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The Most Versatile VoIP Provider: FREE PORTING

Deploying WebRTC with Incredible PBX for Wazo



We continue our open source adventure with Wazo today by introducing Sylvain Boily’s latest masterpiece, WebRTC for Wazo. What started as a simple experiment has now become a full-featured WebRTC implementation that rivals any of the commercial alternatives. Did we mention it’s FREE! Better still, when you install the latest release of Incredible PBX for Wazo with all of its modules, the key components to support WebRTC are already in place thanks to Wazo Snapshots. If you have an earlier version of Incredible PBX for XiVO, we’ve already put together a tutorial on the PIAF Forum to walk you through installing WebRTC.

If you’re new to WebRTC, this slide from AT&T covers it all:

Why WebRTC? Some of you may be asking, “What’s the big deal? Why would I want to deploy WebRTC?” The short answer is it eliminates the need to install and configure a proprietary softphone on every users’ desktop computer before they can communicate. Instead, all the user needs is a web browser that supports Real-Time Communications. By pointing their browser to https://phone.wazo.community/?serverIP=Wazo-ip-address, the user instantly gains a communications platform that’s as feature-rich as the most sophisticated softphone. Not only is it comparable to the dedicated clients of old, but there’s no associated cost nor the hassle of marrying a softphone to every user’s particular desktop operating system! And your web page could easily provide a directory of supported contact names and numbers as part of the user interface. In the case of the Wazo implementation, it does. To make a connection, all an end user needs is the latest Firefox or Chrome browser.



WebRTC Admin Setup with Incredible PBX for Wazo

We’re getting ahead of ourselves. Let’s get WebRTC set up with Incredible PBX for Wazo so your users have something to play with. If you haven’t already installed the latest Incredible PBX for Wazo, start there. This puts all the pieces in place to support WebRTC. Write down the IP address of Incredible PBX for Wazo once you complete the install. You’ll need to provide this IP address to WebRTC users.

The other piece a WebRTC user will need is the random password assigned to their WebRTC extension. Incredible PBX comes with extension 701 preconfigured. You can create additional extensions as needed. Running the /root/show-701-pw script will display the password for the default 701 extension. If you’re missing that script, running the command below from the Linux CLI will display it. Or you can log into the Wazo CLI with your browser and go to IPBX → IPBX Settings → Users. Then edit the Incredible PBX 701 user account by clicking on the Pencil icon and write down the Password assigned to the 701 Wazo Client. By the way, this will be the same password assigned to the Default SIP/m1hqy5f3 Line for the Incredible PBX user.

export PGPASSWORD='proformatique'; psql -P pager=off -U asterisk -d asterisk \\
-c "SELECT secret FROM usersip WHERE id=1"

WebRTC User Setup with Incredible PBX for Wazo

The end user needs 3 pieces of information to get WebRTC running: the IP address of the Incredible PBX for Wazo PBX as well as the end user’s username and password for an extension to be used for WebRTC communications. With those 3 pieces in hand, the actual WebRTC setup is easy.

Here are the steps for the end-user to perform:

(1) Use the extension 701 user credentials as explained above or create a new user account and password choosing SIP (WebRTC) Protocol for the account type.

(2) Using Firefox or Chrome, go to the following link: https://phone.wazo.community/

(3) Before logging in, click on the Gear icon in the lower right corner and click 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.

(4) Login to your WebRTC account with Username 701. The Password is the one you obtained running /root/show-701-pw.

(5) When prompted, authorize WebRTC to use the camera and microphone on the user’s desktop computer.



Once you’re logged in, at Enter number prompt, type in a phone number and click the Phone icon to dial.

There are loads of additional features in the Wazo WebRTC UI. Just follow your nose. Enjoy!

Published: Wednesday, October 26, 2016  Updated: Monday, May 29, 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…

The Loneliest Number: One Remaining Open Source Distro for Asterisk 14


With Asterisk® World just around the corner, this may come as a surprise to some of you. The Asterisk community that has championed open source software development for the past decade now has only one open source distro still standing. All the rest have morphed into closed source or commercial products. Can you guess which one is still carrying the Asterisk 14 open source banner? AsteriskNOW®? Nope. The FreePBX Distro®? Nope. Ombutel™? Nope. PBX in a Flash™? Nope. Elastix™? Nope. The answer is Wazo 17.01, and the latest Incredible PBX installer makes it a turnkey install in less than 15 minutes. If continuing the FOSS tradition is important to you, you really should take Wazo for a spin.


