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2016, The Year of VoIP Choice: Meet Wazo and XiVO 16.15

UPDATE: Wazo 17.01 has been officially released. The complete tutorial is available here.

And you thought the excitement was over for 2016. Well, not so fast. The core development team at XiVO has now forked the project so this will be the last XiVO-branded release until Wazo 16.16 hits the street. Nothing has changed except the name and a boatload of new features with more to come including a new GUI interface a little further down the road. And you’ll have a front row seat at Nerd Vittles. But lets save that discussion for coming weeks. For today, we’ll set the stage with the latest development release of XiVO featuring Incredible PBX and Asterisk® 14.1.2. Yes, there is an easy migration path for every existing XiVO server. That’s what the 2-minute xivo-upgrade is all about. In the meantime, anyone with the pioneering spirit can take a glimpse into the future. If you know XiVO, then you know that development releases normally are almost as stable as production releases because of their unique development methodology and enormous test suite which checks every change for naughty or nice. And, yes, the development team eats their own dog food! But please note that this is a Development Version of Wazo which means changes are happening regularly. The official release will be available in early December. For the pioneers installing now, be advised that there may be install hiccups from time to time as the developers migrate older components to Wazo. If an install fails for you, don’t get frustrated. Just wait 12 hours and try again.

Introducing the Opus Codec and Asterisk 14

We think you will enjoy this first release of Incredible PBX 14 featuring XiVO 16.15 and Asterisk 14 with integrated support for the Opus codec. If you haven’t heard of Opus, you’re in for a treat. You get the wideband voice quality of G.711U (ULAW) calls requiring 80-90kbps of bandwidth using only 16kbps. And, because it’s a variable bandwidth codec based upon your available Internet pipe, Opus can support narrowband calls with equivalent call quality to G.729 and Speex. Simply stated, you can squeeze FIVE wideband calls into the same bandwidth that one ULAW call used to consume. And, when you have the Internet capacity to support it, Opus calls can scale up to 128kbps for MP3-quality sound. Details.

There’s more good news with Opus. XiVO’s WebRTC client now is preconfigured with the Opus codec when you deploy Incredible PBX 14. And, as if that weren’t enough, the WebRTC client with XiVO 16.15 now includes integrated voicemail support so you can play and delete voicemails without ever leaving the WebRTC client. See our WebRTC tutorial for more.

Finally: A New CDR Reporting Module for XiVO

Here’s another important development that many have requested. The Incredible PBX 14 platform includes a terrific new CDR Reporting Module from Bart Fisher on the PIAF Forum. In the XiVO GUI, goto IPX → Call Management → Call Logs:

FLITE TTS Implemented with New Voices

We’re pleased to announce that FLITE 1.4 is now included in Incredible PBX 14 builds on or after November 26. For the first time, you now have a choice of four different voices:

kal (American male)
rms (American male)
awb (Scottish male)
slt (American female)

While it’s a matter of personal taste, the RMS and SLT voices are dramatic improvements over the previous FLITE implementation. To change the voice, edit /etc/asterisk/flite.conf and replace voice=slt with your favorite. Then restart Asterisk. This post on the PIAF Forum includes dialplan code and will walk you through installing FLITE on existing servers. There’s more good news. You now can build your own FLITE voice for use with FLITE.

The Future Vision for Wazo

We don’t want to spill the beans on everything that lies ahead, but let’s talk briefly about the API Framework behind what will soon be the Wazo Telephony Business Engine. With Incredible PBX 14, you will note that you now have direct access to all available XiVO APIs with more to come. Using a browser, head over to https://ServerIPaddress/api/. A series of tutorials on how to use these APIs will be forthcoming now that we’ve gotten a few lessons from Sylvain Boily. Suffice it to say, the idea behind these APIs is that any developer will be able to quickly produce a customized web GUI for Wazo using nothing but API calls in conjunction with open source web development tools such as Bootstrap and Smarty. Think of it as OpenStack for the Telephony Cloud. And a new Wazo GUI is in the works as well. Here are a few examples to give you some idea of what’s possible in just a matter of hours:

Rather than having a hard-coded GUI that uses spaghetti code to generate obscure Asterisk commands, you now will have a fully-documented development platform where the sky’s the limit. Think of it. You can actually contribute code back to the project while developing custom solutions for your organization. It’s what open source development is all about!

