to the fun stuff now. Everything we’ve covered (and much more) can be scripted with these new phones. You can read all about it here. For today, let’s get your Phonebook Contacts populated using your AsteriDex database entries. And then you can press the Down button on the phone to retrieve your Contacts.

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Setting Up Phone Provisioning. Before you can auto-provision your phone, both your phone and your Asterisk server need a little navigation information. Let’s start with the phone so login as admin:789 to get started. Click on the System option and then the Auto Provision tab. Write down the last 12 digits of your phone’s MAC address (CPE Serial Number highlighted above). Check the DownloadDeviceConfig option (as shown). Disable the DHCP Option and the SIP Plug and Play options by clicking on the respective tabs. Then open the Static Provisioning Server option (as shown). Enter the local IP address of your server assuming your phone and server are both behind a firewall. For the Protocol Type, choose HTTP. For the Update Mode, choose Update After Reboot. Then click the Apply button.

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Next, let’s configure the phone so that you can press the Down arrow button to access your Phonebook Contacts. Click on the Function Key option in the left margin. Then look in the Programmable Keys section and locate the row with the settings for the Down button. Change the entry in the Desktop column to Phonebook. Then click the Apply button.

Configuring Asterisk for Phone Provisioning. Now we need to get your server set up to support phone provisioning. The way provisioning works is we will set up a provisioning profile for each phone which will be processed by your web server whenever a phone is rebooted. This profile will also tell the phone where to find your Phonebook Contacts XML file. To get started, navigate to /var/www/html and create a new .cfg file for each of your phones using the 12-character MAC address of the phone, e.g. 000123456789.cfg. The file should look like the following with the exception of the Auto Pbook Url entry which should reflect the local IP address of your server:

<<VOIP CONFIG FILE>>Version:2.0.0.0

<PHONE CONFIG MODULE>
LCD Title          :IncredblePBX

<AUTOUPDATE CONFIG MODULE>
Download CommonConf:0
Download DeviceConf:1
Check FailTimes    :5
update PB Interval :720
Clr PB B4 Import   :1
Trust Certification:0
Enable Auto Upgrade:0
Upgrade Server 1   :
Upgrade Server 2   :
Auto Upgrade intval:24
Auto Pbook Url     :http://192.168.0.108/phonebook.xml

<<END OF FILE>>

Populating Phonebook Contacts with AsteriDex. Now we’re ready to build the Phonebook Contacts file (phonebook.xml) using the AsteriDex 4 database. Just issue the following commands and then reboot each of your phones (Menu+8+Yes):

cd /var/www/html/asteridex4
wget http://incrediblepbx.com/asterisk-phonebook.tar.gz
tar zxvf asterisk-phonebook.tar.gz
rm -f asterisk-phonebook.tar.gz
php asterisk-phonebook.php

Digium A-Series IP Phone User Guide

Last but not least, take a look at Digium’s A-Series IP Phone User Guide (PDF) for more tips.

Final Thoughts on A-Series IP Phones

If you couldn’t already tell, we’re quite impressed with the new A-Series phones from Digium. If you’re on a budget, the $59 model is one terrific bargain for home or SOHO use. The only thing you’re really forfeiting with this phone is the gigabit Ethernet port which will have zero impact on small and medium-sized network implementations of a VoIP server. Rather than buying power adapters for your phones, drop by your favorite WalMart and purchase a network switch that includes POE support. They start at about $30. Then pick one of these phones up from your favorite provider and let us know what you think. You’ll also be helping to fund Digium’s open source Asterisk project. Enjoy!

Originally published: Friday, April 13, 2018




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Need help with VitalPBX? Visit the VitalPBX Forum.


 

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.

blankBOGO 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.

blankThe 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.

blankVitalPBX 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!
 

blankSpecial 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.
 



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