What Went Wrong? The answer is probably nothing. Reality simply set in. We all have to eat. As someone who has been involved in both the shareware and open source revolutions for more than 30 years, I can tell you that earning a living with open source software development is mostly a pipe dream. You can love open source software development and starve. Will some folks donate to the cause? Absolutely. Can you pay your mortgage from the proceeds? Not a chance. So you either find a "real job" that will pay the bills, or you change your business model and develop some sort of recurring revenue stream either through maintenance and support contracts, consulting, or hardware sales. Or you can write a technology blog and hope to find enough advertisers to keep the lights on. 🙂

We don’t mean to suggest that there’s anything wrong with commercial products per se. When it comes to VoIP telephony, commercial solutions make perfect sense. Businesses want their phones to ring when customers call. And the best way to achieve that is with commercially proven software and a support network that stands behind their products 24×7. So then it becomes a matter of comparison shopping for the best price and feature set. With this week’s release of the 3CX commercial product at zero cost to all PIAF users that participate in the PIAF Forum, that really should be a no-brainer. With a network of thousands of 3CX dealers worldwide for support, what have you got to lose? Zero.


Our New Year’s Resolution goes like this. For Nerd Vittles readers and for members of the PIAF community, we want you to have the best of both worlds. So we’ll be pushing our commercial provider to further enhance 3CX with features such as voice recognition and text-to-speech plus a robust API and programming language that makes expandability both simple and participatory. On the open source front, we will continue to work with the Wazo developers to make their platform even more flexible and feature rich than any FOSS product on the market. Please join us on both platforms as we continue our VoIP adventure.

In the meantime, come explore Wazo

Published: Tuesday, January 17, 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…

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
./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
xivo-upgrade
iptables-restart
# 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
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:

Goto(ivr-1,s,1)

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.


Coming Soon to Nerd Vittles: The Autonomous Car




 

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…

2016, The Year of VoIP Choice: Introducing Ombutel



Today we’re pleased to introduce our last (but not least) Unified Communications platform for 2016. Meet Ombutel, a terrific new GUI-based Asterisk® aggregation that was developed jointly by Telesoft and Xorcom, two familiar faces in the Asterisk community. We racked our brain trying to come up with a simple way to explain all of the things that Ombutel can do. We finally concluded "a picture [really] is worth a thousand words!"

Here’s some additional background on the Ombutel project:



As with the other platforms we’ve introduced this year, we think the best way to get started is to install it yourself and kick the tires. For those familiar with FreePBX® or XiVO®, this will be a walk in the park. You set up an Extension and Device, configure a SIP Trunk to handle your calls, define an Inbound and Outbound Route to direct calls to their proper destination, load your extension credentials into a softphone or SIP phone, and you’re done. We were making calls after loading Ombutel into VirtualBox in less than 30 minutes.

To get started, download Ombutel from their web site. The ISO is approximately 1GB in size.

Installing Ombutel. Using the console interface in VirtualBox, we kicked off the install and went through the typical CentOS 7 setup choosing a language, choosing a keyboard, selecting an install destination, and setting up a root password. When the base install completes, you can log in as root to obtain Ombutel’s IP address. All of the remaining setup is completed using a browser pointed to Ombutel’s IP address. Set up an admin password for your server. Then login as admin with your new password. The Dashboard will display.

Creating an Extension. To get started, create an Extension and let Ombutel automatically populate an associated Device: (1) PBX → (2) Extensions → (3) Extensions. The only required entries are the (4) Extension Number and (5) Name. Be sure to set the NAT entry correctly for your network. Once you’ve completed the entries, click the Save button and then the red Reload icon. Notice the list icon in the right column of the window. Clicking on the List pull-down will show all of the extensions you created and allow you to edit them and decipher whether a particular extension is active.



Adding a SIP Trunk. Adding Trunks is equally straight-forward: (1) PBX → (2) External → (3) Trunks. Then fill in the dozen items with your own credentials and settings. We’ve used a RingPlus SIP trunk as an example. NOTE: Be sure to set the From User field to your 10-digit RingPlus number even though this is not shown in the screenshot below. Once you’ve completed the entries, click the Save button and then the red Reload icon. As previously noted, the list icon in the right column will display all of the Trunks you’ve created.

