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


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.

  1. Download WAV file placeholders to FusionPBX
  2. Set up TTS Extensions in FusionPBX
  3. Install the Linux components to support TTS Applications
  4. Insert IBM and ApiXU Credentials, Email Address and Weather Locations
  5. 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.

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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…

  1. 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. []

Back to School: Introducing FusionPBX for FreeSWITCH

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

It’s been quite a week with the surprise acquisition of Digium® and Asterisk® by Sangoma®. It became official on Wednesday, September 5. You can read all about it here, and you can read our cautious optimism here. As with the recent Google Voice transformation, we hope it serves as a gentle reminder to the VoIP community not to put all your eggs in one basket. With the start of the new school year, we could think of no better time to explore an excellent alternative. And today we’re pleased to introduce FusionPBX™ for FreeSwitch™.

9/10 EDIT: We’ll be updating this article in coming days to add tutorials on additional features rather than releasing new articles that force you to jump around. So mark your place at the end of the article and come back soon to see the new additions.

FreeSWITCH is an open source softswitch that’s been around for over a decade. The lead designer is Anthony Minessale, who originally worked on the Asterisk project. FusionPBX is a GUI front end for FreeSWITCH that performs many of the same functions that FreePBX® performs for Asterisk. It’s the brainchild of Mark J. Crane. With that background, let’s dive right in.

Today we’ll get a functioning server set up with trunks and extensions so that you can begin making calls. We’ll also show you how to interconnect with an Incredible PBX server in the Cloud to add Google Voice GVSIP functionality for free calling in the U.S. and Canada. Once you get that far, we’d recommend you pick up a good book on FreeSWITCH, review the excellent FusionPBX documentation, and roll up your sleeves. There’s virtually nothing that FusionPBX and FreeSWITCH can’t do with a telephone.

Creating the Debian 8 Minimal Platform

We’ll be building FusionPBX atop a minimal install of Debian 8 (Jessie). If you’re creating your server in the Cloud with 1GB or less of RAM (such as the $3.50/month Vultr platform), we strongly recommend creation of a swap file after you set up the Debian 8 platform:

dd if=/dev/zero of=/swapfile bs=1024 count=1024k
chown root:root /swapfile
chmod 0600 /swapfile
mkswap /swapfile
swapon /swapfile
echo "/swapfile          swap            swap    defaults        0 0" >> /etc/fstab
sysctl vm.swappiness=10
echo vm.swappiness=10 >> /etc/sysctl.conf
free -h
cat /proc/sys/vm/swappiness

Next, create a very secure root password: passwd

Now put the missing pieces in place to support your FusionPBX install:

apt-get update
apt-get upgrade
apt-get install nano -y
apt-get install dialog -y
apt-get install ca-certificates -y
apt-get install systemd -y
apt-get install systemd-sysv -y
reboot

Installing FusionPBX and FreeSWITCH

Now we’re ready to install FusionPBX with FreeSWITCH. Issue the following command on a single line. Be advised that FusionPBX currently uses FreeSWITCH 1.6, not 1.8. If you buy a book about FreeSWITCH 1.8, just be aware that there may be some features that are not yet available with FusionPBX.

wget -O - https://raw.githubusercontent.com/fusionpbx/fusionpbx-install.sh/master/debian/pre-install.sh | sh; cd /usr/src/fusionpbx-install.sh/debian && ./install.sh

When the install completes, you’ll see a message that looks something like this:

Installation has completed.

   Use a web browser to login.
      domain name: https://45.76.249.125
      username: admin*
      password: D6pHudQGqeYsQUWK

   *The browser domain name is used as part of the authentication.

   If you need to login to a different domain then use username@domain.
      username: admin@45.76.249.125

   Official FusionPBX Training
      Fastest way to learn FusionPBX: https://www.fusionpbx.com.
      Available online and in person. Includes documentation and recording.

      Location:               Online
      Admin Training:          7 -  9 August 2018 (3 Days)
      Advanced Training:      21 - 22 August 2018 (2 Days)
      Continuing Education:        30 August 2018 (1 Day)
      Timezone:               https://www.timeanddate.com/weather/usa/boise

   Additional information.
      https://fusionpbx.com/training.php
      https://fusionpbx.com/support.php
      https://www.fusionpbx.com
      http://docs.fusionpbx.com

If you’re coming from the FreePBX world and you’re new to FusionPBX and FreeSWITCH, be advised that your browser login name is NOT admin. It’s admin@some-IP-address. The reason is because FreeSWITCH supports multiple domains, unlike FreePBX. The default domain will be the IP address from which you performed the installation. On a server in the cloud, it will be your public IP address. On a private LAN, it will be the localhost private IP address, e.g. 127.0.0.1 or 127.0.0.2.

Locking Down Your Server

FusionPBX includes a basic IPtables firewall setup. Those that have followed Nerd Vittles over the years know that we view a firewall whitelist (Travelin’ Man 3) as absolutely essential to avoid security problems down the road. In the case of FusionPBX, we recommend changing the SSH access port from 22 to a random number above 1000. Then it can remain exposed so long as you check regularly to make certain no one is attempting to access your server via SSH: cat /var/log/auth.log. We also recommend locking down HTTP and HTTPS to your whitelisted IP addresses rather than leaving those ports open to the world. Finally, we recommend closing off IPv6 access to your server except from localhost. Here’s how.

Let’s assume you want to change the SSH access port from 22 to 1789. Simply issue the following commands and restart SSH. WARNING: Be careful not to log out of your server until we update the firewall, or you will lock yourself out of your server!

sed -i 's|#Port 22|Port 22|'  /etc/ssh/sshd_config
sed -i 's|Port 22|Port 1789|' /etc/ssh/sshd_config
/etc/init.d/ssh restart

To reconfigure IPtables using a WhiteList of allowed IP addresses, you first need to decipher what those IP addresses actually are. You’ll need the public and private IP addresses of any PCs from which you wish to access FusionPBX. Depending upon your pain threshold and bank account, SIP access can remain open. However, you’ll still need the IP addresses of your hosting providers and the IP addresses of each of the locations where you plan to install a SIP phone for direct access to properly configure FusionPBX. Once you have those IP addresses in hand, it’s time to edit /etc/iptables/rules.v4. The filter section of the default install looks like:

*filter
:INPUT DROP [1:40]
:FORWARD DROP [0:0]
:OUTPUT ACCEPT [58:8069]
-A INPUT -i lo -j ACCEPT
-A INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT
-A INPUT -p udp -m udp --dport 5060:5091 -m string --string "friendly-scanner" --algo bm --to 65535 -j DROP
-A INPUT -p udp -m udp --dport 5060:5091 -m string --string "sipcli/" --algo bm --to 65535 -j DROP
-A INPUT -p udp -m udp --dport 5060:5091 -m string --string "VaxSIPUserAgent/" --algo bm --to 65535 -j DROP
-A INPUT -p tcp -m tcp --dport 22 -j ACCEPT
-A INPUT -p tcp -m tcp --dport 80 -j ACCEPT
-A INPUT -p tcp -m tcp --dport 443 -j ACCEPT
-A INPUT -p tcp -m tcp --dport 7443 -j ACCEPT
-A INPUT -p tcp -m tcp --dport 5060:5091 -j ACCEPT
-A INPUT -p udp -m udp --dport 5060:5091 -j ACCEPT
-A INPUT -p udp -m udp --dport 16384:32768 -j ACCEPT
-A INPUT -p icmp -m icmp --icmp-type 8 -j ACCEPT
-A INPUT -p udp -m udp --dport 1194 -j ACCEPT
COMMIT

1. Modify the SSH rule (–dport 22) replacing 22 with your new SSH port number, e.g. 1789.

2. Using #, comment out the HTTP (–dport 80) and HTTPS (–dport 443) rules. There simply are too many zero day vulnerabilities with PHP and SQL injection to leave web ports exposed to the public.

3. Just above the COMMIT line, whitelist your private LAN IP addresses. Do NOT whitelist the 172 subnet if you’re deploying on Amazon! Amazon treats these as routable IP addresses on their network.

-A INPUT -s 127.0.0.0/8 -j ACCEPT
-A INPUT -s 10.0.0.0/8 -j ACCEPT
-A INPUT -s 172.16.0.0/12 -j ACCEPT
-A INPUT -s 192.168.0.0/16 -j ACCEPT

4. If you’re planning to use NeoRouter VPN, add the following above the COMMIT line:

-A INPUT -p tcp -m tcp --dport 32976 -j ACCEPT

5. Add rules above the COMMIT line for each IP address you wish to WhiteList, e.g.

-A INPUT -s 8.8.8.8 -j ACCEPT

6. Save the file.

7. Edit /etc/iptables/rules.v6 to look like this:

*filter
:INPUT DROP [0:0]
:FORWARD ACCEPT [0:0]
:OUTPUT ACCEPT [0:0]
-A INPUT -s ::1 -j ACCEPT
COMMIT

8. Restart IPtables and Fail2Ban:

/etc/init.d/netfilter-persistent restart
/etc/init.d/fail2ban restart
iptables -nL
ip6tables -nL

9. If your server is on the public Internet and you’d like to add SSL security, which is required for WebRTC deployments, we’re adding a separate tutorial below as part of the WebRTC implementation to show you the easy way to do this. Keep reading.

