Cloud Computing – Page 12

Free Asterisk Voicemail Transcription with IBM Watson STT

Monday, November 12, 2018 / 2 Comments on Free Asterisk Voicemail Transcription with IBM Watson STT

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:

Why IBM's speech recognition breakthrough matters for AI and IoT. Via @techrepublic https://t.co/AJi8MA3E20

— IBM Developer (@IBMDeveloper) March 15, 2017

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

Got Friends? 7 Countries Have Never Visited Nerd Vittles. 2018 Is Calling! https://t.co/wMfmlhAr16 #asterisk #freepbx #wazo #issabel #IncrediblePBX #3CX pic.twitter.com/kAmAEnwVIw

— Ward Mundy (@NerdUno) January 9, 2018

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.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Some Recent Nerd Vittles Articles of Interest…

FusionPBX on Steroids: Text-to-Speech Apps Have Arrived

Monday, September 24, 2018

SECURITY ALERT: https://securityboulevard.com/2019/06/rce-using-caller-id-multiple-vulnerabilities-in-fusionpbx/

And you thought you needed an Asterisk® PBX for your users to enjoy FREE text-to-speech applications such as current News Headlines and Weather reports from the convenience of their telephone. Well, move over Asterisk. FusionPBX™ for FreeSWITCH™ now offers virtually identical functionality with all of the terrific advantages that FusionPBX provides: reliability, updates, performance, security and an unmatched UC platform with no rivals. To get started, make sure you have completed the steps in our FusionPBX introductory tutorial.

Intuitive support in FusionPBX for interactive TTS or STT applications is not (yet) available. So we’re doing the next best thing. Once or more a day, we will use cron jobs to retrieve the latest News Headlines and Weather reports for your local area. Then anyone using your PBX can pick up a phone and listen to the News Headlines by dialing 951 or U.S. weather forecasts by dialing 947, or worldwide weather forecasts from ApiXU by dialing 949.¹ We’ll be using IBM’s awesome TTS engine to handle the text-to-speech chores. We think you will agree that IBM’s offering is the best in the business. And you can’t beat the price. After your first free month, you get a million characters of FREE text-to-speech synthesis every month forever! For ApiXU worldwide weather data, your first 2,500 queries are also FREE every month.

Here’s a sample from the 3CX implementation of these identical applications:

[soundcloud url="https://api.soundcloud.com/tracks/364353344″ params="auto_play=false&hide_related=false&show_comments=true&show_user=true&show_reposts=false&visual=true" width="80%" height="350″ iframe="true" /]

Getting Started with IBM Bluemix TTS Service

NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:

You can start your free, 30-day trial of IBM Bluemix services without providing a credit card. Just sign up here. Once your account is activated, here’s how to obtain credentials for the TTS service to use with FusionPBX. Start by logging in to your IBM Bluemix account. Once you’re logged in, click on your account name (1) in the upper right corner of your web page to reveal the pull-down to select your Region, Organization, and Space. Follow the blue links at the bottom of the pull-down menu to create an Organization and Space for your TTS service.

Next, click the Menu icon which is displayed as three horizontal bars on the left side of the web page. Choose Watson. Click Create Watson Service and select Text to Speech from the applications listing. Watson will generate a new TTS service template and display it. Make certain that your Region, Organization, and Space are shown correctly. Then verify that the Standard Pricing Plan is selected. When everything is correct, click the Create button.

When your Text to Speech application displays, click Service Credentials and then click New Credential (+). When the Add New Credential dialog appears, leave the default settings as they are and click Add. Your Credentials Listing then will appear. Click View Credentials beside the new entry you just created. Write down your URL, username, and password. You’ll need these in Step #4 below to configure the IBM Bluemix TTS service. Logout of the IBM Cloud by clicking on the little face in the upper right corner of your browser window and choose Log Out. Confirm that you do, indeed, wish to log out.

Getting Started with ApiXU Weather

Finding free worldwide weather forecasts has been a difficult nut to crack. So we’re pleased to introduce ApiXU. Your first 5,000 API calls every month are free, but our Worldwide Weather application for FusionPBX actually makes two API calls to retrieve the latest weather conditions AND the weather forecast. What that means is you can make 2,500 free queries a month with the Nerd Vittles application. One or two a day should suffice. While the U.S. weather reports are retrieved by ZIP code, the ApiXU queries are retrieved by city. So long as you don’t choose small towns, the city names should be sufficiently unique to work well with the WorldWide Weather application. HINT: Nicosia in Cyprus (home of 3CX) works great! 😉

Before you can obtain worldwide weather reports, you’ll need to sign up for an account at ApiXU.com. Once you’re registered, log into your account and copy down your API Key. You’ll need it in a minute.

5 Steps to TTS Paradise with FusionPBX

Once you have your IBM TTS credentials in hand, there are only five simple steps to get everything set up for TTS application support on FusionPBX. When we’re finished, anyone on your PBX can pick up a phone and listen to the News Headlines by dialing 951, a U.S. Weather Forecast by dialing 947, or Worldwide Weather for most international cities by dialing 949.

Download WAV file placeholders to FusionPBX
Set up TTS Extensions in FusionPBX
Install the Linux components to support TTS Applications
Insert IBM and ApiXU Credentials, Email Address and Weather Locations
Run the News Headlines and Weather Update Scripts

1. Downloading WAV File Placeholders

Login to your FusionPBX server as root using SSH or Putty. Change to /var/lib/freeswitch/recordings directory. List its contents to decipher the names of any subdirectories that have been created for your various FusionPBX domains. Change to each subdirectory under /var/lib/freeswitch/recordings and issue the following commands to install the TTS placeholders:

wget http://incrediblepbx.com/freeswitch/placeholders.tar.gz
tar zxvf placeholders.tar.gz
rm -f placeholders.tar.gz

IMPORTANT: Once you’ve copied the placeholders into position, use a browser to open the FusionPBX Dashboard for each of your domains. Navigate to Apps then Recordings and play each of the three placeholder files that were uploaded: News-update, Weather-forecast, and Weather-zip. Otherwise, they won’t be available for use in the next step of the setup.

2. Setting Up TTS Apps in FusionPBX

Before you can implement the Nerd Vittles TTS Apps for News Headlines, Weather by ZIP Code, and Worldwide Weather, we first need to create the proper environment on the FusionPBX side to support the new applications. We’ll be using the FusionPBX Dialplan Manager for this purpose. We need to set up three extensions to handle the calls: one for the News Headlines and one for each of the Weather applications.

Login to your FusionPBX Dashboard with a browser.

News Headlines: From the FusionPBX Dashboard, navigate to DialPlan, then Dialplan Manager, and click the Add (+) icon. Using your default Context, insert the following new entry into the Dialplan for News Headlines (951) by filling in the Name, Condition1, Action1, and Description fields as shown below. Leave the other defaults. Then click SAVE.

When the Dialplan listing reappears, find the NewsHeadlines entry in the list and click the pencil icon to Edit the entry. Add 951 in the Number field as shown below. Then click SAVE and BACK.

Now let’s add the Dialplan entries to support the two Weather applications.

Weather by ZIP Code: From the FusionPBX Dashboard, navigate to DialPlan, then Dialplan Manager, and click the Add (+) icon. Using your default Context, insert the following new entry into the Dialplan for Weather by ZIP Code (947) by filling in the Name, Condition1, Action1, and Description fields as shown below. Leave the other defaults. Then click SAVE.