Configuring an Incoming Route. As with other PBXs, incoming routes define how calls from individual DIDs are routed once they arrive. The minimum requirements to set up an Incoming Route are a Description, a DID Pattern (usually the number associated with the DID), and a Destination for the incoming calls. Once you’ve completed the entries, click the Save button and then the red Reload icon. As previously noted, the list icon in the right column will display all of the Incoming Routes you’ve created and let you edit them.

Configuring an Outgoing Route. As with other PBXs, outbound routes define how calls are routed out of your PBX based upon the dial string. You can choose one ore more trunks to associate with each Outbound Route. The dial string for each outbound route needs to be unique. Once you’ve completed the entries, click Save and then the red Reload icon.

Just to Be Safe, Restart Asterisk. Ombutel still is fairly new code. We’ve found that a quirk occurs once in a while during all of the initial configuration. This typically can be squared away (e.g. extensions not connecting) by restarting Asterisk: /etc/init.d/asterisk restart.

Setting Up a Softphone to Connect to Ombutel. 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. With Ombutel, choose PBX → Extensions → Extensions. Then click on the List icon and click on the extension to which you want to connect. Now copy or cut-and-paste your User Device number into Username and Password into Password on 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 same extension (7001) to test 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. As with the Telephone app above, you’ll need the IP address of your server plus your User Device and Password associated with the desired extension. Click OK to save your entries.

Some Thoughts on Network Security. We’ll have more to say about the Ombutel security model with FirewallD at another time. Suffice it to say, it’s not our preferred way of securing an Asterisk server. Here’s why. The following ports are all exposed by default:

1 - SIP udp and tcp 5060
2 - DNS tcp and udp 53
3 - NTP udp 123
4 - DHCP udp 67-68
5 - HTTP tcp 80
6 - SSH tcp 22
7 - RTP udp 10000-20000
8 - IAX2 udp 4569
9 - SwitchBoard tcp 4445
10 - mDNS udp 5353 224.0.0.251

You can check these for yourself in /etc/firewalld/services, and you can list the default firewall setup like this: firewall-cmd --list-all-zones. In fairness to Ombutel, their firewall design is no worse than what you will find with AsteriskNOW or the FreePBX Distro or Elastix. Incredible PBX and PBX in a Flash powered by 3CX take a different approach and don’t put all the responsibility for network security on the system administrator. We simply don’t have sufficient confidence in any Asterisk platform to risk exposing SIP, IAX2, HTTP, and SSH to the Big Bad Internet. For the time being until we can complete work on Incredible PBX for Ombutel, we recommend you run Ombutel behind a hardware-based firewall that does not expose these ports to the Internet for anyone and everyone.

Where To Go From Here. Ombutel has an awesome collection of video tutorials that should be the next stop in your Ombutel adventure. We’ve barely scratched the surface of this powerful platform, and there are still some missing pieces such as Google Voice. For the time being, you can use the Simonics SIP to Google Voice gateway to add this functionality. See this recent tutorial for some hints and a discount coupon.

An Early Stocking Stuffer from Santa. We’ll leave you with a quick tutorial on how to install FLITE so that text-to-speech can be used in your Asterisk custom dialplan.1 In addition, we’re releasing the first of many Incredible PBX components for Ombutel with our Yahoo News application. After installing it, just dial *951 from any extension to listen to the latest Yahoo News Headlines. Both FLITE and the news application are GPL2 open source code. We’ll have more goodies to share with you in coming months.

yum -y upgrade
cd /usr/src
wget http://incrediblepbx.com/Asterisk-Flite-2.2-rc1-flite1.3.tar.gz
tar zxvf Asterisk-Flite*
cd Asterisk-Flite*
yum -y install gcc asterisk-devel
make
make install
make samples
ldconfig
/etc/init.d/asterisk restart
asterisk -rx "core show application like flite"
cd /
wget http://incrediblepbx.com/nv-news-ombutel.tar.gz
tar zxvf nv-news-ombutel.tar.gz
rm -f nv-news-ombutel.tar.gz
asterisk -rx "dialplan reload"

A final cautionary note to would-be Ombutel developers. You can’t use Feature Codes such as *951 as Destinations in the Ombutel GUI. Instead, you first will need to create a Custom Application as shown below. Then you can use Custom Applications → Yahoo News as a Destination in components such as IVRs and Inbound Routes. Enjoy!