Finally, a cautionary note. If you leave your SIP ports exposed to the Internet, then you’ll need to regularly monitor your FreeSWITCH log for attempted attacks. You can download the Incredible Utilities scripts including update-blacklist that we run regularly as a cron job to blacklist all of the most recent bad guys. Please note that IP addresses detected as "bad guys" with this script take precedence over whitelist entries you may have made in step #5 above so be sure to also add the IP addresses from step #5 to this script’s WHITELIST table before running the script, or you may inadvertently lock yourself out of your own server.

cd /
wget http://incrediblepbx.com/freeswitch/incredible-utils-FS.tar.gz
tar zxvf incredible-utils-FS.tar.gz
rm -f incredible-utils-FS.tar.gz

Getting Started with FusionPBX

Using the account credentials displayed after your installation completed, login to FusionPBX using your favorite browser. Don’t forget to include the IP address in the admin field:


Before you do anything else, navigate to Advanced -> Access Controls. Here you will want to whitelist all of the IP addresses of SIP service providers and other PBXs to which you want to interconnect. Simply add Allow entries in the Domains category for each IP address/CIDR entry. HINT: Single IP addresses have a CIDR entry of /32. WARNING: We don’t recommend using FQDN/Domain entries. Despite legitimate FQDNs, all of our entry attempts resulted in "cannot locate" alerts in the FreeSWITCH CLI (fs_cli). This means that future connection attempts from those providers would fail without any indication of what caused the failures. Also, do NOT add entries for IP addresses of phones/softphones that will register to extensions or calls to and from those extensions will fail. This is anything but intuitive but, trust us, you will save hours of hair-pulling.

Creating Extensions in FusionPBX

While you’re still logged into the FusionPBX GUI, let’s add an extension to demonstrate how easy it is. Choose Accounts -> Extensions and click on the + symbol to add a new extension. Here is a sample to get you started, but you really only need the extension number and voicemail PIN entries:



Unlike in FreePBX, the default extension password is not displayed on the template. Once you SAVE the extension, you then have to edit it and click on the Password field to display the default entry. This can be changed, if desired.

Configuring a Softphone for FusionPBX

You can connect virtually any kind of telephone to your new PBX, and FusionPBX includes terrific provisioning tools for dozens of SIP phones. We’ll start with a free SIP softphone today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum when you’re ready to get serious about VoIP telephony.

We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the extension you created above. You’ll need the IP address of your server plus your extension’s password. Fill in the Yate Client template using the IP address of your Server, the extension number for your Username, and whatever Password you assigned to the extension when you created it. Click OK to save your entries.

Once the Yate softphone shows that it has registered with FusionPBX, try a test call by dialing *9664 which should begin playing the default Music on Hold.

Creating Trunks/Providers/Gateways in FusionPBX

In FusionPBX and FreeSWITCH, what FreePBX and Asterisk users call Trunks are referred to as Providers or Gateways. These are commercial outfits that offer to interconnect your PBX with the rest of the telephones in the world using a SIP connection. The first step is to register with the providers of your choice and obtain your SIP credentials and the FQDN(s) of the provider’s servers to which you should register. Most allow authentication by either username/password or by IP address. If you have a static IP address for your server, that is the safer method since you don’t have to worry about someone guessing your password. The only difference in the setup is the Register field should be changed to False.

As luck would have it, there is thorough documentation on the FreeSWITCH site to configure literally hundreds of Providers. Here’s the link.

Before you actually set up your new Provider in FusionPBX, we first need to add the provider’s server to FusionPBX’s Access Control List (ACL). As noted, we’ve encountered problems attempting to register FQDNs in the ACL so we strongly recommend you ping the FQDN of your provider’s server and obtain its actual IP address. Once you have it, navigate to Advanced -> Access Controls -> domains. Click on the Pencil icon to edit the ACL list for domains. Next, click on the + icon at the top of the Nodes listing. Change Type to allow. Enter the IP address of your provider’s server in CIDR. Leave the Domain field blank. Enter your Provider in the Description field. Click the SAVE button.

Now we’re ready to add your Provider. Navigate to Accounts -> Gateways and click the + symbol to add a new one. Click on the Advanced button to expose all of the available fields. Now find your provider in the FreeSWITCH listing and copy the sample entries using your own credentials to the appropriate fields in the FusionPBX template. SAVE your settings when you’re finished. If you chose username/password authentication with Register=True, then your new gateway’s Status should display as Running with a State of REGED.

If you want to take advantage of free calling in the U.S. and Canada using Google Voice, then you’ll need to interconnect your FusionPBX server with an Incredible PBX GVSIP gateway as described in this Nerd Vittles article. On the FusionPBX side, the first step is to add the IP address of the Incredible PBX GVSIP gateway to the ACL (as described above). Next, assuming you followed the tutorial and created a trunk on the Incredible PBX server named FusionPBX in step #2, here’s what the corresponding Gateway should look like on the FusionPBX side:

Gateway: GVSIP-Host
Username: FusionPBX
Password: same as on GVSIP-Host
From User: FusionPBX
From Domain: FusionPBX
Realm: IP address of GVSIP-Host
Expire Seconds: 90
Register: True
Retry Seconds: 30
Auth Username (in Advanced): FusionPBX
Domain: default setting
Context: Public
Profile: external
Description: GVSIP-Host

Be advised that you still need to WhiteList the IP addresses of the two servers on the corresponding sites using IPtables. And you need to whitelist the public IP addresses even if you choose to register the trunk using private VPN addresses. The reason is because FreeSwitch uses the public IP addresses internally, and the registration will fail without the whitelist entries.

Creating Inbound Routes in FusionPBX

As with all PBXs, Inbound Routes define how incoming calls from Trunks/Gateways are routed to destinations on your PBX. Creating inbound routes in FusionPBX (Dialplan -> Inbound Routes) is much the same as the process with FreePBX except the search Conditions are considerably broader than merely a DID or CallerID match and may include Time Conditions to accommodate after-hours calling:



As with FreePBX, the Action can be any destination available on your PBX including an extension, voicemail, company directory, or an IVR:



Typically, inbound calls should be routed to the public Context. And, unlike FreePBX where the first matching inbound route wins, with FusionPBX, you can prioritize the routes numerically to assign a certain search Order.



Creating Outbound Routes in FusionPBX

Outbound Routes tell your PBX how to route calls to destinations outside your PBX using Trunks/Gateways available on your PBX. Creating outbound routes in FusionPBX (Dialplan -> Outbound Routes) is equally flexible offering virtually limitless combinations to assist PBX designers in setting up scenarios for processing outbound calls. As with inbound routes, outbound routes can be prioritized by assigning an Order. And each outbound route can include a primary Gateway as well as up to two Alternates for routing the calls.



Unlike FreePBX which used NXXNXXXXXX and similar combinations as Dialplan Expressions, FusionPBX uses more powerful RegEx coding with many predefined options:



Choosing Providers for FusionPBX

Here’s a shameless plug for our Platinum Sponsor, Vitelity, if you’re looking for an incredible deal on a DID with unlimited inbound calling. You’ll find the offer at the end of this article. If dirt-cheap outbound calls are of interest and Google Voice isn’t an option where you’re calling from or to, then you can’t beat Anveo Direct. The AnveoDirect provider setup for FusionPBX isn’t included in the link we posted above, but it couldn’t be simpler.



To make outbound calls with Anveo Direct, you dial a number with the country code preceded by a special 6-character code starting with 0 which you create on the Anveo Direct web site. You also must whitelist the IP address of your PBX as part of the setup on the Anveo side. Once configured, a call to a number in the U.S. would look like this: 04He9x18005551212@sbc.anveo.com. When creating the Outbound Route for 10-digit dialing using the tutorial above, the AnveoDirect setup would define the Dialplan Expression as 10-digit dialing with a Prefix of 04HE9x1 assuming your 6-character secret code was 04He9x. The trailing 1 in the Prefix converts the 10-digit dialed number to 11-digits as required by Anveo’s international dial code requirement. We think you’ll like their pricing:



Using Gmail as SMTP Smarthost with FusionPBX

Be sure to test sending an email to yourself from the command line to be sure Exim is working properly. Here’s how:

echo "test" | mail -s testmessage yourname@yourmailserver.com

If you don’t receive the email, be advised that many providers block downstream SMTP mail servers in which case you may want to use your Gmail account as an SMTP Smarthost with FusionPBX. Here’s how. Begin by reconfiguring Exim: dpkg-reconfigure exim4-config

  • Type Mail Server: Mail sent by smarthost using SMTP
  • System Mail Name: Your server’s FQDN (see /etc/hostname)
  • Allowed Senders: accept defaults
  • Other Destinations: accept default
  • Relay Mail: leave blank
  • Outgoing SmartHost: smtp.gmail.com::587 (note the double colons)
  • Hide local name: no
  • Keep DNS-queries minimal: no
  • Delivery method local mail: Maildir format in home directory
  • Split config into small files: no
  • Root and Postmaster recipient: root

After exim4 restarts, add the following entries to the end of /etc/exim4/passwd.client using your Gmail credentials:

gmail-smtp.l.google.com:YOUR-NAME@gmail.com:PASSWORD
*.google.com:YOUR-NAME@gmail.com:PASSWORD
smtp.gmail.com:YOUR-NAME@gmail.com:PASSWORD

Finally, issue the following commands to update exim4 and implement your changes:

update-exim4.conf
/etc/init.d/exim4 restart

Send yourself another test email to verify that everything is working properly. If the mail still doesn’t make it, be sure your provider (HiFormance, for example) is not also blocking port 587. You’ll need to open a ticket with them if this is the case. You can test whether the port is blocked with the following command:

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

Solving NAT and Audio Problems with FusionPBX

If you experience one-way audio, no audio, or calls that won’t disconnect when the called party hangs up, you’ve probably entered NAT Hell. First, make sure that SIP ALG is turned off on your router. If that doesn’t solve it, edit /etc/default/freeswitch from the Linux CLI and remove -nonat. Save the file and then systemctl daemon-reload. Switch to the FusionPBX GUI and navigate to Advanced -> SIP Profiles. Edit BOTH the internal and external profiles. Then modify BOTH the ext_rtp_ip AND ext_sip_ip entries and change them to autonat:XXX.XXX.XXX.XXX replacing XXX.XXX.XXX.XXX with your server’s public IP address. Then SAVE both profiles. Finally, return to the Linux CLI and restart FreeSWITCH: service freeswitch restart.