When the Dialplan listing reappears, find the WeatherZIP entry in the list and click the pencil icon to Edit the entry. Add 947 in the Number field as shown below. Then click SAVE and BACK.

Worldwide Weather: From the FusionPBX Dashboard, navigate to DialPlan, then Dialplan Manager, and click the Add (+) icon. Using your default Context, insert the following new entry into the Dialplan for Worldwide Weather (949) by filling in the Name, Condition1, Action1, and Description fields as shown below. Leave the other defaults. Then click SAVE.

When the Dialplan listing reappears, find the WorldwideWeather entry in the list and click the pencil icon to Edit the entry. Add 949 in the Number field as shown below. Then click SAVE and BACK.

Try things out by dialing 947, 949, and 951 from any FusionPBX extension. Be sure these work before proceeding!

3. Installing Linux Components for TTS

First, we need to get the missing pieces in place to support TTS applications using IBM Bluemix TTS and the Nerd Vittles scripts. We want to add PHP support from the Linux CLI only so there will be no security issues. And we want to add support for SQLite 3 so we can look up latitude and longitude data for U.S. zip codes. Just issue the following commands to get everything set up:

apt-get update
apt-get -y install php-fpm php-curl php-cli php-pear php-db php-gd sqlite3 libsqlite3-dev
apt-get -y install sox lame libsox-fmt-mp3
sed -i 's|;cgi.fix_pathinfo=1|cgi.fix_pathinfo=0|' /etc/php/7.1/fpm/php.ini
systemctl restart php7.1-fpm

Next, we need to put the Nerd Vittles scripts and ZIP code database for SQLite 3 in place:

cd /
wget http://incrediblepbx.com/freeswitch/fusionpbx-tts-linux.tar.gz
tar zxvf fusionpbx-tts-linux.tar.gz
rm -f fusionpbx-tts-linux.tar.gz

Finally, we need to add cron jobs to run the three update scripts at least once a day. You can run them more often depending upon your needs. We have these configured to run at 6:15 am and 6:20 am every day. Adjust to meet your own requirements. On a busy PBX, you probably don’t want to run them during the workday.

echo "15 6 * * * root /root/nv-weather-update.sh >/dev/null 2>&1" >> /etc/crontab
echo "20 6 * * * root /root/nv-news-update.sh >/dev/null 2>&1" >> /etc/crontab
echo "25 6 * * * root /root/nv-wwweather-update.sh >/dev/null 2>&1" >> /etc/crontab

4. Adding TTS Credentials to FusionPBX

Now we need to add your IBM TTS and ApiXU credentials, email address, a local ZIP code for Weather by ZIP code reports, and a city for Worldwide Weather reports. Edit the credentials file and save it with your information:

cd /root
nano -w ibm-credentials.php

5. Running the News & Weather Update Scripts

Finally, we need to run the News Headlines and two Weather update scripts once to put current data in place for FusionPBX callers. After the initial setup, the cron jobs will update the News Headlines and Weather reports every day moving forward. Press ENTER as each of the scripts finishes to get back to a command prompt.

cd /root
./nv-news-update.sh
./nv-weather-update.sh
./nv-wwweather-update.sh

Taking the News & Weather Apps for a Spin

Now you’re ready to try things out. From any phone connected to your PBX, dial 951 for current News Headlines. Then dial 947 for a local Weather Report matching your zip code. Finally, dial 949 to retrieve a worldwide weather forecast for most international cities.

If you don’t yet have a FusionPBX server set up but would like to sample the voice quality of the TTS applications running on our FusionPBX server in New York, here are several ways to try them out using an IVR we set up using an IBM voice prompt from last week’s tutorial. Airport codes reflect (PROVIDER LOCATION-SERVER LOCATION-DID LOCATION).

Skyetel DID: 843-970-9997 (SEA-BUF-CHS)
Vitelity DID: 646-666-5997 (DEN-BUF-NYC)
VoIPms DID: 843-606-0444 (ATL-BUF-CHS)
Free iNUM Call: 883510009901997 (ATL-BUF-ATL)
Free SIP Call: 883510009901997@sip.inum.net (ATL-BUF-ATL)

Originally published: Monday, September 24, 2018

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

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.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Some Recent Nerd Vittles Articles of Interest…

The included applications are licensed pursuant to GPL2 with the exception of nv-worldwide-weather.php which is licensed pursuant to The MIT License. Terms and conditions of both licenses are included in /root/COPYING. [↩]

Creating Free IBM Voice Prompts for FusionPBX/FreeSWITCH

Wednesday, September 19, 2018

SECURITY ALERT: https://securityboulevard.com/2019/06/rce-using-caller-id-multiple-vulnerabilities-in-fusionpbx/

One of the first things you’ll need if you choose to migrate to FusionPBX and FreeSWITCH is voice prompts. You can record your own in FusionPBX using the Recordings application by dialing *732. Of course, your PBX will probably sound like you recorded your own prompts. 🙂 Our first recommendation is to always direct folks to Allison Smith whose voice prompts for Asterisk are legendary. But, for those on a tight budget, recordings by a professional voice talent may not be a viable option. So, today, we want to deliver the next best thing with synthesized voice prompts that are second to none. For regular readers of Nerd Vittles, you’re already aware of our enthusiasm for IBM’s new TTS offerings. You can try them out for yourself by clicking View Demo here. Or you can sample the Weather Report for 3CX that we uploaded to SoundCloud:

[soundcloud url="https://api.soundcloud.com/tracks/364353344″ params="auto_play=false&hide_related=false&show_comments=true&show_user=true&show_reposts=false&visual=true" width="80%" height="350″ iframe="true" /]

Cost issues aside, other users may need voice prompts for VoIP applications that require a language other than English. With IBM’s TTS offerings, you have quite a selection of voices and languages from which to choose:

There are many occasions where you may want to build IVRs or applications that require custom voice prompts to obtain information from callers such as requests for a name, a phone number, a part number, a location, a department, or many other pieces of data that are used to formulate data base queries. And now there’s a simple, professional, and free way to create these voice prompts using IBM’s TTS platform and FusionPBX. The first million characters of TTS synthesis and the resultant voice files are free every month. That will be more than ample for almost all of our users. To get started, you’ll need to set up a free account with IBM Bluemix and install FusionPBX on a platform of your choice: a dedicated server, a virtual machine on your desktop PC using VirtualBox, or a cloud-based server.

NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:

Getting Started with IBM Bluemix TTS Service

You can start your free, 30-day trial of IBM Bluemix services without providing a credit card. Just sign up here. Once your account is activated, here’s how to obtain credentials for the TTS service to use with FusionPBX.