Published: Monday, November 21, 2016



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. Customizations to the dialplan can be made by creating files in /etc/asterisk/ombutel with the filename pattern “extensions__NN-*.conf” where NN defines the order in which to load the files. Numbers above 50 are strongly recommended! []

Santa’s Secret: Deploying Google Pixel as a Free VoIP Phone

Nov. 21 UPDATE: As rumored, RingPlus has now announced the termination of ALL existing plans. Details here.

NEWS FLASH: We interrupt our normal editorial schedule to bring you this update. Many of our readers joined us in embracing RingPlus cellular service with the promise of free monthly calling in exchange for a modest upfront contribution. As with almost every something-for-nothing deal, it looks as if the death spiral may have begun with the abrupt termination of more than a dozen RingPlus plans. Even if your plan was not affected in Round #1, it’s probably a good time to begin making some contingency plans particularly if RingPlus is your only cellular provider. We already had prepared the article which follows when this news broke, but you can read between the lines to see that it makes even more sense should RingPlus leave you in the lurch with a smartphone and no cellular service. Here’s the email we received announcing the following RingPlus discontinued plans:

Commitment
Commitment (Member+)
Free Plan
Future – Phase 11
Future – Phase 3
Future – Phase 5
Future – Phase 6
Future – Phase 9
Leonardo 2
Leonardo 3
Michelangelo
Midsummer Night’s Dream
Seashore (Memorial Day)(Non-Member+ Upgrade)
Truly Free
Truly Free 2
Truly Free 3


We always like a challenge. And Google’s new Pixel phone seemed like a perfect candidate to determine whether we could do everything a normal mobile phone could do (and more) using no cellular service. In other words, we wanted to set up our Pixel without a SIM card and see if there was anything we couldn’t do that we’d normally expect out of a top-of-the-line mobile phone. There’s one obvious prerequisite. The Pixel needs an Internet connection. This could be a normal WiFi network connection, or it could be a connection using an LTE-powered WiFi HotSpot, or it could be a WiFi connection established through tethering to an existing smartphone.

Why Would You Do Such a Thing? We can think of a number of reasons. Most importantly, it’s considerably cheaper than adding another mobile phone to your cellular plan unless you happen to use AT&T’s "unlimited" plan where the fourth mobile phone is free. But, typically, adding a mobile phone to your cellphone plan will cost you $50 a month or more before you make the first call. Second, some of us like the flexibility of having BOTH an iPhone and an Android phone because of differences in features and functionality. Third, it’s a perfect way to introduce younger children to mobile phone technology without spending an arm and a leg on cellular service. No, you probably wouldn’t buy them a Pixel which is priced like an iPhone. But you get the idea.

What’s a Typical Use Case for a Non-Cellular Mobile Phone? We can think of several scenarios where this makes perfect sense. We happen to have a Verizon HotSpot that’s still on an unlimited data plan. While it costs almost $100 a month, it lets 7 devices connect to blazing fast LTE service at zero additional cost. If you travel with a group of people that all need mobile phones and that typically travel or work together except when alternative WiFi service is available, this is a real deal. For those with a newer vehicle that includes a WiFi HotSpot or an OBD-II diagnostics port1 and AT&T’s $100 ZTE Mobley device, mobile phones and tablets in the car or truck work perfectly without a cellular connection. And AT&T now lets you add a vehicle’s stand-alone WiFi hotspot or ZTE Mobley to their unlimited data plan for $40 $20 a month.2 If you have four kids and a spouse, you can do the math. Finally, if you and your family or business associates spend 95% of the day either at home or in an office or car with WiFi, everyone now gets the flexibility of a mobile phone with no recurring cost.3

VoIP Requirements for a Non-Cellular Mobile Phone. As we’ve said many times, the beauty of VoIP technology is not having to put all your eggs in one basket. So there’s really no reason to deploy a single technology. In the Google world, that means you can take advantage of Google’s rich collection of messaging applications such as Hangouts and Allo and Duo while also deploying Skype, Facebook Messenger, WebRTC and SIP-based services to connect to traditional hosting providers and PBXs such as Incredible PBX and PIAF5 powered by 3CX (shown below). Today we’ll walk through the setup process for all of these. When we’re finished, you’ll have crystal-clear phone calls as well as SMS messaging with something you don’t get with a cellular provider, multiple layers of redundancy.