Congratulations! You now should have a working PBX. We’ll get deeper into the weeds down the road, but today’s tutorial coupled with the HTML FusionPBX Documentation or PDF version should be sufficient to get you started with a functioning PBX. Take some time to explore all of the Applications that are included in FusionPBX. Enjoy!



9/10 EDIT: New additions begin here…

Implementing WebRTC with FusionPBX

The first step in deploying WebRTC is to add SSL security to your server. The easiest way to do this is to take advantage of the free offering from LetsEncrypt. Begin by assigning a fully-qualified domain name (FQDN) to the public IP address of your server. Wait a few minutes for DNS propagation. Then you’re ready to install your LetsEncrypt certificate. Unlike many of the other LetsEncrypt implementations, the FusionPBX folks have made this a walk in the park. While logged into your server as root, issue the following commands:

cd /usr/src/fusionpbx-install.sh
cd debian/resources
./letsencrypt.sh
service freeswitch restart




Once the certificate is installed and you’ve restarted FreeSWITCH, close your browser and then restart it. Go to the FQDN of your server, and the lock should appear signifying that your site is now fully encrypted. Don’t proceed with the WebRTC steps until this is working.

To get a successful WebRTC implementation where you can make and receive phone calls from a browser, you’re going to need to use the Chrome or Firefox browser. We’ve also had success using the latest Safari browser.

For those that despise implementing complex procedures by viewing video tutorials, we offer the following regurgitation of the steps documented by Mark Crane in his ClueCon video below. This isn’t hard, but it is tedious so don’t skip any steps.



 

While you’re still logged into your server as root, let’s put the FusionPBX WebRTC client in place so you’ll have that option as one of several WebRTC clients to use down the road. The advantage of the FusionPBX WeRTC client is that it can handle your login automatically.

cd /usr/src
git clone https://github.com/fusionpbx/fusionpbx-apps
cd fusionpbx-apps
cp -R sipjs/ /var/www/fusionpbx/app/
chown -R www-data:www-data /var/www/fusionpbx/

Now let’s switch back to your browser and login to FusionPBX using your superadmin credentials. A word of caution… To get WebRTC working, your default Domain must be the FQDN of your server, not an IP address. Once you add this domain, you must switch to it and enter new extensions, trunks, and routes to that domain before proceeding. Begin by adding the domain: Advanced -> Domains -> Add (+). Switch to the domain in the upper right column that’s showing your current domain by clicking on it. It doubles as the Domain Selector.

First, let’s tell FreeSwitch to use your secure SSL setup. Navigate to Advanced -> Variables. Go to the SIP Profile: Internal section and change the false setting of internal_ssl_enable to true. Click SAVE. Go to the SIP Profile: External section and change the false setting of external_ssl_enable to true. Click SAVE. Navigate to Status -> SIP Status and click FLUSH CACHE. Switch back to your SSH session as root and restart FreeSWITCH: service freeswitch restart. Back in your browser, return to Status -> SIP Status, click REFRESH, and verify that both the Internal and External interfaces show TLS enabled.

Navigate to Advanced -> SIP Profiles -> Internal and set wss-binding to true. Switch back to your SSH session as root and restart FreeSWITCH: service freeswitch restart. Back in your browser, return to Status -> SIP Status, click FLUSH CACHE and then REFRESH. You now should see an internal entry for Secure Web Sockets (WSS) in your internal SIP Profile. Finally, to do video with WebRTC, navigate to Advanced -> Variables and add H264 to the list of supported codecs in both outbound_codec_prefs and global_codec_prefs: ULAW, ALAW, H264. Click SAVE. Navigate to Status -> SIP Status and click FLUSH CACHE then RESCAN the internal profile. Clicking on sofia status profile internal will let you verify that the H264 codec has been added successfully. That completes the required pieces to support WebRTC with FusionPBX.

To use the FusionPBX WebRTC client that we installed earlier, we first need to update the FusionPBX menus in the browser: Advanced -> Upgrade -> Menu Defaults and EXECUTE.

Now create an extension to use with WebRTC: Accounts -> Extensions -> Add (+). Once you’ve created the new Extension, drop down to the fourth item (Users) and click on the pull-down menu. Choose the Admin user and click the ADD button followed by SAVE. Next, log out and back into FusionPBX to associate the extension with your user account.

We’re now ready to try out the FusionPBX WebRTC client. Navigate to Apps -> SIPjs which will activate the WebRTC client with your extension credentials. In a separate window, you can verify that SIPjs is registered to your extension by navigating to Status -> Registrations. Verify that you can make a call by dialing *9664 for some nice Hold Music.

Adding Free IBM Voice Prompts to FusionPBX

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

One of the first things you’ll need with your new FusionPBX server is voice prompts for IVRs and custom applications. We’ve now added a tutorial which will walk you through setting up a platform to obtain free IBM voice prompts for your server. Here’s the link.

Blocking SIP Access by IP Address

If you’ve implemented SSL security with an FQDN as recommended above, then you’ll reduce the hammering your server takes from the bad guys by blocking those that attempt SIP registrations or calls using the IP address of your server. This, of course, means that all of your SIP registrations must be made using the FQDN of your server, not by IP address. For providers, you MUST whitelist their IP addresses in the ignoreip field of /etc/fail2ban/jail.conf and restart Fail2Ban, or they will be blocked when they attempt to send data by IP address. We’ve included a script in /root which will tell you which IP addresses currently are blocked: sip-attackers-blocked.

cd /
wget http://incrediblepbx.com/freeswitch/block-sip-by-ip.tar.gz
tar zxvf block-sip-by-ip.tar.gz
rm -f block-sip-by-ip.tar.gz
service fail2ban restart

Adding Free News/Weather TTS Apps

We’ve rolled out the first three Incredible PBX text-to-speech applications for FusionPBX: Yahoo News Headlines, Weather Reports by ZIP Code, and Worldwide Weather Forecasts. This new Nerd Vittles tutorial will walk you through the simple installation steps.

Originally published: Monday, September 3, 2018  Updated: 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 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…

It’s Soup: The Definitive Quick Start Guide to Wazo 17.17

Today we’re pleased to introduce Wazo 17.17, the latest iteration in the Wazo fork of XiVO. Twelve years in the making with the same development team, Wazo 17.17 is really something special. It’s the latest release featuring Asterisk® 15 with multi-party videoconferencing and also includes a new easy-to-use WebRTC feature and support for Asterisk FollowMe Roaming. We are pleased to announce that this release of Wazo 17.17 using the latest Incredible PBX 15 installer for Asterisk 15 now supports native Google Voice trunks with OAuth 2. Of course, all of your favorite Incredible PBX add-ons are ready as well. What follows is a new soup-to-nuts tutorial covering everything you need to know to get started with Incredible PBX for Wazo 17.17.

Wazo 17.17 enhances the brand new user interface to complement Wazo’s existing GUI. Revolutionary is probably the best way to describe its design. The reason is that it is generated using pure API calls to the Wazo engine. What that means to the administrator is you can tweak it, enhance it, replace it, or build additional functionality into your UI using the same API calls that the Wazo developers employed to build these components. This is obviously the first of many iterations that will be released in coming months, but it gives you an opportunity to kick the tires. Once you complete the installation process which follows, simply use a browser to log into the IP address of your server at the following link: https://serverIP/admin/. To begin, login with your root GUI credentials and click Plugins to display available options (pictured above). Then install each of the following plugins to begin: Users, Extensions, Contexts, and Devices. The remaining plugins are optional; however, the following plugins are not yet operational: Outcalls and Trunks. The Parking Lots and Switchboard plugins are functional but do not yet show existing setups configured through the legacy GUI. Additional tips and getting started tips are available here and here.

If you’ve been following along in our XiVO adventure with Incredible PBX last year, then you already know that there were a significant number of configuration hoops to jump through once the base install was finished. While these steps are well documented in the original Incredible PBX for XiVO tutorial, there still were plenty of opportunities for typos and skipping steps. Any misstep could spell the difference in a perfectly functioning PBX and one that couldn’t make or receive calls. So we introduced XiVO Snapshots, and today we bring the same technology to the Wazo platform as well.

A Wazo Snapshot is just what the name implies. It’s a snapshot of a working Wazo PBX that has virtually everything already configured: SIP settings that work with Asterisk®, a SIP extension that works with a SIP phone plus your cellphone, a SIP extension preconfigured for WebRTC that uses the new Opus codec, SIP and Google Voice trunk setups for many of the major commercial providers, and default inbound and outbound routes to ease the task of routing calls into and out of your PBX. Basically, you plug in your credentials from your favorite provider after running the Incredible PBX for Wazo installer, tell Wazo how to route the calls, and you’re done. You can have a stable and functional PBX making calls to anywhere in the world in a matter of minutes. Then you can review our numerous tutorials to add additional bells and whistles while actually using your Wazo PBX while you learn.