Getting Started with FusionPBX

We won’t repeat the tutorial that walks you through installation of FusionPBX. Just follow the steps outlined here. Once your server is up and running, log into your server as root using SSH or Putty. We need to add MP3 support to the SOX application before we can create voice prompts reliably with IBM’s Bluemix TTS service. Here’s how:

apt-get update
apt-get install libsox-fmt-mp3 -y

Installing the Voice Prompts Script for TTS

Now we’re ready to install the Nerd Vittles Voice Prompts script that we’ll use to actually create the custom voice prompts. While you’re still logged into your server as root with SSH or Putty, issue the following commands:

cd /root
wget http://incrediblepbx.com/freeswitch/ibmprompt-fusion.tar.gz
tar zxvf ibmprompt-fusion.tar.gz
rm -f ibmprompt-fusion.tar.gz

Adding Your Credentials to the Script

Using your favorite editor, it’s time to add your IBM TTS credentials to the Voice Prompt script: nano -w ibmprompt.php. Simply replace the x’s in $API_KEY with your credentials from above. If you prefer a different voice for your voice prompts, update the $IBM_voice option using the examples shown below. For example, for the Brazilian Portuguese voice, use $IBM_voice = "pt-BR_IsabelaVoice". Verify that the $IBM_url matches what was provided with your credentials. Once you’ve updated the entries, save the file: Ctrl-X, Y, and ENTER.

Taking Voice Prompts Script for a Test Drive

Now we’re ready to try things out. The syntax while logged into the /root folder looks like this. If creating a prompt in a different language, text should be in native language, not English.

./ibmprompt.php "Text of your voice prompt"

Once the voice prompt is generated, you’ll find voiceprompt.wav in the /root folder. We’ve added a second script to actually move the new voice prompts into place to use with FusionPBX. So, once you’ve created the voice prompt above, issue the following command to assign a prompt name and copy the prompt into your FusionPBX recordings folders. Don’t forget the .wav extension on the prompt name you choose, and don’t put any spaces in the prompt name. Use hyphens.

./ibmprompt-to-fusionpbx prompt-name.wav

Once you have put the recordings in place in FusionPBX, you can edit and play them back within the FusionPBX GUI by navigating to Apps -> Recordings. Enjoy!

Originally published: Wednesday, September 19, 2018

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

Need help with Asterisk or 3CX? 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.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Some Recent Nerd Vittles Articles of Interest…

One Minute Cloud VPS: Meet Incredible PBX for HiFormance

Monday, June 18, 2018 / 5 Comments on One Minute Cloud VPS: Meet Incredible PBX for HiFormance

Dec. 17 Update: Before signing up for HiFormance service, read the latest update on this provider here.

Part of the challenge of deploying an Asterisk®-based server on a cloud platform is getting all of 1,000+ pieces in place without a hiccup during the installation. Particularly for first-time users of a VoIP platform, this can be problematic. HiFormance is one of our favorite low-cost cloud providers, and today they’re introducing something that no other cloud provider offers: a configured Incredible PBX 13-13 VPS that’s ready to accept calls in under a minute. We’ve been inching toward a new low-cost plateau for VoIP cloud providers, and today we have a new milestone that finally makes running VoIP servers out of your home or office look like the horse-and-buggy days.

$10 to $13 a year now buys you a cloud platform that is less expensive than the cost of electricity to run a server on premise. You get 2GB of RAM, 20GB of SSD storage, two virtual core processors, and 2TB of monthly bandwidth. If you prepay for 3 years, you can double either the RAM or SSD storage by simply opening a ticket after you sign up. The $10/year offering is in limited supply and gives you your choice of the following sites: L.A., Buffalo, or Dallas. The $13/year offering adds the following additional sites: Chicago, Atlanta, and New York. It’s a near perfect platform for Incredible PBX 13-13 with CentOS 6.9. Add a Google Voice trunk and you get unlimited calling in the U.S. and Canada combined with a feature set that you’ll be hard-pressed to find on any PBX at any price. Putting all the pieces in place is about as simple as preparing slice-and-bake cookies, and you’ll be up and running before the cookies come out of the oven. Skip that hamburger lunch and come join the party!

So what’s the catch? Well, there’s no catch with Incredible PBX 13-13 and CentOS 6.9. This HiFormance platform uses OpenVZ with SolusVM. Someone always asks, "If the platform is so great, why aren’t you using it?" And our answer is we are. We have deployed HiFormance cloud-based VoIP servers running Incredible PBX 13-13 in Atlanta, Buffalo, Chicago, and Los Angeles without any hiccups in service. Performance is excellent. Support is excellent. So run, don’t walk, to sign up for one of these before they’re all gone. You won’t be disappointed. Just fill out the entries as shown above once you log into the HiFormance site. Nerd Vittles receives no commissions from signups.

June 20 Update: For those of you on the West Coast and readers in the Far East, we are pleased to announce that HiFormance has extended the Incredible PBX offer to their new $1/month VZ-2019 platform featuring 4GB RAM, 20GB SSD storage, 4TB of bandwidth, and DDOS protection at the QuadraNet Los Angeles data center. As with the other offers, you can double the bandwidth or storage by prepaying for three years in advance and opening a ticket. For those in the Far East, you can upgrade to the Asia Optimized CN2, Los Angeles offering for an additional $10.99 a year.

Getting Started with Incredible PBX 13-13

Once you’ve signed up for an account, choose a location for your server and select Incredible PBX 13-13 as your "operating system." It’s a preconfigured implementation that uses the latest CentOS 6.9 platform. Once you receive your credentials by email, login to your SolusVM Control Panel with a browser. Click the Manage tab for your virtual machine. In the Settings tab at the bottom of the control panel, turn TUN/TAP ON if you plan to use the NeoRouter VPN. Next, click the Root/Admin Password tab and set a default root password for your virtual machine. Take a look around in the control panel while you’re there. This is where you’d return if you need to restart your virtual machine or if you wish to reinstall your virtual machine with the same or a different operating system.

Now that your virtual machine is up and running, log into your server with SSH as root and type YES to get started. Be sure to log in from a desktop machine that you plan to use to manage your server since this IP address will be whitelisted. You’ll be prompted to immediately change all of your passwords. When the setup finishes, just type "reboot" to reboot your server. Log back in as root and the Automatic Update Utility will bring your server up to current specs. If you’re not in the Eastern U.S. time zone, type: /root/timezone-setup. Then you’re ready to go.

One of the unique features of Incredible PBX 13-13 is that most of the major components of the aggregation including Asterisk were compiled from source, and the source code is actually on your server. The advantage is you can quickly make any future modifications desired to meet your own unique requirements. You won’t find this in any other VoIP implementation. For example, here’s all it takes to upgrade to the very latest release of Asterisk. As with the current version, both support SRTP encrypted communications!