What Does All of this Really Cost? You obviously have to purchase either a Pixel or some other Android phone. When we’re finished today, you’ll be able to make calls as well as send and receive SMS messages in multiple ways. Calls and SMS messages to U.S. and Canada destinations are free using Google’s services. Skype-to-Skype calls worldwide are free. SMS messages sent and received using Pinger/Textfree as well as Facebook Messenger are also free of charge. Calls placed and received using a RingPlus SIP account are free up to your monthly allocation of free minutes, typically 1,000 to 5,000+ minutes per month. With calls made using a SIP softphone or WebRTC connection to an Incredible PBX or PIAF5 PBX, you only pay the standard VoIP tariff for the calls, typically less than a penny a minute for domestic calls. Calls to many international destinations are free using FreeVoipDeal.com.

Numerous SIP softphones for Android devices are available at no cost including Zoiper, CSipSimple and many others. Still others are available for less than $10 and can be installed on as many Android devices as you happen to own, e.g. Acrobits and Bria. And, of course, the 3CX softphone above is free with PIAF5. Stick with softphones with 4 stars or better!

Putting the VoIP Pieces in Place on the Pixel. Once you have your SIM-free phone in hand and you’ve gone through the basic setup using a WiFi connection in your home or office, then it’s time to add the components you’ll need to turn your Pixel into a fully-functioning VoIP phone. If they’re not already on your phone, download the following apps from the Google Play Store: Hangouts, Hangouts Dialer, Allo, Duo, Skype, Facebook Messenger, Textfree, Port Knocker,4 DynDNS Client,5 and the VoIP softphones of your choice.

We recommend reserving the Google Voice number associated with the primary Gmail account on your Pixel for use with Hangouts, Allo, and Duo. The reason is that you can’t really use these services satisfactorily while also using the same Google Voice number with Google Chat and the Asterisk® XMPP module. Last week’s Nerd Vittles tutorial will walk you through obtaining a second Google Voice number to use with Incredible PBX or PIAF5.

Pictured above is the layout we actually use. Keep in mind that the bottom row stays in place as you scroll through other screens on your smartphone. Long-press on an existing icon on the bottom row and drag it off the row. Then long-press on the app you want to add and drag it onto the bottom row. We recommend replacing the default Phone and Messaging apps with the Hangouts Dialer and Allo (as shown). We also include a SIP softphone on the bottom row which gives you multiple ways to place and receive calls.


[soundcloud url="https://api.soundcloud.com/tracks/293184354″ params="auto_play=false&hide_related=false&show_comments=true&show_user=true&show_reposts=false&visual=true" width="400″ height="300″ iframe="true" /]

But I Really Want a Cellphone Provider. Yes, we hear you. Backup cellphone service has its virtues. Here are 3 Android phones from Google ranging in price from $199 to $649 with easy payment plans ranging from $8 to $27 a month. Each gives you unlimited domestic calling as well as unlimited domestic and international texting with multiple cellphone carriers. Rates start at $20 a month plus $10/GB for data. You even get bill credits for any data you don’t use. Project Fi is worth a careful look if you’re on a budget and limit most of your data usage to WiFi connections. Here’s a great article explaining the pro’s and con’s of Project Fi after six months of actual usage. Also check out HillClimber’s comment below which documents a terrific deal with T-Mobile that provides 100 talk minutes and 5GB of 4G data monthly plus unlimited streaming of music and video for $30 a month.

Bottom Line. On the Pixel phone we have the following services activated and functioning reliably: Google Voice with Hangouts, Allo and Pinger for SMS messaging, Bria for VoIP calling with Incredible PBX for XiVO, CSipSimple and Zoiper for SIP calling with RingPlus, Facebook Messenger, Skype, plus the 3CX Dialer for calling with PIAF5 powered by 3CX. That translates into 5 different phone lines supporting free incoming and outgoing voice calls, plus 2 additional lines for free SMS messaging, plus the Facebook and Skype services to reach over a billion people worldwide at no cost. And both the PIAF5 and XiVO lines can support calls via multiple trunks using customized dial prefixes. Even with all these services running, the Pixel has sufficient horsepower to make it through a busy day, and a 15-minute charge buys you another 7 hours of cellphone usage. What are you waiting for?