Update Your Address Book: New Wazo Links

Incredible PBX 14 for XiVO 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 14 for XiVO. By the way, the whole process takes less than an hour, half of that in the Cloud.

  1. Set Up Desired PBX Platform: Stand-alone PC, Virtual Machine, or Cloud-Based Server
  2. Run the Incredible PBX for XiVO installer and Activate All Options
  3. Set Up One or More SIP or Google Voice Trunks for Your PBX
  4. Tell XiVO Where to Direct Incoming Calls from Each Trunk
  5. Tell XiVO 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. Activating DISA with Incredible PBX for XiVO (if desired)
  10. Test Drive Incredible PBX for XiVO

1. Incredible PBX for XiVO Hardware Platform Setup

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. The last four 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.

If your situation falls somewhere in between all of these, here’s a quick summary. For stand-alone systems and virtual machine platforms that you own (such as VirtualBox and VMware ESXi), download and install the 64-bit version of XiVO using the XiVO ISO. For most other virtual machine platforms in the Cloud, you’ll start by creating a 64-bit Debian 8 virtual machine with at least 1GB of RAM and a 20GB drive.

2. Running the Incredible PBX for XiVO 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 13 for XiVO 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. You will be prompted whether to activate about a dozen applications for Incredible PBX. Choose Y for each option if you want to take advantage of the XiVO Snapshot with all components preconfigured. Otherwise, you’ll need to jump over to the original tutorial and manually configure all of the XiVO components.

cd /root
wget http://incrediblepbx.com/IncrediblePBX13-XiVO.sh
chmod +x IncrediblePBX13-XiVO.sh

When you have completed the Incredible PBX 13 install, you then can log into your server as root and upgrade to Incredible PBX 14 with Asterisk 14 and the development version of XiVO/WAZO. Here are the steps:

xivo-dist xivo-dev
/etc/init.d/netfilter-persistent stop
# restore Incredible PBX module and ODBC configuration
cp -p /etc/asterisk/modules.conf /etc/asterisk/modules.conf-new
cp -p /etc/asterisk/res_odbc.conf /etc/asterisk/res_odbc.conf-new
cp -p /etc/asterisk/modules.conf.dpkg-old /etc/asterisk/modules.conf
cp -p /etc/asterisk/res_odbc.conf.dpkg-old /etc/asterisk/res_odbc.conf
# add Google Voice OAuth support for Asterisk 14
cd /usr/src
git clone https://github.com/sboily/asterisk-res-xmpp-oauth.git
cd asterisk*
make patch
make install
xivo-service restart
# put the Incredible PBX web add-ons back in place
cd /
wget http://incrediblepbx.com/incredible-nginx.tar.gz
tar zxvf incredible-nginx.tar.gz
rm -f incredible-nginx.tar.gz
ln -s /etc/nginx/locations/https-available/01_incrediblepbx /etc/nginx/locations/https-enabled/.
cd /etc/nginx
wget http://incrediblepbx.com/nginx-config.tar.gz
tar zxvf nginx-config.tar.gz
rm -f /etc/nginx/sites-enabled/default
/etc/init.d/nginx restart
sed -i 's|13.|14.|' /etc/pbx/.version

While this may sound convoluted, there’s a reason for it. The WAZO Development Version is undergoing some major plumbing changes which affect the PostGreSQL database structure. Because Incredible PBX uses database snapshots to preconfigure a number of components, there would be major breakage if the Dev version database structure was different than the Incredible PBX snapshot. By performing an upgrade, we avoid the problem while preserving all of the Incredible PBX settings.