cd /usr/src
rm -rf asterisk-13.21.0
wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-13.21.1.tar.gz
tar zxvf asterisk-13.21.1.tar.gz
mv asterisk-13.21.1.tar.gz /tmp
cd asterisk-13.21.1
contrib/scripts/get_mp3_source.sh
 mv res/res_xmpp.c res/res_xmpp.c.orig
 wget http://incrediblepbx.com/res_xmpp-13.tar.gz
 tar zxvf res_xmpp-13.tar.gz
 cp -p res/res_xmpp.c res/res_xmpp.c.oauth
make distclean
wget http://incrediblepbx.com/menuselect-incredible13.8.tar.gz
tar zxvf menuselect-incredible*
rm -f menuselect-incredible*
wget https://issues.asterisk.org/jira/secure/attachment/54233/srtp_aead_and_big_aes.patch
wget http://incrediblepbx.com/srtp_aead_and_big_aes.patch.patch
patch -p0 < ./srtp_aead_and_big_aes.patch.patch
patch -p0 <./srtp_aead_and_big_aes.patch
CFLAGS='-DENABLE_SRTP_AES_256 -DENABLE_SRTP_AES_GCM' ./configure --libdir=/usr/lib64 --with-pjproject-bundled
 make menuselect.makeopts
 menuselect/menuselect --enable-category MENUSELECT_ADDONS menuselect.makeopts
 menuselect/menuselect --enable-category MENUSELECT_CODECS menuselect.makeopts
 menuselect/menuselect --disable-category MENUSELECT_TESTS menuselect.makeopts
 menuselect/menuselect --enable CORE-SOUNDS-EN-GSM --enable MOH-OPSOUND-WAV --enable EXTRA-SOUNDS-EN-GSM --enable cdr_mysql menuselect.makeopts
 menuselect/menuselect --disable app_mysql --disable app_setcallerid --disable func_audiohookinherit --disable res_fax_spandsp menuselect.makeopts
 menuselect/menuselect --disable test_named_lock --disable test_res_pjsip_scheduler --disable test_file --disable test_bridging --disable test_res_pjsip_scheduler menuselect.makeopts
 sed -i 's|march=native|mtune=native|' Makefile.rules
expect -c 'set timeout 60;spawn make menuselect;expect "Save";send "\t\t\r";interact'
make
make install
ldconfig
cd /usr/src/Asterisk-Flite-2.2-rc1-flite1.3
make clean
make
make install
amportal restart

WebMin is also installed and configured as part of the base install. The root password for access is the same as your Linux root password. We strongly recommend that you not use WebMin to make configuration changes to your server. You may inadvertently damage the operation of your PBX beyond repair. WebMin is an excellent tool to LOOK at how your server is configured. When used for that purpose, we highly recommend WebMin as a way to become familiar with your Linux configuration.

Using the Incredible PBX 13-13 Web GUI

Most of the configuration of your PBX will be performed using the web-based Incredible PBX GUI with its FreePBX 13 GPL modules. Use a browser pointed to the IP address of your server and choose Incredible PBX Admin. Log in as admin with the password you configured in the previous step. HINT: You can always change it if you happen to forget it: /root/admin-pw-change. You can safely ignore the warning about a missing swap file. You have plenty of RAM, and OpenVZ platforms don't permit swap files. If you're worried about it, choose the 3-year prepayment option and double your ram from 2GB to 4GB which is more than ample for even the busiest PBXs.

To get a basic system set up so that you can make and receive calls, you'll need to configure a VoIP trunk, create one or more extensions, set up an inbound route to send incoming calls to an extension, and set up an outbound route to send calls placed from your extension to a VoIP trunk that connects to telephones in the real world. You'll also need a SIP phone or softphone to use as an extension on your PBX.

Continue Reading: Configuring Extensions, Trunks & Routes

Introducing Incredible PBX Whole Enchilada

The Whole Enchilada upgrade already is included on the HiFormance platform. You can review the tutorial for the three dozen apps here.

Reconfiguring NeoRouter VPN for OpenVZ

On OpenVZ platforms including HiFormance, you'll need to enable TUN/TAP in the Control Panel for your VPS. After adjusting the setting, reboot your server. Then the NeoRouter VPN client will function properly.

Installing Incredible Fax with HylaFax/AvantFax

To install Incredible Fax, log into your server as root and issue the following commands. Install time: 2 minutes.

cd /root
./incrediblefax13.sh

After entering your email address to receive incoming faxes, you'll be prompted about two dozen times to choose options as part of the install. Simple press the ENTER key at each prompt and accept all of the defaults. When the install finishes, make certain that you reboot your server to bring Incredible Fax on line. There will be a new AvantFax option in the Incredible PBX GUI. The default credentials for AvantFax GUI are admin:password; however, you first may be prompted for your Apache admin credentials which were set when you installed your server. Then you'll be asked to change your AvantFax password.

Be advised that HiFormance blocks outbound email ports including port 587 by default. You'll need to open a ticket with them to use email from your server. You can test whether port 25 and 587 are blocked with the following command:

telnet gmail-smtp-msa.l.google.com 587

Backing Up Your Server

One of the reasons we love the CentOS 6.9 platform is the ease with which you can make full system backups, particularly on cloud platforms. There's no MySQL InnoDB databases to worry about so you can treat MySQL databases just like any other Linux file. We're offering the following script for your use pursuant to the GPL2 license. By using the script at no cost, you agree to assume all risks and absolve us from any liability regarding bugs, performance, or any other failure in the code. If that's acceptable to you, copy the commands below and create a backup-full script in the /root folder of your primary server. After saving the script, make it executable: chmod +x backup-full.

#!/bin/bash
# backup-full for Incredible PBX, Copyright (c) 2008-2018, Ward Mundy & Associates, LLC
# Licensed pursuant to GPL2. See /root/COPYING on any Incredible PBX server for details
amportal stop
service mysqld stop
service httpd stop
service sendmail stop
cd /
tar -cf /tmp/backup.tar /bin /etc /home /lib /lib64 /media /mnt /opt /root /sbin /usr /var
service sendmail start
service httpd start
service mysqld start
amportal start
tar --delete -f /tmp/backup.tar etc/udev/rules.d
tar --delete -f /tmp/backup.tar etc/sysconfig/network-scripts
tar --delete -f /tmp/backup.tar var/lib/dhclient
tar --delete -f /tmp/backup.tar etc/fstab
tar --delete -f /tmp/backup.tar etc/resolv.conf
tar --delete -f /tmp/backup.tar etc/hosts
tar --delete -f /tmp/backup.tar etc/hostname
gzip /tmp/backup.tar
echo "Your backup is available: /tmp/backup.tar.gz"
echo "Copy it and test it in a safe place OFF SITE"
echo " "

To run the script, execute the following command: /root/backup-full

We recommend running the backup-full script during hours when your PBX is not in active use since Asterisk and other services typically are shut down for 2-10 minutes. Depending upon the size and performance of your server, the complete backup process typically takes 5-20 minutes. Once the backup script finishes, copy /tmp/backup.tar.gz to a safe place away from the primary server every week. You can automate the backup and the copying procedure with a cron job if desired.

To restore, copy your backup.tar.gz file back to the /tmp directory on the new server and then:

#!/bin/bash
# restore-full for Incredible PBX, Copyright (c) 2008-2018, Ward Mundy & Associates, LLC
# Licensed pursuant to GPL2. See /root/COPYING on any Incredible PBX server for details
amportal stop
service mysqld stop
service httpd stop
service sendmail stop
cd /
tar zxvf /tmp/backup.tar.gz
reboot

Upgrading to IBM Speech Engines

NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:

If you've endured Google's Death by a Thousand Cuts with text-to-speech (TTS) and voice recognition (STT) over the years, then we don't have to tell you what a welcome addition IBM's new speech utilities are. We can't say enough good things about the new IBM Watson TTS and STT offerings. While IBM's services are not free, that’s really theoretical for most of our readers. Your first month on the platform is entirely free. And, after that, you get 1,000 minutes a month of free STT voice recognition services. And the first million characters of text-to-speech synthesis are FREE every month as well. So let's put the pieces in place so you'll be ready to play with the Whole Enchilada. Here's our tutorial that will walk you through the one-time IBM setup.