Incredible PBX for Wazo Installation Overview

Before we roll up our sleeves and walk you through the installation process, we wanted to provide a quick summary of the 10 Basic Steps in setting up Incredible PBX for Wazo. By the way, the whole process takes less than an hour!

  1. Set Up Desired PBX Platform: Stand-alone PC, Virtual Machine, or Cloud-Based Server
  2. Run the Incredible PBX for Wazo installer
  3. Set Up One or More SIP or Google Voice Trunks for Your PBX
  4. Tell Wazo Where to Direct Incoming Calls from Each Trunk
  5. Tell Wazo Which Trunk to Use for Every Outbound Calling Digit Sequence
  6. Set Up a SoftPhone or WebRTC Phone (or both)
  7. Decide Whether to Activate Simultaneous Ringing on your Cellphone
  8. Add Google Speech Recognition Key (if desired)
  9. Activate DISA with Incredible PBX for Wazo (if desired)
  10. Test Drive Incredible PBX for Wazo

1. Incredible PBX for Wazo Hardware Platform Setup

NOTE: Ignore references to Debian 9 in the platform setup tutorials. Debian 9 is not quite ready yet so stick with Debian 8.

The first step is to choose your hardware platform and decide whether you want to babysit a server and network or leave those tasks to others. We’ve taken the guesswork out of the setups documented below. Five of the options are cloud providers, each of whom provides a generous discount to let you kick the tires. So click on the links below to review the terms and our walkthrough of the setup process on each platform.

2. Running the Incredible PBX for Wazo Installer

Once you have your hardware platform up and running, the rest of the initial setup process is easy. Simply download and run the Incredible PBX for Wazo installer. On some platforms, it first updates Debian 8 to current specs and reboots. Then log back in and rerun the installer a second time if you are prompted to do so.

cd /root
wget http://incrediblepbx.com/IncrediblePBX15-Wazo.sh
chmod +x IncrediblePBX15-Wazo.sh
./IncrediblePBX15-Wazo.sh


3. Setting Up SIP and Google Voice Trunks with Wazo

When the installation is finished, you can make toll-free calls in the U.S. and Canada without doing anything except dialing "1″ and the 10-digit number from any phone connected to your server. For other calls, there are two steps in setting up trunks to use with Incredible PBX. First, you have to sign up with the provider of your choice and obtain trunk credentials. These typically include the FQDN of the provider’s server as well as your username and password to use for access to that server. Second, you have to configure a trunk on the Incredible PBX for Wazo server so that you can make or receive calls outside of your PBX. As with the platform tutorials, we have taken the guesswork out of the trunk setup procedure for roughly a dozen respected providers around the globe. In addition, Wazo Snapshots goes a step further and actually creates the trunks for you, minus your credentials, as part of the initial Incredible PBX install.

For Google Voice trunks with Incredible PBX 15, log into your server as root and run ./add-gvtrunk. When prompted, insert your 10-digit Google Voice number, your Google Voice email address and your OAuth 2 token. The native Google Voice OAuth tutorial explains how to obtain it.

For the other providers, review the setup procedure below and then edit the preconfigured trunk for that provider by logging into the Wazo web GUI and choosing IPX → Trunk Management → SIP Protocol. Edit the setup for your provider (as shown above) and fill in your credentials and CallerID number in the General tab. Activate the trunk in the Register tab after again filling in your credentials. Save your settings when finished. No additional configuration for these providers is required using the Incredible PBX for Wazo Snapshot.

4. Directing Incoming Calls from Wazo Trunks

Registered Wazo trunks typically include a DID number. With the exception of CallCentric, this is the number that callers would dial to reach your PBX. With CallCentric, it’s the 11-digit account number of your account, e.g. 17771234567. In the Wazo web GUI, we use IPX → Call Management → Incoming Calls to create inbound routes for every DID and trunk associated with your PBX. Two sample DIDs have been preconfigured to show you how to route calls to an extension or to an IVR. To use these, simply edit their settings and change the DID to match your trunk. Or you can create new incoming routes to send calls to dozens of other destinations on your PBX.

5. Routing Outgoing Calls from Wazo to Providers

Outgoing calls from extensions on your Wazo PBX must be routed to a trunk provider to reach call destinations outside your PBX. Outgoing call routing is managed in IPX → Call Management → Outgoing Calls. You tell Wazo which trunk provider to use in the General tab. Then you assign a Calling Digit Sequence to this provider in the Exten tab. For example, if NXXNXXXXXX were assigned to Vitelity, this would tell Wazo to send calls to Vitelity if the caller dialed a 10-digit number. Wazo has the flexibility to add and remove digits from a dialed number as part of the outbound call routing process. For example, you might want callers to dial 48NXXNXXXXXX to send calls to a Google Voice trunk where 48 spells "GV" on the phone keypad. We obviously don’t want to send the entire dial string to Google Voice so we tell Wazo to strip the first 2 digits (48) from the number before routing the call out your Google Voice trunk. We’ve included two examples in the Wazo Snapshot to get you started. Skype Connect (shown below) is an example showing how to strip digits and also add digits before sending a call on its way:


6. Setting Up a Softphone & WebRTC with Wazo

If you’re a Mac user, you’re lucky (and smart). Download and install Telephone from the Mac App Store. Start up the application and choose Telephone:Preference:Accounts. Click on the + icon to add a new account. To set up your softphone, you need 3 pieces of information: the IP address of your server (Domain), and your Username and Password. In the World of Wazo, you’ll find these under IPBX → Services → Lines. Just click on the Pencil icon beside the extension to which you want to connect. Now copy or cut-and-paste your Username and Password into the Accounts dialog of the Telephone app. Click Done when you’re finished, and your new softphone will come to life and should show Available. Dial the IVR (4871) to try things out. With Telephone, you can use over two dozen soft phones simultaneously on your desktop.

For everyone else, we recommend the YateClient softphone which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the Wazo Line. You’ll need the IP address of your server plus your Line username and password associated with the 701 extension. On the Wazo platform, do NOT use an actual extension number for your username with Wazo. Go to IPBX Settings → Lines to decipher the appropriate username and password for the desired extension. Click OK to save your entries.


WebRTC allows you to use your Chrome or Firefox browser as a softphone. Extension 701 comes preconfigured for WebRTC access with Incredible PBX for Wazo. It shares the same password as the Line associated with extension 701, but the username is 701 rather than the username associated with the Line. You can decipher the password by accessing the Wazo Web GUI and then IPBX → Services → Users → Incredible PBX → XiVO Client Password. Or log into your server as root using SSH or Putty and run: /root/show-701-pw. Wazo introduces several new features to WebRTC including support for the awesome new Opus codec plus voicemail management and even Gravatar support. It’s all preconfigured!

Special Note: Beginning with version 17.02 of Wazo, WebRTC is fully integrated with NGINX on your server, and a simplified method for configuring WebRTC users has been added. When you create a new User account, simply choose the SIP (WebRTC) Protocol when creating a new user account, and all of the Advanced Line options required to support WebRTC will be preconfigured for you.

To use WebRTC, you no longer need to accept the different SSL certificates associated with the WebRTC app. From your browser, go to the following site and click on each link to accept the certificates. Once you’ve completed this process, simply visit the Wazo WebRTC site.

Before logging in, click on the Gear icon in the lower right corner and then click on the Pencil icon to edit your Settings. Fill in the public IP address of your Wazo server and specify 443 for the Port. Leave the Backend field blank and click Save. Now login to your WebRTC account with Username 701. The Password is the one you obtained running show-701-pw. The IP Address (if required) is the address of your Wazo PBX.

Implementing FollowMe Roaming with a CellPhone

In addition to ringing your SIP extension when incoming calls arrive, Wazo 17.17 can also ring your cellphone simultaneously. This obviously requires at least one outbound trunk. If that trunk provider also supports CallerID spoofing, then Wazo will pass the CallerID number of the caller rather than the DID associated with the trunk. Incredible PBX for Wazo comes with cellphone support for extension 702 ready to go. To enable it, access the Wazo Web GUI and go to IPBX → Services → Users → Incredible PBX and insert your Mobile Phone Number using the same dial string format associated with the trunk you wish to use to place the calls to your cellphone. You then can answer the incoming calls on either your cellphone or the registered SIP phone. If you answer on your cellphone, you will be prompted whether you wish to accept the call. If you press 1 after observing the CallerID, the caller will be connected. If you decline, the caller will be routed to the Wazo voicemail account of the extension.

8. Activating Voice Recognition for Wazo

Google has changed the licensing of their speech recognition engine about as many times as you change diapers on a newborn baby. Today’s rule restricts use to “personal and development use.” Assuming you qualify, the very first order of business is to enable speech recognition for your Wazo PBX. Once enabled, the Incredible PBX feature set grows exponentially. You’ll ultimately have access to the Voice Dialer for AsteriDex, Worldwide Weather Reports where you can say the name of a city and state or province to get a weather forecast for almost anywhere, Wolfram Alpha for a Siri-like encyclopedia for your PBX, and Lefteris Zafiris’ speech recognition software to build additional Asterisk apps limited only by your imagination. And, rumor has it, Google is about to announce new licensing terms, but we’re not there yet. To try out the Voice Dialer in today’s demo IVR, you’ll need to obtain a license key from Google. This Nerd Vittles tutorial will walk you through that process. Add your key to /var/lib/asterisk/agi-bin/speech-recog.agi on line 72.