Published: Wednesday, November 16, 2016



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. OBD-II port is mandatory on vehicles sold in the U.S. since 1996. But you may not need a vehicle at all. 🙂 []
  2. DirecTV service is no longer required to take advantage of AT&T’s Unlimited Data Plan offering as of mid-March, 2017. []
  3. We drove our daughter to the school bus stop in I’On Village recently and happened to check for WiFi access because the cellular service was so horrible. There were 27 separate WiFi HotSpots, all of which were secured. Seems we weren’t the only ones having difficulty with cell service in the neighborhood. []
  4. We strongly recommend setting up PortKnocker with the credentials found in /root/knock.FAQ on your Incredible PBX server. This will provide a back door to assure that you aren’t inadvertently locked out of your server by the Incredible PBX firewall while you are traveling. []
  5. You’ll need to set up a dynamic DNS client on your Android phone in order to keep your IP address updated and whitelisted with the Incredible PBX firewall. []

Free At Last: Introducing PBX in a Flash 5



Today is a big day. We are thrilled to introduce PBX in a Flash 5 powered by 3CX®. As many of you know, 3CX has been a platinum sponsor of Nerd Vittles for quite some time so this may not be a complete surprise. The good news is a new Debian-based PIAF5 ISO is now available to ease the installation process for those getting their feet wet with Linux for the first time. Debian 8 is a terrific Linux distribution used in the very best server products.

The most important change is the transition from Asterisk®/FreePBX® to 3CX. Say what, 3CX? Isn’t that a commercial product? Yes, but PIAF5 remains free for up to 8 simultaneous calls with a SIP trunk as well as 5-user web conferencing. That’s sufficient to support about 25 employees and represents a very large segment of the existing PIAF installed base. While the code is not open source, it is standards-based. Keep in mind that neither Sangoma’s FreePBX Distro® nor Digium’s AsteriskNOW® product is open source software either. When Digium decided to adopt the Sangoma business model, we decided to take a fresh look at the Unified Communications landscape. Navigating Sangoma’s licensing labyrinth coupled with the commingling of GPL modules and nagware for dozens of commercial VoIP components plus a closed source ISO was no longer an acceptable business model for us.

Some of our users prefer open source code, and we will continue to enhance Incredible PBX for XiVO in the grandest GPL tradition. But others wanted a product that offered 24×7 commercial support, and we’ve heard you loud and clear. After carefully reviewing available UC offerings, 3CX was the hands down winner in the commercial sector. Frankly, our only reservation was its Windows platform requirement. PIAF5’s new Debian ISO solves that.

In reality, what matters to users are reliability, support, upgradeability, and ease of use. 3CX has all of them in spades not to mention a feature set that is second to none. And now it’s available on the Debian platform with PIAF5.

We know some are wondering how 3CX became the new PIAF5 platform. So let’s start there.

First, the 3CX installed base includes almost 100,000 companies. That’s not downloads. And it’s not hobbyists. It’s entire companies that are actively using and relying upon 3CX for their day-to-day operations. Simply stated, 3CX is a proven, stable, and dependable product that you’d be willing to stake your business on. Many have including some of the world’s finest corporations. Stay tuned for a special PIAF5 hosting offer from our friends at Vitelity!

Second, 3CX is incredibly flexible, easy to configure, and simple to manage. Whether you’re new to PBXs or a diehard telecom guy, you’re in for a pleasant surprise when you see how intuitive 3CX is to set up and manage. Nothing comes close in the open source world.

Third, the 3CX feature set is impressive. You won’t be nickel and dimed for every component you wish to add. While there are standard and enterprise editions of 3CX as well, we think you’ll find the free version has the vast majority of components you would expect to find in any PBX, particularly for use in a home or small business. But don’t take our word for it. Review the 3CX feature comparison chart, and you can judge for yourself.

Last but not least, support is dirt cheap for end-users and free for resellers. We hope many of our long-time gurus will consider signing up as 3CX resellers and make yourself some money after all of these years wrestling with FreePBX. You won’t be disappointed!