Next, login to your Incredible PBX server and issue these commands to update your Asterisk dialplan and edit ibmtts.php:

cd /var/lib/asterisk/agi-bin
./install-ibmtts-dialplan.sh
nano -w ibmtts.php

Insert your credentials in $IBM_username and $IBM_password. Verify that $IBM_url matches the entry provided when you registered with IBM. Then save the file: Ctrl-X, Y, then ENTER. Now reload the Asterisk dialplan: asterisk -rx "dialplan reload". Try things out by dialing 951 (news) or 947 (Weather) from an extension registered on your PBX.

To get IBM's Speech to Text service configured, while still logged in to your Incredible PBX server, issue these commands to edit getnumber.sh:

cd /var/lib/asterisk/agi-bin
nano -w getnumber.sh

Insert your API_USERNAME and API_PASSWORD in the fields provided. Then save the file: Ctrl-X, Y, then ENTER. Update your Voice Dialer (411) to use the new IBM STT service:

sed -i '\\:// BEGIN Call by Name:,\\:// END Call by Name:d' /etc/asterisk/extensions_custom.conf
sed -i '/\\[from-internal-custom\]/r ibm-411.txt' /etc/asterisk/extensions_custom.conf
asterisk -rx "dialplan reload"

Now try out the Incredible PBX Voice Dialer with AsteriDex by dialing 411 and saying "Delta Airlines."

Configuring Google Voice with Incredible PBX

The advantage of Google Voice trunks for those of you in the United States is that all of your calls within the U.S. and Canada are free. You can't beat the price, and it has worked reliably for many, many years but it may disappear any day now. If so, we'll have a fresh tutorial with a workaround for you soon. Unless Google pulls the plug on XMPP support for Google Voice, there are three different ways to set up Google Voice trunks with Incredible PBX. For a one-time fee of $4.99 with this coupon, you can use the Simonics GV/SIP gateway to configure a Google Voice account using OAuth 2 authentication. Then just set up the Simonics SIP trunk on your PBX to point to the Simonics gateway. A second option is to choose the (recommended) OAuth 2 authentication method for Google Voice when you initially install Incredible PBX 13-13. Finally, you can choose plain-text passwords for Google Voice when you set up Incredible PBX. The drawback of this last option is Google has hinted that they may discontinue support of plain-text passwords.

Here are the initial setup steps on the Google side:

1. Set up a dedicated Gmail and Google Voice account to use exclusively for this Google Voice setup on your PBX. Head over to the Google Voice site and register. You'll need to provide a U.S. phone number to verify your account by either text message or phone call.

2. Once you have verified your account by entering your verification code, you'll get a welcome message from Mr. Google. Click Continue to Google Voice.

3. Provide an existing U.S. phone number for verification. It can be the same one you used to set up your Google account in step #1.

4. Once your phone number has been verified, choose a DID in the area code of your choice.

Special Note: Google continues to tighten up on obtaining more than one Google Voice number from the same computer or the same IP address. If this is a problem for you, here's a workaround. From your smartphone, install the Google Voice app from iPhone App Store or Google's Play Store. Then open the app and login to your new Google account. Choose your new Google Voice number when prompted and provide a cell number with SMS as your callback number for verification. Once the number is verified, log out of Google Voice. Do NOT make any calls. Now head back to your PC's browser and login to https://voice.google.com. You will be presented with the new Google Voice interface which does not include the Google Chat option. But fear not. At least for now there's still a way to get there. After you have set up your new phone number and opened the Google Voice interface, click on the 3 vertical dots in the left sidebar (it's labeled More). When it opens, click Legacy Google Voice in the sidebar. That will return you to the old UI. Now click on the Gear icon (upper right) and choose Settings. Make sure the Google Chat option is selected and disable forwarding calls to whatever default phone number you set up.

5. When your DID has been assigned, click the More icon at the bottom of the left column of the Google Voice desktop. Click Legacy Google Voice. Now click the Settings icon on your legacy Google Voice desktop. Make certain that Forward Calls to Google chat is checked and disable calls to your forwarding number. Click on the Calls tab and select Call Screening:OFF, CallerID (Incoming):Display Caller's Number, and Global Spam Filtering:checked. The remaining entries should be blank.

6. Google Voice configuration is now complete. Sign out of your Google Voice account.

The Simonics GV-SIP Gateway Solution. Here's the quick thumbnail of the steps to put all the pieces in place. First, we set up a Google Voice account at Google as documented above. Next, we'll set up an account at the Simonics site to link our Google Voice account to the Simonics SIP Gateway. Then we'll plug our Simonics SIP credentials into the preconfigured Simonics trunk on Incredible PBX. Finally, we'll add Incoming and Outgoing Routes to tell Incredible PBX how to process Google Voice calls.

Now you're ready to set up an account on the Simonics site. With this Nerd Vittles link, there's a one-time fee of $4.99.

1. Start by registering your new Google account.

2. After paying the $4.99 registration fee via PayPal, proceed through the setup process to link your Google Voice account and 11-digit Google Voice phone number to the Simonics SIP Gateway.

3. You then will be provided your SIP username and password as well as the gateway address, gvgw.simonics.com, to use in building your SIP trunk on your PBX.

4. If your SIP credentials ever get compromised, regenerate your password by logging back into the Simonics GW site.

Now it's time to configure your Simonics trunk in Incredible PBX. Start by logging into the web interface as admin with your admin password from above. Click Connectivity:Trunks and choose the Simonics trunk in the PBX Configuration menu. The Simonics trunk template will display:

1. Untick the Disable Trunk check box.

2. In Outbound CallerID, insert your 10-digit Google Voice number.

3. In username, insert GV1 followed by your 10-digit Google Voice number.

4. In secret, insert your Simonics SIP password.

5. In the Registration String, insert GV1 followed by your 10-digit Google Voice number followed by a colon (:)

6. In the Registration String after the colon, insert your Simonics SIP password.

7. In the tail of the Registration String after the slash (/), insert your 10-digit Google Voice number.

8. Click Submit Changes and then Reload the Dialplan when prompted.

Configuring GV Trunk with Motif in the GUI. If you elect to configure your Google Voice trunk natively using the Incredible PBX GUI, you first will need to obtain a Refresh_Token if you elected to use OAuth 2 authentication.

1. Be sure you are still logged into your Google Voice account. If not, log back in at https://voice.google.com.

2. In a separate browser tab, go to the Google OAUTH Playground using your browser while still logged into your Google Voice account.

3. Once logged in to Google OAUTH Playground, click on the Gear icon in upper right corner (as shown below).

3a. Check the box: Use your own OAuth credentials
3b. Enter Incredible PBX OAuth Client ID:

466295438629-prpknsovs0b8gjfcrs0sn04s9hgn8j3d.apps.googleusercontent.com

3c. Enter Incredible PBX OAuth Client secret: 4ewzJaCx275clcT4i4Hfxqo2
3d. Click Close

4. Click Step 1: Select and Authorize APIs (as shown below)

4a. In OAUTH Scope field, enter: https://www.googleapis.com/auth/googletalk
4b. Click Authorize APIs (blue) button.

5. Click Step 2: Exchange authorization code for tokens

5a. Click Exchange authorization code for tokens (blue) button

5b. When the tokens have been generated, Step 2 will close.

6. Reopen Step 2 and copy your Refresh_Token. This is the "password" you will need to enter (together with your Gmail account name and 10-digit GV phone number) when you add your GV trunk in the Incredible PBX GUI. Store this refresh_token in a safe place. Google doesn't permanently store it!