9. Adding DISA Support to Your Wazo PBX

If you’re new to PBX lingo, DISA stands for Direct Inward System Access. As the name implies, it lets you make calls from outside your PBX using the call resources inside your PBX. This gives anybody with your DISA credentials the ability to make calls through your PBX on your nickel. It probably ranks up there as the most abused and one of the most loved features of the modern PBX.

There are three ways to implement DISA with Incredible PBX for Wazo. You can continue reading this section for our custom implementation with two-step authentication. There also are two native Wazo methods for implementing DISA using a PIN for security. First, you can dedicate a DID to incoming DISA calls. Or you can add a DISA option to an existing IVR. Both methods are documented in our tutorial on the PIAF Forum.

We prefer two-step authentication with DISA to make it harder for the bad guys. First, the outside phone number has to match the whitelist of numbers authorized to use your DISA service. And, second, you have to supply the DISA password for your server before you get dialtone to place an outbound call. Ultimately, of course, the monkey is on your back to create a very secure DISA password and to change it regularly. If all this sounds too scary, don’t install DISA on your PBX.

1. To get started, edit /root/disa-xivo.txt. When the editor opens the dialplan code, move the cursor down to the following line:

exten => 3472,n,GotoIf($["${CALLERID(number)}"="701"]?disago1)  ; Good guy

2. Clone the line by pressing Ctrl-K and then Ctrl-U. Add copies of the line by pressing Ctrl-U again for each phone number you’d like to whitelist so that the caller can access DISA on your server. Now edit each line and replace 701 with the 10-digit number to be whitelisted.

3. Move the cursor down to the following line and replace 12341234 with the 8-digit numeric password that callers will have to enter to access DISA on your server:

exten => 3472,n,GotoIf($["${MYCODE}" = "12341234"]?disago2:bad,1)

4. Save the dialplan changes by pressing Ctrl-X, then Y, then ENTER.

5. Now copy the dialplan code into your Wazo setup, remove any previous copies of the code, and restart Asterisk:

cd /root
sed -i '\:// BEGIN DISA:,\:// END DISA\:d' /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf
cat disa-xivo.txt >> /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf
/etc/init.d/asterisk reload

6. The traditional way to access DISA is to add it as an undisclosed option in an IVR that is assigned to one of your inbound trunks (DIDs). For the demo IVR that is installed, edit the ivr-1.conf configuration file and change the "option 0″ line so that it looks like this. Then SAVE your changes.

exten => 0,1(ivrsel-0),Dial(Local/3472@default)

7. Adjust the inbound calls route of one of your DIDs to point to the demo IVR by changing the destination to Customized with the following Command:

Goto(ivr-1,s,1)

A sample is included in the Wazo Snapshot. Here’s how ours looks for the Demo IVR:



8. Now you should be able to call your DID and choose option 0 to access DISA assuming you have whitelisted the number from which you are calling. When prompted, enter the DISA password you assigned and press #. You then should be able to dial a 10-digit number to make an outside call from within your PBX.

SECURITY HINT: Whenever you implement a new IVR on your PBX, it’s always a good idea to call in from an outside number 13 TIMES and try every key from your phone to make sure there is no unanticipated hole in your setup. Be sure to also let the IVR timeout to see what result you get.

10. NEW: Travelin’ Man 3 WhiteList Firewall

All new installs of Incredible PBX for Wazo include the newly enhanced and preconfigured IPtables Firewall featuring Travelin’ Man 3. Please read the new tutorial when you have a few minutes. It documents how to enable access to your server for remote phones and traveling users.

PortKnocker previously let a remote user ping the server with a 3-digit code to temporarily add the remote user’s IP address to the IPtables Firewall WhiteList so that the user could gain remote access. We now are adding the ability to let end-users make successful PortKnocks permanent just as if an administrator had used add-ip on the server to add an IP address. Administrators still can remove these entries using del-acct although you will have to know the date and time that the entry was added. The file names consist of timestamp.iptables in /root and the additions are made to the custom WhiteList in /usr/local/sbin/iptables-custom. To enable permanent PortKnocker additions, simply run this command while logged into your server as root: iptables-knock activate.

11. Test Drive Incredible PBX for Wazo

To give you a good idea of what to expect with Incredible PBX for Wazo, just pick up a phone and dial any toll-free number in the U.S. and Canada using a 1 prefix. We’ve also set up a sample IVR using voice prompts from Allison. Try it out from any phone on your PBX by dialing 4871 (IVR1):

Nerd Vittles Demo IVR Options
1 – Call by Name (say "Delta Airlines" or "American Airlines" to try it out)
2 – MeetMe Conference
3 – Wolfram Alpha (Coming Soon!)
4 – Lenny (The Telemarketer’s Worst Nightmare)
5 – Today’s News Headlines
6 – Weather Forecast (enter a 5-digit ZIP code)
7 – Today in History (Coming Soon!)
8 – Speak to a Real Person (or maybe just Lenny if we’re out)

What To Do and Where to Go Next?

Here are a boatload of projects to get you started exploring Wazo on your own. Just plug the keywords into the search bar at the top of Nerd Vittles to find numerous tutorials covering the topics or simply follow our links. Unless there is an asterisk (*) the components already are in place so do NOT reinstall them. Just read the previous tutorials to learn how to configure each component. Be sure to also join the PIAF Forum to keep track of the latest tips and tricks with Wazo. There’s a treasure trove of information that awaits.

Wazo and Incredible PBX Dial Code Cheat Sheets

Complete Wazo documentation is available here. But here are two cheat sheets in PDF format for Wazo Star Codes and Incredible PBX Dial Codes.

Published: Wednesday, December 14, 2016  Last Updated: Monday, December 12, 2017


Don’t forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number.



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.
 



Twofer Tuesday: $1.50 Cloud Bargains for VoIP Deployments

We’ve been big fans of $5/month VPS offerings of Digital Ocean and Vultr for many years. When Vultr reduced their lowest tier to $2.50/month, we were ecstatic. These weren’t ideal VoIP platforms because of their 512MB memory constraint, but they were perfectly suitable as a sandbox for experimentation. And then along came OVH with a 2GB VPS that was nearly perfect for VoIP at $3.49/month. As we all know, the Earth does not stand still, and WootHosting now has once again changed the landscape with two different $1.50/month offerings that include 2GB of RAM. That’s cheaper than the cost of electricity to run a server in your home or office. Never mind that you also have to purchase a server.

As most of you know, we eat our own dog food before recommending products, and we’ve deployed both the Wazo and Issabel PBXs on the WootHosting platform being reviewed today. In addition, we’ve deployed a multi-purpose web server to host more than a dozen of our personal sites using an even better second offering that we also will cover today.

The first offering (pictured above) actually provides a platform for two separate VoIP servers. For each of the servers, you have a choice of sites: New York, Miami, or Los Angeles. Why would you want two servers? The most obvious answer is redundancy. Wazo already offers High Availability (HA) redundant servers with the click of a button. Our deployment tutorial is available here. By deploying identical servers in two cities, you have a failsafe VoIP platform that can survive almost any natural or man-made disaster. And the total cost for both cloud servers is just $3 a month. A similar implementation for other Incredible PBX platforms is now under development on the PIAF Forum. Compare these free options to HA solutions from other VoIP providers costing $3,000 plus maintenance.

If a New York-based cloud offering will meet your needs, the second WootHosting offer is even more impressive with 4 CPU core allocations, 2GB RAM and swap space, a whopping 150GB of storage, 3TB of monthly bandwidth, and advanced DDOS protection for $1.50/mo.:



As we mentioned, we actually use this second VPS offering to host more than a dozen of our personal web sites without a hiccup. But it is sufficiently robust to host very large VoIP implementations with support for dozens of simultaneous calls. A deployment guide for Wazo is available here. As with all cloud-based servers, we strongly recommend redundant system deployments in separate locations. Additional WootHosting specials in their various locations are documented on the New York ordering page. Enjoy!

Published: Tuesday, August 15, 2017  



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…

Introducing IVRs in a Flash with Incredible PBX for Wazo

Last week we introduced the all-new, API-driven Admin User Interface for Wazo 17.07. But pretty pictures are only the icing on the cake. It’s what’s under the hood that really matters. And today we want to walk you through the new IVR User Interface with Incredible PBX for Wazo. The Wazo developers heard from many of you that were missing the GUI-driven IVR tools available in other products. And we’re pleased to tell you it’s another Home Run for Wazo. The UI is well-documented, easy to use, and incredibly simple to implement.

We should mention that our focus today is on the mechanics of using the new Wazo IVR Builder. We won’t be covering how to design a good IVR. If you want some great examples of poor IVR design, look no further than the main number of your favorite cellular or cable provider. Insofar as good design tips go, you can’t do better than learning from the master, Allison Smith. Here’s a link to her previous articles on Nerd Vittles.

We’re assuming you followed last week’s tutorial and added the basic components for the Admin user interface after using your browser to login to https://IPaddress/admin/. If not, start there and install the following plugins to begin: Users, Extensions, Contexts, and Devices. Next, add the plugins for Incalls, Outcalls, Trunks, and IVR. The whole setup procedure takes less than a minute. When you’re finished, your Navigation Sidebar should look like what’s shown above.