3. Setting Up SIP and Google Voice Trunks with XiVO

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 XiVO 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, XiVO Snapshots goes a step further and actually creates the trunks for you, minus 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 XiVO 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 when using the XiVO Snapshot.

4. Directing Incoming Calls from XiVO Trunks

Registered XiVO 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 XiVO 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 XiVO to Providers

Outgoing calls from extensions on your XiVO 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 XiVO 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 XiVO to send calls to Vitelity if the caller dialed a 10-digit number. XiVO 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 XiVO 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 XiVO 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 Softphone & WebRTC to Connect to XiVO

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 XiVO, 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 XiVO Line. You’ll need the IP address of your server plus your Line username and password associated with the 701 extension. On the XiVO platform, do NOT use an actual extension number for your username with XiVO. 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 XiVO. 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 XiVO Web GUI and then IPBX → Services → Users → Incredible PBX → XiVO Client Password. Or you can log into your server as root and run: /root/show-701-pw

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. The Username is 701. The Password is the one you obtained above. The IP Address is the address of your XiVO PBX.

7. Setting Up a CellPhone Extension with XiVO

In addition to ringing your SIP extension when incoming calls arrive, XiVO can also ring your cellphone simultaneously. This obviously requires at least one outbound trunk. If that trunk provider also supports CallerID spoofing, then XiVO will pass the CallerID number of the caller rather than the DID associated with the trunk. Incredible PBX for XiVO comes preconfigured with cellphone support for extension 701. To enable it, access the XiVO 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 can answer the incoming calls on either your cellphone or the phone registered to extension 701.

8. Activating Voice Recognition for XiVO

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 XiVO 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. Don’t forget to add your key to /var/lib/asterisk/agi-bin/speech-recog.agi on line 72.

9. Adding DISA Support to Your XiVO 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 XiVO. You can continue reading this section for our custom implementation with two-step authentication. There also are two native XiVO 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 XiVO 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:


A sample is included in the XiVO Snapshot. Here’s how ours looks for the Nerd Vittles XiVO 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. Test Drive Incredible PBX 14 for XiVO

To give you a good idea of what to expect with Incredible PBX for XiVO, we’ve set up a sample IVR using voice prompts from Allison. Give it a call and try out some of the features including voice recognition. Dial 1-843-606-0555.

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 Baker’s Dozen projects to get you started exploring XiVO 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. Note that all of these 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 XiVO. There’s a treasure trove of information that awaits.

XiVO and Incredible PBX 14 Dial Code Cheat Sheets

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

Published: Monday, November 28, 2016

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

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  1. Builtin Opus support is good news for WebRTC and ObiHai devices, but I can’t think of a single VOIP provider who supports it.

    [WM: Give them a little time. :-)]

  2. I don’t see mention of Raspberry Pi’s on this release?

    [WM: The RasPi version never has been maintained by the core development team or anyone affiliated with the company. Even assuming they continue to support WAZO, they’ve never released development versions of the software. We really can’t recommend the RasPi as a WAZO platform. The Asterisk real time environment is extremely disk intensive, and that doesn’t bode well for microSD drive longevity.]

  3. WTF? What is happening with XiVO? have the developers had a falling out? The source on GitHub has been pulled, the forums have gone. Very concerning, we just finished migrating our business to XiVO 16.13.

    [WM: Not to worry. The sky is not falling. There was a change in management in the French arm of the company. The original developers in Canada and the French company have decided to move in different directions. Because of trademark issues, there will be a new release called WAZO from the original XiVO developers in early December. You’ll be able to easily upgrade to WAZO 16.16 from your existing XiVO 16.13 platform. Same code base as before. That’s what open source and the GPL are all about. The previous XiVO Forum has a new address. The previous XiVO WebRTC site has a new address as does the new GitHub forked repo.

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