7. Authorization tokens NEVER expire! If you ever need to remove your authorization tokens, go here and delete Incredible PBX Google Voice OAUTH entry by clicking on it and choosing DELETE option.

Switch back to your Gmail account and click on the Phone icon at the bottom of the window to place one test call. Once you successfully place a call, you can log out of Google Voice and Gmail.

Yes, this is a convoluted process. Setting up a secure computing environment often is. Just follow the steps and don't skip any. It's easy once you get the hang of it. And you'll sleep better.

Now you’re ready to configure your Google Voice account in Incredible PBX. You do it from within the Incredible PBX GUI by choosing Connectivity:Google Voice. Just plug in your Google Voice Username, enter your refresh_token from Step #6 above as your Google Voice Password, enter your 10-digit Google Voice Phone Number, and check the first two boxes: Add Trunk and Add Outbound Routes. Then click Submit and Apply Settings to save your new entries.

If you elected to use plain-text passwords for Google Voice, simply skip obtaining OAuth 2 credentials and substitute your plain-text password for the refresh_token when you create the Google Voice trunk above. If you have trouble getting Google Voice to work using a plain-text password, try this Google Voice Reset Procedure. It usually fixes connectivity problems. If it still doesn’t work, enable Less Secure Apps using this Google tool.

IMPORTANT: Once you’ve entered your credentials, you MUST restart Asterisk from the Linux command line, or Google Voice calls will fail: amportal restart

Incredible PBX Wholesale Providers Access

Nerd Vittles has negotiated a special offer that gives you instant access to 300+ wholesale carriers around the globe. In lieu of paying the $650 annual fee for the service, a 13% wholesale surcharge is assessed to cover operational costs of TelecomsXchange. In addition, TelecomsXchange has generously offered to contribute a portion of the surcharge to support the Incredible PBX open source project. See this Nerd Vittles tutorial for installation instructions and signup details.

Continue Reading: Configuring Extensions, Trunks & Routes

Don't Miss: Incredible PBX Application User's Guide covering the 31 Whole Enchilada apps

Originally published: Monday, June 18, 2018

Support 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.

2018 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 11-digit DID beginning with a 1. Enjoy!

Need help with Asterisk? Join our new MeWe Support Site.

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

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

Sunday, May 27, 2018 / 4 Comments on Dare to Compare: The Best (free) VoIP Offerings for 2018

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.

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.

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.

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

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.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Some Recent Nerd Vittles Articles of Interest…

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. [↩]
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. [↩]

Cloud 9: Free Incredible PBX in the Cloud Hosting until 2019

Friday, May 25, 2018

These deals don’t come along every day so we’re interrupting our regular programming to alert you to a terrific, limited time cloud hosting offer for first-time users of Vultr. If you hurry, you can take advantage of a $25 credit on Vultr which translates into 10 free months of cloud hosting service. We can’t say enough about Vultr. They’ve been one of our key resources for development and testing of new releases of Incredible PBX for many years. Historically, they’ve supported our open source projects through generous referral revenue although that does not apply with this special offer. If you’ve always wondered whether cloud hosting was a viable alternative to on-premise solutions, now’s your chance to kick the tires at zero cost. And the other good news is you have your choice of the following Incredible PBX offerings. Simply load the required OS or upload the ISO for the platform of your choice and follow the linked tutorials below. Enjoy!

Originally published: Friday, May 25, 2018

Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! 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.

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.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Some Recent Nerd Vittles Articles of Interest…

VoiceMail Transcription for VitalPBX Using IBM Watson STT

Monday, April 23, 2018 / 2 Comments on VoiceMail Transcription for VitalPBX Using IBM Watson STT

Our VitalPBX adventure resumes today with one of the most requested PBX features regardless of platform. VoiceMail Transcription simply means that recorded voicemail messages are transcribed using a speech-to-text (STT) engine before being delivered in both written and recorded formats via email. The good news is we’ll show you how to harness IBM Watson’s STT to do the heavy lifting. Their platform is hands-down the best in the industry. And today we’ll walk you through the 5-minute setup procedure for your VitalPBX server.

IBM Watson’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. On the standard plan, voicemail transcription is 2 cents per minute, or you can opt for the LITE plan which provides 100 free minutes 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

Obtaining IBM Watson STT Credentials

NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:

If you’ve already installed the Incredible PBX add-on for VitalPBX, then IBM Watson STT already is in place. All you need is your STT (not TTS) credentials. If you haven’t installed the Incredible PBX add-on, you have two choices to get started. You can either install the Incredible PBX Custom Context now, or you can skip Incredible PBX and set up an IBM Watson account and obtain STT credentials. So start there and write down your STT credentials. You’ll need them in a minute.

Outgoing SMTP Email Setup

You obviously can’t receive voicemail messages by email if your server can’t send emails. So the next step is to configure VitalPBX to assure reliable delivery of outbound email. We strongly recommend using a Gmail account for email relay for the simple reason that many ISPs (such as Comcast) block downstream SMTP mail messages. By using Gmail as a relay host for messages sent from VitalPBX, you avoid the problem. Here’s a simple test to determine whether your server can send emails reliably. Just substitute your email address for yourname@your-email-domain.com.

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

To configure Gmail as an SMTP relay on your VitalPBX server, login to the GUI and go to Admin:System Settings:Email Settings. Click Use External Mail Server in the Server options. Choose Gmail as the Provider. Insert the From Address to match your Gmail account name. And then enter your Gmail credentials. If you use two-step authentication with your Gmail account, you’ll first need to Obtain an Application Password to use in lieu of your regular Gmail password. Once you’ve completed all of the entries, Save your settings and Reload the Dialplan when prompted. Then send yourself a test email using the fields provided. Don’t proceed until you get this working reliably.

Installing VitalPBX Voice Recognition Engine

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

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

2. Now restart Asterisk core services: asterisk -rx "core reload"

3. Edit /usr/sbin/sendmailibm and insert your IBM Watson STT credentials on lines 30 and 31. Change the language on line 34 if you don’t want en-US. Then save the file. NOTE: For new deployments, your API Username should be apikey. And your API Password will be your actual APIkey.

4. Log back into the VitalPBX GUI and configure the extensions desired for email delivery of voicemail. In PBX:Extensions:General, enter an Email Address for each extension. In PBX:Extensions:Voicemail, enter the following data using the password and timezone for each extension. Don’t enable the Delete tab until you have first tested things out.

5. If you’re using Google Voice trunks with an inbound route connecting to one or more extensions, you’ll also need to adjust the Ring Time for incoming calls, or Google Voice’s voicemail may pick up the calls before VitalPBX does. You’ll find the Ring Time setting in PBX:Extensions:Advanced for each extension. We’ve found that 20 seconds works reliably.

Originally published: Monday, April 23, 2018

Got Friends? 7 Countries Have Never Visited Nerd Vittles. 2018 Is Calling! https://t.co/wMfmlhAr16 #asterisk #freepbx #wazo #issabel #IncrediblePBX #3CX pic.twitter.com/kAmAEnwVIw

— Ward Mundy (@NerdUno) January 9, 2018

Need help with VitalPBX? Visit the VitalPBX Forum.