You’re obviously going to need a SIP or Google Voice trunk to handle incoming IVR calls so start there if you haven’t already set up at least one trunk that can be dedicated to your IVR. For the time being, add an Inbound Route for the Trunk that points to an extension.

Now we’re ready to create your first IVR using the new IVR template in the Admin UI. Using your web browser, log back in to https://IPaddress/admin/ and (1) click IVR in the Navigation bar. Then (2) click + Add to create a new IVR.

Since the pieces already are in place for the hard-coded Demo IVR that comes with Incredible PBX for Wazo, let’s use those components to demonstrate how easy it is to build your own IVR with the new GUI. As you can see from the completed form below, there are two main sections in setting up an IVR. There are some basic settings that handle answering the call, playing the welcome message, and managing error conditions. Then there are IVR options that correspond to the Digit pressed by the caller using a touchtone phone. Each of these options has a destination. Currently, the available options include playing a sound file, ringing an extension, running custom dialplan code, or hanging up a call. There also is the ability to nest IVRs. Once you have built the secondary IVRs, the options list will also include an IVR option.

Since our sample IVR uses custom contexts for each of the Incredible PBX applications, we’ll rely heavily on the Custom option to route calls to extensions that were created when these applications were installed: Call by Name (411), Join Conference (2663), Lenny (53669), Yahoo News (951), and Weather Forecasts by ZIP Code (947). We’ll also show you how to direct a call to an extension using the default WebRTC extension (701). To get things going, simply copy the entries shown below and then Save/Update the template. The Wazo syntax to call a Custom destination looks like this: Dial(Local/951@default).

Before you attempt to build your own IVR, keep in mind that you’ll need at least a greeting message that tells callers which buttons to press. This file should be placed in /var/lib/xivo/sounds/playback. You can also use any of the default sound recordings found in /usr/share/asterisk/sounds/en for error messages and retry alerts.

Once you have your IVR built, the last step is to adjust an incoming route for one of your DIDs so that it points to your newly created IVR. We do this using the Incalls option in the Navigation bar. If you’re adjusting an existing inbound route, just call it up and make the changes as shown below:

You’re good to go now. Call your DID from another phone and try out your first custom IVR.

Enjoy!

Published: Monday, May 22, 2017  


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.



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…

Cloud 9: VoIP on Steroids with Wazo and Vultr for $2.50/mo.



It tells you just how far we’ve come in Cloud Computing when $2.50 per month now buys you an Incredible PBX™ platform with great performance for a SOHO or home deployment. So what’s the catch? What else do you have to buy? If you’re in the United States, nothing! With a free Google Voice trunk, all of your incoming and outgoing calls to the United States and Canada are totally free. Does it matter where your phones are situated? Not at all. And how about scaling for large organizations? Not to worry! When you decide to move your entire organization to the Cloud, our friends at RentPBX stand ready to provide a 24×7 platform specifically engineered for VoIP at only $15 a month. Coupon code: NOGOTCHAS.

Getting Started with Vultr

We’ve used Vultr as our primary development platform for Nerd Vittles not only because of its price but also because of its reliability and feature set. With the recent reduction in the price of cloud instances, it’s now more than compelling as a stand-alone cloud solution for VoIP applications. An additional 50¢ a month buys you automatic daily, weekly, or monthly backups to a separate, fault tolerant storage system in the same data center. And, of course, Wazo itself generates nightly backups as well. What’s not to like? But don’t take our word for it. Walk through this tutorial with Vultr. You can even pay by the hour and kick the tires. Five hours of experimentation will set you back a whopping 2¢. Nope, that’s not a typo!

As you might imagine, the $2.50/month cloud instances at Vultr have been well received. In fact, that’s an understatement. Rather than oversubscribe users on their cloud platforms, Vultr limits the number of $2.50 cloud instances on each server. Lucky for all of us, Vultr maintains cloud hosting services in 15 cities around the world. As this is written, the $2.50 instances still were available in New York and Miami. Those locations obviously can change by the hour, but we have yet to see them disappear completely. A word to the wise: HURRY!

Getting Started with Incredible PBX for Wazo

After you have set up your account at Vultr using our referral link,1 the first step is to create your new cloud instance. Choose the desired site that has $2.50/month availability and select the 64-bit Debian 8 as your server platform. IMPORTANT: With Wazo, it’s extremely important that you leave the Server Hostname & Label blank. Otherwise, Wazo’s setup wizard will fail.

(1) Once you’ve built and started your new virtual machine, log into your server as root using SSH/Putty and immediately change your root password: passwd.

(2) With the $2.50 size VULTR virtual machine, you must create a swapfile before beginning the Incredible PBX for Wazo installation. Here are the commands:

dd if=/dev/zero of=/swapfile bs=1024 count=1024k
chown root:root /swapfile
chmod 0600 /swapfile
mkswap /swapfile
swapon /swapfile
echo "/swapfile swap swap defaults 0 0">>/etc/fstab
sysctl vm.swappiness=10
echo vm.swappiness=10>>/etc/sysctl.conf
free -h
cat /proc/sys/vm/swappiness

(3) Now you’re ready to kick off the Incredible PBX for Wazo install. The first phase is to bring Debian 8 up to current specs. Here are the commands:

cd /root
apt-get -y install cloud-init
wget http://incrediblepbx.com/IncrediblePBX14-Wazo.sh
chmod +x IncrediblePBX14-Wazo.sh
./IncrediblePBX14-Wazo.sh

(4) Once the Debian 8 upgrade is complete, reboot your server and log back in as root. Then run the Incredible PBX for Wazo installer a second time:

./IncrediblePBX14-Wazo.sh

(5) Your server will pause after the Wazo install is complete. Verify that Wazo started successfully by pressing the ENTER key.

(6) Your server then will run the Wazo setup wizard. Verify that the wizard completed without errors by pressing the ENTER key.

(7) The Incredible PBX for Wazo installer then will complete the remainder of the setup without user intervention. It takes less than 10 minutes.

(8) Now proceed to the Incredible PBX Initial Configuration tutorial to continue your setup and begin your adventure. Enjoy!

Published: Monday, May 15, 2017  


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.



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…

  1. Vultr provides a modest referral credit to Nerd Vittles for those that use our referral code. It in no way colors our recommendation of Vultr. As we noted, we use Vultr as our primary development platform, and we did so long before there were referral credits. You won’t be disappointed. []

Free SMS Messaging Returns to Incredible PBX for Wazo


We continue our Wazo adventure today with a few simple additions to the Incredible PBX for Wazo dialplan. We will enable SMS messaging both from SIP phones such as the Yealink T46G and using voice recognition from any phone connected to Wazo including WebRTC. Of course, you still can use the command line to send SMS messages to one or more recipients using SMS Message Blasting which is covered below. To implement SMS messaging, you’ll need at least one Google Voice account configured. To implement the voice recognition option, you’ll also need to first enable voice recognition with Incredible PBX for Wazo.

The prerequisites for SMS Messaging with Incredible PBX for Wazo look like this:

  1. Incredible PBX for Wazo Server
  2. Preconfigured Google Voice Trunk
  3. Activate Google Speech Recognition on your server

Activating Google Speech Recognition on Your Server. If you haven’t previously activated Google Speech Recognition on your Wazo server, you’ll need to complete this step if you want to be able to dictate SMS messages from phones connected to Wazo. Follow this tutorial to obtain a speech recognition API key and activate the Google service on your server. It only takes a couple minutes.

SMS Dictator for Wazo. True to form, Google broke the command-line SMS messaging component a few weeks ago. So let’s get that squared away before we begin. Install the latest Google Voice python scripts and the updated SMS Dictator components by issuing the following commands after logging into your server as root. You’ll need your Google Voice credentials to install the update:

cd /root
wget http://incrediblepbx.com/sms-dictator-wazo.tar.gz
tar zxvf sms-dictator-wazo.tar.gz
rm -f sms-dictator-wazo.tar.gz
./sms-dictator.sh

 
Wazo SMS Messaging Setup. You may also want to enable your SIP phones to send SMS messages directly. Login to your Wazo PBX using your favorite web browser. We need to review the existing gv.conf file by navigating to IPX Configuration → Configuration Files → gv.conf. Review the first context in the file. It should look like the following. Be sure that the third from the last line below does NOT wrap to a separate line in the Wazo editor! Insert your Google Voice credentials for GVACCT and GVPASS.

;# // BEGIN gv-outcall
[subr-gv-outcall]
exten = s,1,Set(XIVO_CALLOPTIONS=r)
same  =   n,GotoIf($["${MESSAGE(body)}" = ""]?skipsms)
same  =   n,Set(GVACCT=yourname@gmail.com)
same  =   n,Set(GVPASS=yourpassword)
same  =   n,System(/usr/bin/gvoice -e ${GVACCT} -p ${GVPASS} send_sms ${XIVO_DSTNUM} "${MESSAGE(body)}")
same  =   n(skipsms),Return()
;# // END gv-outcall

Once you get this set up and since we’ll be using plain text passwords to send the SMS messages through Google Voice, you’ll need to perform these two additional steps after first logging into your Google account with a browser using the same IP address as your Wazo server: (1) Enable Less Secure Apps and (2) Activate the Google Voice Reset Procedure. Now promptly send an SMS message from a phone registered to your Wazo server.

Dial S-M-S (767) from any phone connected to your Wazo PBX to send an SMS message. After dictating your message, you have the choice of keying in a 10-digit phone number for the SMS recipient or you can say the name of anyone in your AsteriDex phone book.