PIAF5 deploys on premise with Linux-compatible, local hardware, or you can set it up as a virtual machine, or you can install it in the Cloud using most Linux VPS providers including Google, OVH, Digital Ocean, and Vultr. Use our referral links and take PIAF5 for a free or almost free spin for a few months while supporting Nerd Vittles. You have nothing to lose!

So there you have it. We think it was worth the wait. We encourage everyone to try out PIAF5 for yourself. And, just to repeat, Incredible PBX for XiVO isn’t going anywhere. It will remain our featured open source, GPL alternative as we move forward. And now you have a Real Choice in free alternatives with the best of both worlds, commercial and open source.

Getting Started with PIAF5 on Dedicated Hardware or a Virtual Machine. If your platform supports ISO installs, here are the simple steps to get PIAF5 up and running. First, download the PIAF5 ISO and burn it to a CD or thumb drive. Second, obtain a free license key for 3CX. Next, boot your server from the ISO image and walk through the Debian setup process. We recommend 2GB of RAM and a 20GB drive for PIAF5, but it will run on even a minimal CloudAtCost server. When the install is finished, make note of the IP address to access with a web browser to complete the setup. Enter your 3CX license key when prompted. Set up a SIP trunk with inbound and outbound call routes. Once you have the ISO and your license key in hand, the installation procedure takes less than 10 minutes.

Getting Started with PIAF5 in the Cloud. Begin by setting up a 64-bit Debian 8 platform. Obtain a free license key for 3CX. Once your Debian install is finished, log in as root using SSH or Putty and issue these commands. NOTE: What appears as the third line below needs to be added to line #2!

wget -O- http://downloads.3cx.com/downloads/3cxpbx/public.key | apt-key add -
echo "deb http://downloads.3cx.com/downloads/3cxpbx/ /" | tee /etc/apt/sources.list.d/3cxpbx.list
apt-get update
apt-get install 3cxpbx

When the initial setup finishes, choose the Web Interface Wizard and complete the install using your favorite web browser. Enter your 3CX license key when prompted. Set up a SIP trunk with inbound and outbound call routes. Done.

Configuring Gmail as SMTP RelayHost for 3CX. 3CX has a detailed tutorial explaining how to set up your Gmail account as the SMTP relay host for 3CX. Be advised that there is one additional step before Google will authorize access from an IP address it doesn’t already have for your GMail account. In addition to Enabling Less Secure Apps (as covered in the 3CX tutorial), you also will need to activate the Google Reset Procedure while logged into your Gmail account. Otherwise, Google will block access. Once you have configured Gmail as your relay host and performed the two enabling steps above, immediately test email delivery within the 3CX GUI while Google security is relaxed: Settings → Email → TEST.

Free Calling in the U.S. and Canada with PIAF5. We know our more frugal U.S. residents are wondering if there’s a way to make free calls even with 3CX. You didn’t really think there would be a release of PBX in a Flash without Google Voice support, did you? It’s easy using the Simonics SIP to Google Voice gateway service. Setup time is about a minute, and the one-time cost is $4.99 using this Nerd Vittles link. Setup instructions for the 3CX side are straight-forward as well, and we’ve documented the procedure on the PIAF Forum.

Free Calling Worldwide with SIP URIs. There’s another free calling option as well. PIAF5 and 3CX support worldwide SIP URI calling at no cost. As part of the PIAF5 install procedure, 3CX registers an FQDN for you with one of the 3CX domains if you indicate that your server has a dynamic IP address. Unless you really know what you’re doing with DNS, it’s a good idea to tell 3CX you have a dynamic IP address whether you do or not. Here’s why. Once you have an assigned FQDN in the 3CX universe, one very slick feature is the ease with which you can publish a SIP URI address for any or all of your 3CX extensions thereby allowing PIAF5 users to receive calls from any SIP client worldwide at no cost. Setup takes less than a minute. It’s as easy as 1-2-3. Here’s how:

1. Login to the 3CX GUI and go to Settings → Network → FQDN. Tick "Allow calls from/to external SIP URIs" and make note of your FQDN, e.g. mypiaf5server.3cx.us. Click OK.

2. For an extension to enable (e.g. 001), go to Extensions → Edit 001 → Options → SIP ID and create any desired SIP URI alias for this extension, e.g. billybob. Click OK.