Sad Day. Today we say goodbye to an old friend. Feedjit has been an informative piece in the Nerd Vittles landscape for many years providing a real-time snapshot of the location of our site’s visitors and what they were reading. The following was posted on their web site today: "Due to emerging cyber risks and regulatory requirements, it is not possible to continue to operate Feedjit as a not-for-profit fun service without incurring significant costs. For this reason we are regrettably shutting down the service." We want to join the multitudes who have thanked Mark Maunder and his partner, Kerry, for their tireless efforts in providing this incredible service. We, of course, hope they will reconsider even if it means converting the site into a commercial endeavor. It was a one-of-a-kind offering that will be sorely missed in the blogosphere.

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Incredible PBX in the Cloud: A $10/Year VoIP Cloud Platform

Monday, April 16, 2018 / 3 Comments on Incredible PBX in the Cloud: A $10/Year VoIP Cloud Platform

We’ve been inching toward a new low-cost plateau for VoIP cloud providers, and today we have a new milestone that finally makes running VoIP servers out of your home or office look like the horse-and-buggy days. $10 a year now buys you a cloud platform that is less expensive than the cost of electricity to run a server on premise. You get 2GB of RAM, 20GB of SSD storage, two virtual core processors, and 2TB of monthly bandwidth. If you prepay for 3 years, you can double either the RAM or SSD storage by simply opening a ticket after you sign up. It’s a near perfect platform for Incredible PBX 13-13 with CentOS 6.9. Add a Google Voice trunk and you get unlimited calling in the U.S. and Canada combined with a feature set that you’ll be hard-pressed to find on any PBX at any price. Putting all the pieces in place is about as simple as preparing slice-and-bake cookies, and you’ll be up and running before the cookies come out of the oven. Skip that hamburger lunch and come join the VoIP revolution!

So what’s the catch? Well, there’s no catch with Incredible PBX 13-13 and CentOS 6.9. But this HiFormance platform uses OpenVZ with SolusVM, and SolusVM has some serious bugs with their CentOS 7 and Debian 9 implementations. That rules out using VitalPBX, Issabel, or Wazo. Someone always asks, "If the platform is so great, why aren’t you using it?" And our answer is we are. We have deployed HiFormance cloud-based VoIP servers running Incredible PBX 13-13 in Atlanta, Buffalo, Chicago, and Los Angeles without any hiccups in service. Performance is excellent. Support is excellent. So run, don’t walk, to sign up for one of these before they’re all gone. You won’t be disappointed. Just fill out the entries as shown above once you log into the HiFormance site. Nerd Vittles receives no commissions from signups.

Getting Started with Incredible PBX 13-13

Once your virtual machine is up and running with CentOS 6.9, log into your server as root and issue the following commands to get started. Use the first command to immediately change your root password. Then you’ll be ready to begin the Incredible PBX install. It’s a two-step process. First, the installer will bring your version of CentOS up to current specs and load the necessary packages to support Asterisk® and FreePBX®. The first stage takes 22 minutes.

passwd
cd /root
yum -y update
yum -y install net-tools nano wget tar
wget http://incrediblepbx.com/incrediblepbx-13-13-LEAN.tar.gz
tar zxvf incrediblepbx-13-13-LEAN.tar.gz
rm -f incrediblepbx-13-13-LEAN.tar.gz
./IncrediblePBX-13-13.sh

When the base install finishes, your server will reboot. Simply log back in as root and run the installer a second time. Be sure your console window is at least 80 x 28, or the install will fail. If in doubt, expand it to full screen. You’ll be prompted whether to implement Google Voice plain text or OAuth 2 passwords.¹ OAuth is strongly recommended. In fact, OAuth is required if you wish to install the Whole Enchilada upgrade which gets you several dozen preconfigured applications for Asterisk. Make your selection, and the installer will work its magic. Return in 12 minutes.

./IncrediblePBX-13-13.sh

Reboot one final time when the installer finishes the setup, and Incredible PBX LEAN will be ready to go. Log back in as root. This will kick off the Automatic Update Utility to load any last minute additions, bug fixes, and security patches. After the status menu displays, run the following apps to set a very secure admin password for web access to the GUI and to choose your default time zone:

/root/admin-pw-change
/root/timezone-setup

One of the unique features of Incredible PBX 13-13 is that most of the major components of the aggregation including Asterisk are compiled from source code on the fly. This has several advantages. First, you always get the latest version of the source code. And, second, the source code is available on your server so that you can make any future modifications desired to meet your own unique requirements. You won’t find this in any other VoIP implementation. It’s one of the reasons Incredible PBX takes a bit longer to install than many of the canned offerings that rely upon precompiled packages that are difficult to modify.

WebMin is also installed and configured as part of the base install. The root password for access is the same as your Linux root password. We strongly recommend that you not use WebMin to make configuration changes to your server. You may inadvertently damage the operation of your PBX beyond repair. WebMin is an excellent tool to LOOK at how your server is configured. When used for that purpose, we highly recommend WebMin as a way to become familiar with your Linux configuration.

Using the Incredible PBX 13-13 Web GUI

Most of the configuration of your PBX will be performed using the web-based Incredible PBX GUI with its FreePBX 13 GPL modules. Use a browser pointed to the IP address of your server and choose Incredible PBX Admin. Log in as admin with the password you configured in the previous step. HINT: You can always change it if you happen to forget it. You can safely ignore the warning about a missing swap file. You have plenty of RAM, and OpenVZ platforms don’t permit swap files. If you’re worried about it, choose the 3-year prepayment option and double your ram from 2GB to 4GB which is more than ample for even the busiest PBXs.

NOTE: If you plan to upgrade to the Whole Enchilada, you can skip the rest of this section. It’s for those that wish to roll their own PBX from the ground up.

To get a basic system set up so that you can make and receive calls, you’ll need to add a VoIP trunk, create one or more extensions, set up an inbound route to send incoming calls to an extension, and set up an outbound route to send calls placed from your extension to a VoIP trunk that connects to telephones in the real world. You’ll also need a SIP phone or softphone to use as an extension on your PBX. Our previous tutorial will walk you through this setup procedure. Over the years, we’ve built a number of command line utilities including a script to preconfigure SIP trunks for more than a dozen providers in seconds. You’ll find links to all of them here.

Continue Reading: Configuring Extensions, Trunks & Routes

Reconfiguring PortKnocker for OpenVZ

By default, PortKnocker monitors activity on eth0. Most OpenVZ platforms including HiFormance use venet0:0 as the default Ethernet port. Issue the following commands to get PortKnocker up and running. Then pbxstatus should show PortKnocker working.

echo 'OPTIONS="-i venet0:0"' >> /etc/sysconfig/knockd
service knockd restart
pbxstatus

Reconfiguring NeoRouter VPN for OpenVZ

On OpenVZ platforms including HiFormance, you’ll need to enable TUN/TAP in the Control Panel for your VPS. After adjusting the setting, reboot your server. Then the NeoRouter VPN client will function properly.

Upgrading to Incredible PBX Whole Enchilada

There now are two more pieces to put in place. The sequence matters! Be sure to upgrade to the Whole Enchilada before you install Incredible Fax. If you perform the steps backwards, you may irreparably damage your fax setup by overwriting parts of it.