Sending SMS Messages. We obviously can’t cover the SMS messaging methodology for every SIP phone on the market. But here’s how to send an SMS message using Yealink’s T46G. First, configure one of the buttons on the phone as an extension on your Wazo PBX. Next, press the Menu button. Highlight Messages and press OK. Choose Text Message and OK. Choose New Message and OK. Type your SMS message using the keypad and press Send button. For the From: field, use the left and right arrow keys to select your Wazo extension. Press the down arrow and fill in the SMS number of your recipient just as you would do on your smartphone. Press the Send button. "Sending Message" will appear briefly on the T46G’s display. Wazo’s Asterisk CLI also will show transmission of the SMS message.

Interestingly, the same SMS functionality exists on the $29 UTP E-62 (if you can find one). Choose Menu → Applications → SMS → New. Type your SMS message using the keypad and press Send button. For the From: field, use the left and right arrow keys to select your Wazo extension. Press the down arrow and fill in the SMS number of your recipient just as you would do on your smartphone. Press the Send button. "Sending Message" will appear briefly on the UTP’s display. Wazo’s Asterisk CLI also will show the SMS transmission.

For bargain hunters that can’t find a UTP E-62, Yealink’s $50 YEA-SIP-T19P-E2 Entry-level SIP phone also appears to support SMS messaging. As with the UTP phones, you’ll need a $9 power supply unless your network supports POE.

Creating an SMS Message Blast with Wazo. Here’s how to take advantage of SMS Message Blasting using a Google Voice account with Incredible PBX for Wazo. Log into your server as root and do the following:

  1. Edit /root/smsmsg.txt and insert the text message to be sent
  2. Edit /root/smslist.txt and create a list of the phone numbers to receive the message
  3. Edit /root/smsblast and insert your gvoice username and password
  4. Run /root/smsblast to kick off the SMS Blast

Receiving SMS Messages. Typically reply messages to Google Voice numbers are forwarded either to an email address or to Hangouts. We don’t recommend enabling incoming mail on your Wazo PBX. Instead, add a New Alternate Email Address to your Google Voice account in Settings → Voicemail & Text. After verifying the new email address, set it as your Voicemail Notification email address and Save changes. Go back into Settings → Voicemail & Text and make certain that you have also checked the Text Forwarding checkbox which now should reflect your alternate email address. Now all of your incoming SMS messages will be delivered to this email address.

TIP: Google will no longer let you forward incoming SMS messages directly to another SMS destination, but you can cheat. If you have your own mail server or a non-Gmail account on which you can redirect incoming mail without verification, then simply set up the alternate email address as documented above. Then reroute that email address to point to an SMS-email gateway that forwards incoming messages to SMS, e.g. 8431234567@txt.att.net to send an SMS message to your AT&T cellphone. The complete list of providers is here. Enjoy!

Originally published: Monday, April 3, 2017



Need help with Asterisk? Come join 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…

Chasing Rainbows: The VoIP in the Cloud Trifecta



Week after week, the VoIP landscape for Cloud Computing continues to improve. And today we have more terrific news. Not only is there a new release of Wazo with simplified support for WebRTC and FollowMe roaming, but the Wazo 17.02 release also is now available on the RentPBX platform worldwide. Coupling Wazo and RentPBX with a secondary Cloud platform to achieve total VoIP redundancy is the VoIP in the Cloud Trifecta if ever there were one. RentPBX has been a platinum sponsor of Nerd Vittles for many years and, while they may not be the cheapest Cloud provider, they are certainly the best when it comes to VoIP. The reason is simple. Their cloud platform is only used for VoIP so you’re not competing for server resources with a zillion customers that are compiling millions of lines of code all day long. You also get free support! Their worldwide hosting locations translate into crystal clear VoIP calling without jitter using your favorite VoIP providers. With the Nerd Vittles NoGotchas coupon code, monthly service is just $15. For mission critical VoIP platforms, we recommend you set up Wazo with RentPBX as your primary server and configure a secondary server at OVH or Vultr.com or Digital Ocean for an additional $3.50 to $5 per month. Using Wazo’s native High Availability feature, your business gets a fault-tolerant platform with automatic failover for less than $20 a month.

Installing Incredible PBX for Wazo at RentPBX

We want to quickly walk you through the installation procedure at RentPBX because it’s the easiest cloud platform to get up and running, period. First, sign up for an account at RentPBX and order Incredible PBX for Wazo which you’ll find under the PBX in a Flash section of their site. Next, choose your favorite hosting location. We strongly recommend their Miami site if you’re east of the Rockies. For example, ping times to atlanta.voip.ms are under 14 milliseconds. The LA node works great for those on the Left Coast. Then choose Incredible PBX Wazo (Debian 8 Asterisk 14) for your platform. Enter a hostname for your server (HINT: test.rentpbx.com works fine if you don’t have your own) and click Continue. Enter NoGotchas for your Promo Code. Click Validate Code and then Checkout. Once you receive your credentials, login to your new server as root using SSH or Putty. The RentPBX setup procedure is a two-step install. First, you get Debian up to date. Then you reboot and the main Incredible PBX installer will be run.

Because of some new certificates, you will get an exim prompt during the initial phase of the install. Just type q to proceed. After initial reboot, log back in with your root credentials and complete the prompts to add your Wazo web password, a telephone reminders numeric password, and a PPTP username and password. Review your passwords carefully. Then press ENTER to proceed with the installation of Incredible PBX for Wazo. Set your time zone when prompted. After about 5-10 minutes, you will be prompted to verify that the Wazo base install completed successfully. It’s perfectly normal that some of the Wazo services are disabled at this juncture. If you see “Wazo fully booted” after the listing of services, you’re good to go. Just press ENTER to proceed. The installer then will run the Wazo Wizard. Within a minute or two, you will again be asked to verify that it completed successfully. If you see no error messages, press ENTER and go have a cup of coffee. The rest of the install will proceed without further prompting. In 10-20 minutes, your server will be ready to use.

Setting Up SIP and Google Voice Trunks with Wazo

When the installation is finished, you can make toll-free calls in the U.S. and Canada without doing anything except dialing "1″ and the 10-digit number from any phone connected to your server. For other calls, there are two steps in setting up trunks to use with Incredible PBX. First, you have to sign up with the provider of your choice and obtain trunk credentials. These typically include the FQDN of the provider’s server as well as your username and password to use for access to that server. Second, you have to configure a trunk on the Incredible PBX for Wazo server so that you can make or receive calls outside of your PBX. As with the platform tutorials, we have taken the guesswork out of the trunk setup procedure for roughly a dozen respected providers around the globe. In addition, Wazo Snapshots goes a step further and actually creates the trunks for you, minus your credentials, as part of the initial Incredible PBX install.

For Google Voice trunks, log into your server as root and run ./add-gvtrunk. When prompted, insert your 10-digit Google Voice number, your Google Voice email address and OAuth 2 token. The native Google Voice OAuth tutorial explains how to obtain it.

For the other providers, review the setup procedure below and then edit the preconfigured trunk for that provider by logging into the Wazo web GUI and choosing IPX → Trunk Management → SIP Protocol. Edit the setup for your provider (as shown above) and fill in your credentials and CallerID number in the General tab. Activate the trunk in the Register tab after again filling in your credentials. Save your settings when finished. No additional configuration for these providers is required using the Incredible PBX for Wazo Snapshot.

Directing Incoming Calls from Wazo Trunks

Registered Wazo trunks typically include a DID number. With the exception of CallCentric, this is the number that callers would dial to reach your PBX. With CallCentric, it’s the 11-digit account number of your account, e.g. 17771234567. In the Wazo web GUI, we use IPX → Call Management → Incoming Calls to create inbound routes for every DID and trunk associated with your PBX. Two sample DIDs have been preconfigured to show you how to route calls to an extension or to an IVR. To use these, simply edit their settings and change the DID to match your trunk. Or you can create new incoming routes to send calls to dozens of other destinations on your PBX.

Routing Outgoing Calls from Wazo to Providers

Outgoing calls from extensions on your Wazo PBX must be routed to a trunk provider to reach call destinations outside your PBX. Outgoing call routing is managed in IPX → Call Management → Outgoing Calls. You tell Wazo which trunk provider to use in the General tab. Then you assign a Calling Digit Sequence to this provider in the Exten tab. For example, if NXXNXXXXXX were assigned to Vitelity, this would tell Wazo to send calls to Vitelity if the caller dialed a 10-digit number. Wazo has the flexibility to add and remove digits from a dialed number as part of the outbound call routing process. For example, you might want callers to dial 48NXXNXXXXXX to send calls to a Google Voice trunk where 48 spells "GV" on the phone keypad. We obviously don’t want to send the entire dial string to Google Voice so we tell Wazo to strip the first 2 digits (48) from the number before routing the call out your Google Voice trunk. We’ve included two examples in the Wazo Snapshot to get you started. Skype Connect (shown below) is an example showing how to strip digits and also add digits before sending a call on its way:

Setting Up a Softphone & WebRTC with Wazo

If you’re a Mac user, you’re lucky (and smart). Download and install Telephone from the Mac App Store. Start up the application and choose Telephone:Preference:Accounts. Click on the + icon to add a new account. To set up your softphone, you need 3 pieces of information: the IP address of your server (Domain), and your Username and Password. In the World of Wazo, you’ll find these under IPBX → Services → Lines. Just click on the Pencil icon beside the extension to which you want to connect. Now copy or cut-and-paste your Username and Password into the Accounts dialog of the Telephone app. Click Done when you’re finished, and your new softphone will come to life and should show Available. Dial the IVR (4871) to try things out. With Telephone, you can use over two dozen soft phones simultaneously on your desktop.

For everyone else, we recommend the YateClient softphone which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the Wazo Line. You’ll need the IP address of your server plus your Line username and password associated with the 701 extension. On the Wazo platform, do NOT use an actual extension number for your username with Wazo. Go to IPBX Settings → Lines to decipher the appropriate username and password for the desired extension. Click OK to save your entries.


WebRTC allows you to use your Chrome or Firefox browser as a softphone. Extension 701 comes preconfigured for WebRTC access with Incredible PBX for Wazo. It shares the same password as the Line associated with extension 701, but the username is 701 rather than the username associated with the Line. You can decipher the password by accessing the Wazo Web GUI and then IPBX → Services → Users → Incredible PBX → XiVO Client Password. Or log into your server as root using SSH or Putty and run: /root/show-701-pw. Wazo introduces several new features to WebRTC including support for the awesome new Opus codec plus voicemail management and even Gravatar support. It’s all preconfigured!

Special Note: Beginning with this version of Wazo (17.02), WebRTC is fully integrated with NGINX on your server, and a simplified method for configuring WebRTC users has been added. When you create a new User account, simply choose the SIP (WebRTC) Protocol when creating a new user account, and all of the Advanced Line options required to support WebRTC will be preconfigured for you.

To use WebRTC, you first need to accept the different SSL certificates associated with the WebRTC app. From your browser, go to the following site and click on each link to accept the certificates. Once you’ve completed this process, visit the Wazo WebRTC site.

Before logging in, click on the Gear icon in the lower right corner and then click on the Pencil icon to edit your Settings. Fill in the public IP address of your Wazo server and specify 443 for the Port. Leave the Backend field blank and click Save. Now login to your WebRTC account with Username 701. The Password is the one you obtained running show-701-pw. The IP Address (if required) is the address of your Wazo PBX.

Implementing FollowMe Roaming with a CellPhone

In addition to ringing your SIP extension when incoming calls arrive, Wazo 17.02 can also ring your cellphone simultaneously. This obviously requires at least one outbound trunk. If that trunk provider also supports CallerID spoofing, then Wazo will pass the CallerID number of the caller rather than the DID associated with the trunk. Incredible PBX for Wazo comes with cellphone support for extension 702 ready to go. To enable it, access the Wazo Web GUI and go to IPBX → Services → Users → Incredible PBX and insert your Mobile Phone Number using the same dial string format associated with the trunk you wish to use to place the calls to your cellphone. You then can answer the incoming calls on either your cellphone or the registered SIP phone. If you answer on your cellphone, you will be prompted whether you wish to accept the call. If you press 1 after observing the CallerID, the caller will be connected. If you decline, the caller will be routed to the Wazo voicemail account of the extension.

Activating Voice Recognition for Wazo

Google has changed the licensing of their speech recognition engine about as many times as you change diapers on a newborn baby. Today’s rule restricts use to “personal and development use.” Assuming you qualify, the very first order of business is to enable speech recognition for your Wazo PBX. Once enabled, the Incredible PBX feature set grows exponentially. You’ll ultimately have access to the Voice Dialer for AsteriDex, Worldwide Weather Reports where you can say the name of a city and state or province to get a weather forecast for almost anywhere, Wolfram Alpha for a Siri-like encyclopedia for your PBX, and Lefteris Zafiris’ speech recognition software to build additional Asterisk apps limited only by your imagination. And, rumor has it, Google is about to announce new licensing terms, but we’re not there yet. To try out the Voice Dialer in today’s demo IVR, you’ll need to obtain a license key from Google. This Nerd Vittles tutorial will walk you through that process. Add your key to /var/lib/asterisk/agi-bin/speech-recog.agi on line 72.

Adding DISA Support to Your Wazo PBX

If you’re new to PBX lingo, DISA stands for Direct Inward System Access. As the name implies, it lets you make calls from outside your PBX using the call resources inside your PBX. This gives anybody with your DISA credentials the ability to make calls through your PBX on your nickel. It probably ranks up there as the most abused and one of the most loved features of the modern PBX.

There are three ways to implement DISA with Incredible PBX for Wazo. You can continue reading this section for our custom implementation with two-step authentication. There also are two native Wazo methods for implementing DISA using a PIN for security. First, you can dedicate a DID to incoming DISA calls. Or you can add a DISA option to an existing IVR. Both methods are documented in our tutorial on the PIAF Forum.

We prefer two-step authentication with DISA to make it harder for the bad guys. First, the outside phone number has to match the whitelist of numbers authorized to use your DISA service. And, second, you have to supply the DISA password for your server before you get dialtone to place an outbound call. Ultimately, of course, the monkey is on your back to create a very secure DISA password and to change it regularly. If all this sounds too scary, don’t install DISA on your PBX.

1. To get started, edit /root/disa-xivo.txt. When the editor opens the dialplan code, move the cursor down to the following line:

exten => 3472,n,GotoIf($["${CALLERID(number)}"="701"]?disago1)  ; Good guy

2. Clone the line by pressing Ctrl-K and then Ctrl-U. Add copies of the line by pressing Ctrl-U again for each phone number you’d like to whitelist so that the caller can access DISA on your server. Now edit each line and replace 701 with the 10-digit number to be whitelisted.

3. Move the cursor down to the following line and replace 12341234 with the 8-digit numeric password that callers will have to enter to access DISA on your server:

exten => 3472,n,GotoIf($["${MYCODE}" = "12341234"]?disago2:bad,1)

4. Save the dialplan changes by pressing Ctrl-X, then Y, then ENTER.

5. Now copy the dialplan code into your Wazo setup, remove any previous copies of the code, and restart Asterisk:

cd /root
sed -i '\:// BEGIN DISA:,\:// END DISA:d' /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf
cat disa-xivo.txt >> /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf
/etc/init.d/asterisk reload

6. The traditional way to access DISA is to add it as an undisclosed option in an IVR that is assigned to one of your inbound trunks (DIDs). For the demo IVR that is installed, edit the ivr-1.conf configuration file and change the "option 0″ line so that it looks like this. Then SAVE your changes.

exten => 0,1(ivrsel-0),Dial(Local/3472@default)

7. Adjust the inbound calls route of one of your DIDs to point to the demo IVR by changing the destination to Customized with the following Command:

Goto(ivr-1,s,1)

A sample is included in the Wazo Snapshot. Here’s how ours looks for the Demo IVR:



8. Now you should be able to call your DID and choose option 0 to access DISA assuming you have whitelisted the number from which you are calling. When prompted, enter the DISA password you assigned and press #. You then should be able to dial a 10-digit number to make an outside call from within your PBX.

SECURITY HINT: Whenever you implement a new IVR on your PBX, it’s always a good idea to call in from an outside number 13 TIMES and try every key from your phone to make sure there is no unanticipated hole in your setup. Be sure to also let the IVR timeout to see what result you get.

Implementing HA Redundancy with Wazo

With a business phone system, nothing is more important than never missing a call. Wazo’s High Availability (HA) option makes this a no-brainer, and it’s free! Just set up a second server either in the cloud or in your office and walk through our HA tutorial to set up the second server and activate HA. Even though located just across the border in Canada, OVH is hard to beat at $3.49 a month with 2 gigs of RAM and 10 gigs of storage. Vultr.com and Digital Ocean are also good candidates for a slave server, and the cost is still just $5 a month. Their 512MB platforms work fine with a drive cache, especially for a backup server. To get started, create a new Wazo platform using one of the highlighted links above. Be sure to use the same version of Wazo. Once the server is up and running, go to our Wazo HA tutorial and we’ll walk you through installing the NeoRouter Server and completing the Wazo setup. Be sure to configure Google Voice on the backup server before activating HA!



Test Drive Incredible PBX for Wazo

To give you a good idea of what to expect with Incredible PBX for Wazo, just pick up a phone and dial any toll-free number in the U.S. and Canada using a 1 prefix. We’ve also set up a sample IVR using voice prompts from Allison. Try it out from any phone on your PBX by dialing 4871 (IVR1):

Nerd Vittles Demo IVR Options
1 – Call by Name (say "Delta Airlines" or "American Airlines" to try it out)
2 – MeetMe Conference
3 – Wolfram Alpha (Coming Soon!)
4 – Lenny (The Telemarketer’s Worst Nightmare)
5 – Today’s News Headlines
6 – Weather Forecast (enter a 5-digit ZIP code)
7 – Today in History (Coming Soon!)
8 – Speak to a Real Person (or maybe just Lenny if we’re out)

What To Do and Where to Go Next?

Here are a boatload of projects to get you started exploring Wazo on your own. Just plug the keywords into the search bar at the top of Nerd Vittles to find numerous tutorials covering the topics or simply follow our links. Unless there is an asterisk (*) the components already are in place so do NOT reinstall them. Just read the previous tutorials to learn how to configure each component. Be sure to also join the PIAF Forum to keep track of the latest tips and tricks with Wazo. There’s a treasure trove of information that awaits.

Wazo and Incredible PBX Dial Code Cheat Sheets

Complete Wazo documentation is available here. But here are two cheat sheets in PDF format for Wazo Star Codes and Incredible PBX Dial Codes.

Published: Monday, January 30, 2017





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.

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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.
 



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