3. Anyone with a SIP client anywhere worldwide can now call extension 001 using SIP URI: billybob@mypiaf5server.3cx.us.

SMS Messaging with PIAF5 and Google Voice. Just to demonstrate why you’re going to love the new PIAF5 platform, here’s a sneak peek at one of many applications which are on the way with Incredible PBX for PIAF5. Meet SMS Messaging. First, complete the two Google enabling steps documented in the Gmail SMTP RelayHost section above: Enable Less Secure Apps and Activate Google Reset Procedure. Then install the Google Voice CLI tools as root:

cd /root
apt-get -y install python-setuptools
wget http://incrediblepbx.com/install-gv-cli
chmod +x install-gv-cli
./install-gv-cli

To Send an SMS Message Blast to one or more destinations, (1) create a message in /root/smsmsg.txt, (2) specify the SMS numbers in /root/smslist.txt, (3) insert your Google credentials into /root/smsblast, and (4) run /root/smsblast to send the message. Enjoy!

Published: Wednesday, October 19, 2016




 

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 Incredible PBX with XiVO Snapshots

If you’ve been following along in our XiVO adventure with Incredible PBX, 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. Today we’re pleased to report that approach is now going the way of cars with a stick shift. If you want to continue to manually configure your XiVO PBX, you still have that option. Just jump to the original tutorial and run the installer choosing the options you wish to activate. But if you prefer a self-driving Tesla, that’s now an option as well. Continue reading, and we’ll walk you through using XiVO Snapshots.

A XiVO Snapshot is just what the name implies. It’s a snapshot of a working XiVO PBX that has virtually everything already configured: SIP settings to work with Asterisk®, a SIP extension to work with a SIP phone, or softphone, or WebRTC plus your cellphone, SIP and Google Voice trunk setups for most 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 XiVO installer with all Incredible PBX options enabled. Then you tell XiVO 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 you’re already enjoying a fully functional PBX.

Incredible PBX 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 for XiVO. 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 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 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
./IncrediblePBX13-XiVO.sh

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.

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 XiVO 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:

Goto(ivr-1,s,1)

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 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 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, October 10, 2016



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…

Type It or Say It: Asterisk SMS Messaging Returns with Incredible PBX for XiVO


We continue our XiVO adventure today with two simple additions to the Incredible PBX for XiVO dialplan that enable SMS messaging both from SIP phones such as the Yealink T46G and using voice recognition from any XiVO phone. 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 on your Incredible PBX for XiVO server.

The prerequisites for SMS Messaging from a SIP phone with XiVO look like this:

  1. Incredible PBX for XiVO Server
  2. Preconfigured Google Voice Trunk
  3. SIP Phone capable of SMS Messaging, e.g. Yealink T46G 1

SIP Phone SMS Messaging. To begin, login to your XiVO PBX using your favorite web browser. We need to edit the existing gv.conf file by navigating to IPX Configuration → Configuration Files → gv.conf. The first context in the file should look like this:

[subr-gv-outcall]
exten = s,1,Set(XIVO_CALLOPTIONS=r)
same  =   n,Return()

Replace the entire context by cutting and pasting the following code and substituting your actual Google Voice account name and password for yourname and yourpassword below. Then Save the file changes leaving the Reload Dialplan option checked. Be sure that the third from the last line below does NOT wrap to a separate line in the XiVO editor!

;# // 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: (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 XiVO server.



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 XiVO 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 XiVO 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. XiVO’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 XiVO 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. XiVO’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.

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 XiVO 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.

SMS Dictator for XiVO. Okay. We hear you. Yes, typing SMS messages with a 12-button keypad can be tedious especially if your message is sprinkled with S’s. Pressing the 7 key eight times for every "s" in your text message is painful. If you’ve activated voice recognition on your Incredible PBX for XiVO server, then you can simply dictate your SMS messages by first dialing 767 (S-M-S) from any phone connected to your XiVO PBX. 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.

To install SMS Dictator on your Incredible PBX for XiVO server, issue the following commands and enter your Google Voice account name (with @gmail.com) and password when prompted:

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

3/2/2017 Update: A patched version of pygooglevoice to support SMS messaging is now available here.

Now simply dial S-M-S (767) from any phone connected to your XiVO PBX to send an SMS message. Enjoy!

Originally published: Monday, October 3, 2016



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.
 



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