The Whole Enchilada upgrade script now is included in the Incredible PBX LEAN tarball. Upgrading to the Whole Enchilada is simple. Log into your server as root and issue the following commands. Be advised that this upgrade will overwrite all of your existing Incredible PBX setup including any extensions, trunks, and routes you may have created previously. You also will be prompted to reset all of your passwords as part of the upgrade. Install time: 2 minutes.

cd /root
./Enchilada*

If you accidentally installed Incredible Fax before upgrading to the Whole Enchilada, you may be able to recover your Incredible Fax setup by executing the following commands. It’s worth a try anyway.

amportal a ma install avantfax
amportal a r

Installing Incredible Fax with HylaFax/AvantFax

You don’t need to upgrade to the Whole Enchilada in order to use Incredible Fax; however, you may forfeit the opportunity to later upgrade to the Whole Enchilada if you install Incredible Fax first. But the choice is completely up to you. To install Incredible Fax, log into your server as root and issue the following commands. Install time: 2 minutes.

cd /root
./incrediblefax13.sh

After entering your email address to receive incoming faxes, you’ll be prompted about two dozen times to choose options as part of the install. Simple press the ENTER key at each prompt and accept all of the defaults. When the install finishes, make certain that you reboot your server to bring Incredible Fax on line. There will be a new AvantFax option in the Incredible PBX GUI. The default credentials for AvantFax GUI are admin:password; however, you first will be prompted for your Apache admin credentials which were set when you installed Incredible PBX 13-13 LEAN or the Whole Enchilada. Then you’ll be asked to change your AvantFax password.

Upgrading to IBM Speech Engines

NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:

If you’ve endured Google’s Death by a Thousand Cuts with text-to-speech (TTS) and voice recognition (STT) over the years, then we don’t have to tell you what a welcome addition IBM’s new speech utilities are. We can’t say enough good things about the new IBM Watson TTS and STT offerings. While IBM’s services are not free, that’s really theoretical for most of our readers. Your first month on the platform is entirely free. And, after that, you get 1,000 minutes a month of free STT voice recognition services. And the first million characters of text-to-speech synthesis are FREE every month as well. So let’s put the pieces in place so you’ll be ready to play with the Whole Enchilada. Here’s our tutorial that will walk you through the one-time IBM setup.

Next, login to your Incredible PBX server and issue these commands to update your Asterisk dialplan and edit ibmtts.php:

cd /var/lib/asterisk/agi-bin
./install-ibmtts-dialplan.sh
nano -w ibmtts.php

Insert your credentials in $IBM_username and $IBM_password. Verify that $IBM_url matches the entry provided when you registered with IBM. Then save the file: Ctrl-X, Y, then ENTER. Now reload the Asterisk dialplan: asterisk -rx "dialplan reload". Try things out by dialing 951 (news) or 947 (Weather) from an extension registered on your PBX.

To get IBM’s Speech to Text service configured, while still logged in to your Incredible PBX server, issue these commands to edit getnumber.sh:

cd /var/lib/asterisk/agi-bin
nano -w getnumber.sh

Insert your API_USERNAME and API_PASSWORD in the fields provided. Then save the file: Ctrl-X, Y, then ENTER. Update your Voice Dialer (411) to use the new IBM STT service:

sed -i '\\:// BEGIN Call by Name:,\\:// END Call by Name:d' /etc/asterisk/extensions_custom.conf
sed -i '/\\[from-internal-custom\]/r ibm-411.txt' /etc/asterisk/extensions_custom.conf
asterisk -rx "dialplan reload"

Now try out the Incredible PBX Voice Dialer with AsteriDex by dialing 411 and saying "Delta Airlines." Check back next week for the Whole Enchilada apps tutorial.

Configuring Google Voice with Incredible PBX

The advantage of Google Voice trunks for those of you in the United States is that all of your calls within the U.S. and Canada are free. You can’t beat the price, and it has worked reliably for many, many years. There are three different ways to set up Google Voice trunks with Incredible PBX. For a one-time fee of $4.99 with this coupon, you can use the Simonics GV/SIP gateway to configure a Google Voice account using OAuth 2 authentication. Then just set up the Simonics SIP trunk on your PBX to point to the Simonics gateway. A second option is to choose the (recommended) OAuth 2 authentication method for Google Voice when you initially install Incredible PBX 13-13. Finally, you can choose plain-text passwords for Google Voice when you set up Incredible PBX. The drawback of this last option is Google has hinted that they may discontinue support of plain-text passwords.

Here are the initial setup steps on the Google side:

1. Set up a dedicated Gmail and Google Voice account to use exclusively for this Google Voice setup on your PBX. Head over to the Google Voice site and register. You’ll need to provide a U.S. phone number to verify your account by either text message or phone call.

2. Once you have verified your account by entering your verification code, you’ll get a welcome message from Mr. Google. Click Continue to Google Voice.

3. Provide an existing U.S. phone number for verification. It can be the same one you used to set up your Google account in step #1.

4. Once your phone number has been verified, choose a DID in the area code of your choice.

Special Note: Google continues to tighten up on obtaining more than one Google Voice number from the same computer or the same IP address. If this is a problem for you, here’s a workaround. From your smartphone, install the Google Voice app from iPhone App Store or Google’s Play Store. Then open the app and login to your new Google account. Choose your new Google Voice number when prompted and provide a cell number with SMS as your callback number for verification. Once the number is verified, log out of Google Voice. Do NOT make any calls. Now head back to your PC’s browser and login to https://voice.google.com. You will be presented with the new Google Voice interface which does not include the Google Chat option. But fear not. At least for now there’s still a way to get there. After you have set up your new phone number and opened the Google Voice interface, click on the 3 vertical dots in the left sidebar (it’s labeled More). When it opens, click Legacy Google Voice in the sidebar. That will return you to the old UI. Now click on the Gear icon (upper right) and choose Settings. Make sure the Google Chat option is selected and disable forwarding calls to whatever default phone number you set up.

5. When your DID has been assigned, click the More icon at the bottom of the left column of the Google Voice desktop. Click Legacy Google Voice. Now click the Settings icon on your legacy Google Voice desktop. Make certain that Forward Calls to Google chat is checked and disable calls to your forwarding number. Click on the Calls tab and select Call Screening:OFF, CallerID (Incoming):Display Caller’s Number, and Global Spam Filtering:checked. The remaining entries should be blank.

6. Google Voice configuration is now complete. Sign out of your Google Voice account.

The Simonics GV-SIP Gateway Solution. Here’s the quick thumbnail of the steps to put all the pieces in place. First, we set up a Google Voice account at Google as documented above. Next, we’ll set up an account at the Simonics site to link our Google Voice account to the Simonics SIP Gateway. Then we’ll plug our Simonics SIP credentials into the preconfigured Simonics trunk on Incredible PBX. Finally, we’ll add Incoming and Outgoing Routes to tell Incredible PBX how to process Google Voice calls.

Now you’re ready to set up an account on the Simonics site. With this Nerd Vittles link, there’s a one-time fee of $4.99.

1. Start by registering your new Google account.

2. After paying the $4.99 registration fee via PayPal, proceed through the setup process to link your Google Voice account and 11-digit Google Voice phone number to the Simonics SIP Gateway.

3. You then will be provided your SIP username and password as well as the gateway address, gvgw.simonics.com, to use in building your SIP trunk on your PBX.

4. If your SIP credentials ever get compromised, regenerate your password by logging back into the Simonics GW site.

Now it’s time to configure your Simonics trunk in Incredible PBX. Start by logging into the web interface as admin with your admin password from above. Click Connectivity:Trunks and choose the Simonics trunk in the PBX Configuration menu. The Simonics trunk template will display: