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The Most Versatile VoIP Provider: FREE PORTING

Integrating SIP URIs into XiVO for Free Worldwide Calling

It’s been a while since we’ve explored SIP URIs and all of the advantages that SIP URI calling brings to your PBX. Number one on that list is FREE calling to and from anyone on the planet so long as both of you have an Internet connection with a SIP phone or a VoIP server such as Incredible PBX for XiVO. SIP URIs are the fundamental building blocks for VoIP technology. Consider this. If everyone in the world had a SIP address instead of a phone number, every call to every person in the world via the Internet would be free. That pretty much sums up why SIP URIs are important. The syntax for SIP URIs depends upon your platform. With Asterisk® they look like this: SIP/somebody@FQDN.yourdomain.com. On SIP phones, SIP URIs look like this: sip:somenameORnumber@FQDN.yourdomain.com. Others use somenameORnumber@FQDN.yourdomain.com. Assuming you have a reliable Internet connection, once you have “dialed” a SIP URI, the destination SIP device will ring just as if the called party had a POTS phone. Asterisk® processes SIP URIs in much the same way as other calls originating from trunks and, as noted, SIP URI calls of any duration to anywhere are free. Today we’ll show you how to set things up on your XiVO PBX without exposing any ports to the Internet in a way that would jeopardize your server’s security.

Placing Outbound SIP URI Calls with a SIP Softphone

There are two ways to place outbound SIP calls. You can use a SIP phone or softphone that supports SIP URI calling to dial SIP URIs directly. If you have a Mac, the best free softphone for SIP URI calling is Telephone which you can download from the App Store. On other platforms as well as Macs, Zoiper is a great no-cost option. Both of these softphones support the sip:someone@FQDN.yourdomain.com syntax. An excellent way to test this is to call our friend Lenny and strike up a conversation: sip:2233435945@sip2sip.info.

Configuring Outbound SIP URIs with XiVO

The major drawback of SIP URIs is they’re difficult both to remember and to dial. It’s much simpler to dial a short number using a traditional phone. And, with Incredible PBX for XiVO, it’s easy to create custom extensions that can be accessed simply by dialing a few digits from any phone connected to your server. Here’s how to set it up in the XiVO GUI.

1. Create a User and assign the Customized Protocol and an Extension Number to that user:

TIP: If you’d prefer to use a different series of numbers for speeddials so you don’t get them mixed up with your standard extension numbers, just add a new range of numbers for XiVO: IPX Configuration → Contexts → Default → Users. Then choose one of them above.

2. Access the new Line that was generated for the new User:

3. Replace the Interface entry for the Line with the desired SIP URI for your speeddial, e.g. SIP/2233435945@sip2sip.info. Then SAVE your new Line settings.

4. Dial 750 from an Extension on your XiVO PBX to try out Lenny using your new SIP URI.

A Better Way to Create SpeedDials with XiVO

We’ve gone through the XiVO GUI approach to demonstrate that it is indeed possible to create speeddials for SIP URIs. However, there is a better way unless you’re one of the naysayers that believes everything is better in a GUI. If you have dozens or even hundreds of speeddials to create, you may change your mind. The GUI approach could obviously become tedious. Instead, with one line of Asterisk dialplan code, you can create as many speeddials as you like keeping in mind that it’s your responsibility to assure that SIP URI extension numbers don’t conflict with existing extensions on your server. Insert a new section of code at the bottom of /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf and reload your dialplan: asterisk -rx "dialplan reload".

You can also insert this code from within the XiVO GUI itself: IPX Configuration → Configuration Files. Edit xivo-extrafeatures.conf and insert the following code snippet at the end of the file and Save your entries. The dialplan will be reloaded automatically.

Some of our favorites include the following:

;# // BEGIN SpeedDials
exten = 882,1,Dial(SIP/200901@login.zipdx.com)     ; V-U-C on Fridays at noon EST
exten = 8378,1,Dial(SIP/thetestcall@getonsip.com)  ; T-E-S-T everything VoIP
exten = 53669,1,Dial(SIP/2233435945@sip2sip.info)  ; L-E-N-N-Y
exten = 68742,1,Dial(SIP/0289304@zero-nine.biz)    ; M-U-S-I-C
exten = 3733411,1,Dial(SIP/411@ideasip.com)        ; F-R-E-E-4-1-1 Directory Asst
;# // END SpeedDials

Creating a SIP URI Address for Your XiVO PBX

Free calls to other folks is only half of the story, of course. You’re also going to want a way for people to call you without incurring charges for the calls. There are many SIP URI approaches for inbound calls. Most of them are not safe with Asterisk. Let me say that again. Most of them are not safe with Asterisk. The reason is because most of them force you to open SIP access to your server for everybody in the world. Unfortunately, that means they can not only call you, but they can also attempt to use your extensions and trunks to place very expensive calls to others. Don’t even think about opening the SIP floodgate by exposing port 5060 unless Bill Gates sends you a check every week. You’ve been warned!

Setting Up an iNum SIP URI Trunk with XiVO

The better and safer way to add SIP URI connectivity to your XiVO server is to first obtain a freely available iNum DID from one of the many providers that support iNum and then use that provider as a SIP intermediary. All SIP calls pass only over your registered trunk with your provider. Our favorites in no particular order are VoIP.ms, LocalPhone and CallCentric. There are many, many others. In order to obtain a free iNum DID, you will need an account with one of these providers. All require some sort of minimal deposit, but you usually can get back unused funds if you decide to close your account down the road. Our XiVO tutorials for VoIP.ms, LocalPhone, and CallCentric will walk you through creating your SIP account and registering it with your XiVO server. Then verify that your SIP account is registered:

asterisk -rx "sip show registry"

Configuring an iNum DID with VoIP.ms

Our trunk tutorials for LocalPhone and CallCentric will walk you through their setup procedures for iNUM DIDs. VoIP.ms provides more flexibility in redirecting trunks so let us quickly walk you through their procedure. Log in to your VoIP.ms account and then order your free iNum DID at this link. Your iNum DID then will appear in your DID Listing here. Write down your iNum DID which you’ll need in a minute to configure the XiVO side of things. Then click on the Edit DID icon beside your iNum DID and assign the DID to your registered Main Account or the SubAccount that you’ve already registered with XiVO. Be sure to use the same DID POP that you used when you registered your VoIP.ms account with XiVO. Don’t enable VoiceMail and set the ring time to 60 seconds just to keep things simple.

Configuring XiVO to Support Your iNum DID

Now for the XiVO part. Using a browser, log into the XiVO GUI. Navigate to IPX Configuration → Contexts → Default → Users. For VoIP.ms and LocalPhone, add a new Number Range starting and ending with your iNum DID. Then click Save. For CallCentric, do the same thing but substitute your CallCentric username which will be an 11-digit number starting with 1777.

Repeat the above in IPX Configuration → Contexts → from-extern (Incalls) → Users.

For CallCentric only, also click on the Incoming Calls tab and add a new Number Range. For the Starting value, use your 11-digit LocalPhone username. For the DID length, set it to 11. You do NOT need to include a Number Range ending value. Click Save when you’re finished.

For VoIP.ms, navigate to IPX Settings → Users. Then Add a new User for your iNum DID. In the General tab, name the User VoIP.ms iNum. In the Lines tab, provide your actual iNum DID number. This must be the same number you added to the Number Range in the Default context above. In the No Answer tab, set the Fail option to the Destination of your choice, e.g. an extension, a ring group, an IVR, etc. Then click Save.

For LocalPhone, navigate to Call Management → Incoming Calls and Add a new Inbound Route for your DID specifying the destination for the calls using your iNum DID number:

For CallCentric, navigate to Call Management → Incoming Calls and Add a new Inbound Route using your 11-digit CallCentric username as the DID. Then specify the destination for the calls and click Save.

Calling Your XiVO PBX Using Your iNum SIP URI

To receive SIP URI calls safely on your iNum DID, your SIP URI is your iNum DID number followed by @sip.inum.net, e.g. 883510012345678@sip.inum.net. Neither the identity of your XiVO PBX or your SIP service provider is ever exposed. Enjoy your safe, free calling!

Originally published: Monday, September 26, 2016





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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…

Never Miss a Meeting: Google Calendar Alerts for XiVO




Today we’re pleased to dust off an oldie but goodie and to introduce Google Calendar integration for XiVO and Asterisk® 13.1 This gets you a reminder call at any number you choose based upon the Notification time that is set whenever you schedule a meeting or appointment in your Google Calendar. Our special thanks to Terry Wilson for his pioneering work at Digium® on the calendaring API. Together with the flexibility that XiVO affords out of the box, it made this incredibly easy, and the new design makes it simple to support as many Google calendars as you would like.



Calendar Design Methodology. The design of contexts in XiVO makes it easy to keep separate development projects separate. In this way, adding and changing code is straight-forward without having to worry about breaking some other working application in Asterisk. In keeping with this design, Incredible PBX for XiVO separates out different applications into different configuration files in the /etc/asterisk/extensions_extra.d directory. For calendar reminders, we’ve created the calendars.conf template which tells Asterisk what to do when an appointment reminder is triggered for the default myGoogleCal calendar.

As you can see below, this is standard Asterisk dialplan code so you can make it as sophisticated as you like. And you can add separate extensions to manage different calendars and do different things. To get you started, here’s what gets generated when you run today’s setup script. It tells XiVO to place a call to the notification number you have chosen and play a message using the Festival TTS engine that announces the time and location of a scheduled event from your Google Calendar:

[calendars]

exten => 225,1,Answer
exten => 225,n,Wait(1)
exten => 225,n,Festival("Here is an appointment reminder from your Google calendar: ")
exten => 225,n,Festival("${CALENDAR_EVENT(summary)} at ${STRFTIME(${CALENDAR_EVENT(start)},,%l:%M %p: local time %A)} at ${CALENDAR_EVENT(location)} ${CALENDAR_EVENT(description)}. Have a nice day. Good bye.")
exten => 225,n,Hangup

The Asterisk design for calendar reminders is equally intuitive. For each calendar, you create a context in /etc/asterisk/calendar.conf to identify the calendar, your credentials, and the Asterisk context, the extension and notification phone number which will be triggered when a calendar reminder (known as a Notification) is triggered. In the initial setup today, we will generate the [myGoogleCal] context with all of the required pieces:2



Once the setup is finished, you can run the following command to tell you the status of your Google Calendar upcoming events for the next two hours. These events get refreshed by Asterisk every 10 minutes based upon the current entries in your calendar.

asterisk -rx "calendar show calendar myGoogleCal"



Once you integrate Google Calendar into Incredible PBX, you can use it to alert you to upcoming appointments. You can also schedule and activate conference calls, and even log the date, time, length, and recipient of all your outbound and/or incoming phone calls. For today, we’ll get the basic pieces installed and functioning. And we’ll set up a simple reminder system based upon appointment entries in your Google Calendar. We’ll also give you some good reference materials so that you can take it from there. This is one of Asterisk’s most powerful features and one where a little study will reward you handsomely. And, as we said, XiVO makes it easy!

Prerequisites. As mentioned, you’ll need a Gmail account to which your Google Calendar is attached. NOTE: Gmail accounts with 2-Step Verification cannot be accessed using your regular Gmail password, but there’s a simple work-around covered in this footnote.3 You also need to first install Incredible PBX for XiVO. By default, you’ll need at least one outbound trunk that supports calls in the same format that you plan to use with your calendar alerts. For NANPA trunks, it will be a 10-digit number, but any other dial string is supported so long as you have enabled AND tested it in XiVO. Then just follow along in today’s tutorial. The entire setup process only takes a couple minutes, and the setup script is licensed as GPL code so knock yourself out and make any changes desired to meet your own requirements.

Installation. Once you have Incredible PBX for XiVO up and running, make a test call from any extension to the phone number you plan to use for Google Calendar notifications. When you’re satisfied that you can reach that number from a XiVO extension, then it’s time to download the installer and run it. Log into your server as root and issue the following commands. Plug in your Gmail credentials and the 10-digit (or other format) notification number when prompted. If you decide to change your Google Calendar or notification number down the road, simply run the setup script again. Be advised that it will always overwrite all of the existing contents of /etc/asterisk/calendar.conf and /etc/asterisk/extensions_extra.d/calendars.conf.

cd /root
wget http://incrediblepbx.com/setup-xivo-cal.sh
chmod +x setup-xivo-cal.sh
./setup-xivo-cal.sh

Test Drive. The proof is in the pudding, as they say. So set up an appointment in your Google Calendar that’s about an hour in the future, and set the Notification time to 30 minutes. You should receive a call in about 30 minutes assuming your Google Calendar and XiVO server are in the same time zone. For additional tips and tricks, start here. Enjoy!



Originally published: Monday, September 19, 2016



Don’t forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your numbers to the Do Not Call Registry. Or just call 888-382-1222 from your new number. Last but not least, set up the FCC’s BlackList on your server and kiss the RoboCallers goodbye.
 

 






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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Some Recent Nerd Vittles Articles of Interest…

  1. We originally introduced this five years ago in the Asterisk 1.8 days. We’ve now upgraded everything to support Incredible PBX for XiVO with Asterisk 13. []
  2. Note that you can set an AutoReminder to schedule a call for every appointment at the same preconfigured interval. This will override all individual Notifications that are set on a per-appointment basis. []
  3. For Google accounts with 2-Step Verification, simply create an Application-Specific Password. Select the Application (Calendar), select your device (Other: XiVO), and click Generate. When setting up Google Calendar Alerts for XiVO, use your actual Gmail address and substitute the 16-character password that is generated for your standard Google password. []

2016, The Year of VoIP Choice: Redundancy and Multi-Tenant with Wazo



As we celebrate Labor Day, it seemed appropriate to document why Wazo separates the men from the boys so your phones don’t end up as boat anchors buried in the sand. Today our focus is "High Availability (HA)" and "Multi-Tenant (MT)", two very expensive options for many PBXs including some that loosely tout their platforms as free.

In the PBX context, HA means that, when your server fails, there’s another one waiting in the wings to automatically take over. Much of this technology is based upon open source tools, but Sangoma sells a pair of limited term licenses as a FreePBX® add-on for a cool $3,000 not including hardware AND annual maintenance fees. With Wazo, it’s FREE! You can pair two Raspberry Pi’s or two Cloud servers, or you can mix-and-match with any combination of servers you choose. Here’s how we did it in 3 minutes flat:



Multi-tenant has been discussed for the FreePBX platform for the better part of a decade. As best we can tell, it’s still a pipe dream. Virtual machines running separate servers are the suggested solution even though this requires managing multiple Asterisk platforms forever. With Wazo’s FREE Entities module, MT is a cake walk. We’ll walk you through the 5-minute setup process thanks to the tips provided by Amy Grant on the PIAF Forum.

Deploying Wazo HA Servers with NeoRouter

Here’s the HA setup drill. First, you build two identical Wazo platforms running the same version of Incredible PBX for Wazo. Then you set the first server up as the Master and the second one as the Slave. As we said, these servers don’t need to be on the same hardware platform. And they need not be colocated although they have to share the same private LAN. We’ll handle that little detail by taking advantage of the NeoRouter client software that’s already installed as part of every Incredible PBX for Wazo build.

Unless both of your servers reside on the same local area network, you will need to deploy a NeoRouter server somewhere, but NOT on your Wazo Master since the NeoRouter server itself would become a single point of failure should it die along with your primary server. The Slave server would be a great choice. We covered the NeoRouter Server setup a long time ago in this tutorial, but don’t use the vintage install script. Instead you’ll need to deploy a current version of the Free NeoRouter Server that matches your server platform now that we support operating systems other than CentOS. Incidentally, all of the supported Cloud platforms that we’ve documented for Wazo also support NeoRouter.

We’ve made NeoRouter Server setup easy with this script which works with CentOS/SL, Ubuntu, Debian, and Raspbian. The actual setup steps covered in our original tutorial still are the same.

cd /root
wget http://incrediblepbx.com/install-neorouter-server
chmod +x install-neorouter-server
./install-neorouter-server

After you have your Free NeoRouter Server in place, the next step is to run nrclientcmd on each Wazo server and login to your NeoRouter Server with your credentials. The NeoRouter Server will assign a private IP address to each machine on the NeoRouter VPN. The addresses will be in the range 10.0.0.1 to 10.0.0.255. We’ll use these assigned addresses when setting up the Master and Slave Wazo HA servers.

High Availability Prerequisites with Wazo

In the Incredible PBX for Wazo context, the prerequisites list for your two HA servers is a short one. (1) You need two functioning Incredible PBX for Wazo machines on the same local area network. (2) Both the Master and Slave must be running the same version of Wazo. (3) All trunk registration timeouts (expiry) must be less than 300 seconds. (4) The Slave server must have no phone provisioning plugins installed.

For those using Google Voice trunks with OAuth in conjunction with Incredible PBX for Wazo, keep in mind that this is NOT an integral component of Wazo so it technically is not supported. However, you can easily make it work by configuring any desired Google Voice trunks on BOTH the Master and Slave machines using add-gvtrunk before enabling High Availability. Then the Google Voice trunks will continue to work even after a failover to Slave.

High Availability Limitations with Wazo

When the Master node fails, some features are not available on the Slave:

  • Call history / call records are not recorded.
  • Voicemail messages saved on the Master node are not available.
  • Custom voicemail greetings recorded on the Master node are not available.
  • Phone provisioning is disabled, i.e. a phone will always keep the same configuration, even after restarting it.
  • Phone remote directory is not accessible because the provisioned IP address points to the Master.

Configuring Your Servers for High Availability

Like most Wazo tasks, setting up High Availability on your Master and Slave servers is a 5-minute process. Begin by configuring HA in the Web interface: Configuration ‣ Management ‣ High Availability. (1) Configure the first server as Master with the Remote Address of the Slave. (2) Login to the Linux CLI of Master as root and restart Wazo: xivo-service restart. (3) For the second machine, configure the server as Slave with the Remote Address of the Master.

Next, return to the Linux CLI of Master while still logged in as root. (1) Set up file synchronization by running this script: xivo-sync -i. (2) Start configuration synchronization by running: xivo-master-slave-db-replication 192.168.1.2 using the actual IP address of your Slave. (3) Finally, synchronize the two servers by running xivo-sync on Master. Done! Isn’t it nice saving $3,000 for 5 minutes work using open source software? 🙂

If you love the nitty gritty details, you can read up on Wazo HA in their excellent documentation.

Here’s what pbxstatus will show on Master and Slave while both servers are operational:

And here’s what happens when you halt Master. Within a minute or two, your designated Slave server will come to life:

Choosing Compatible Phones for High Availability

That’s only half the story, of course. Now that you have HA up and running, the remaining trick is that you want your phones to continue to work when things switch over to Slave. To accomplish this, you’ll need to use SIP phones that are compatible with HA technology. Some are, and many are not. Wazo has made it easy for you by publishing a compatibility list. Their documentation includes Officially Supported Devices as well as Community Supported Devices. HINT: Snom, Yealink, and Aastra 6700i and 9000i series phones are your safest bets.1 Here’s what a SIP extension setup would look like on Yealink’s popular T46G:

Deploying Multi-Tenant Technology with Wazo

If you’re new to MT technology, the idea here is to provide separate extensions and trunks for use by different departments within an organization. The reasons should be obvious. These departments have separate budgets and separate clientele, and you probably don’t want the public calling a central number in order to reach everyone in an organization. And the organization wants to identify costs and log calls associated with its various departments.

Wazo handles MT using Entities. When you set up Incredible PBX for Wazo, it automatically created a single Entity named Incredible PBX. You can create additional ones and name them anything you like in the Wazo Web interface: Configuration ‣ Management ‣ Entities.

Next, create Contexts to support your new Entity. Mimic the existing contexts in IPX ‣ IPX Configuration ‣ Contexts and provide unique names for each of them. Be sure you associate each of the new contexts with the new entity you created. Then set up users, lines, trunks, and call routing for the new entity in the same way you did it for the original IncrediblePBX entity. Take a look at Amy Grant’s setup with Google Voice on the PIAF Forum for additional tips. Simple and it’s FREE!

Originally published: Monday, September 5, 2016  Updated: Saturday, January 28, 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…

  1. HA failover even works great using $29 UTP-E62 if you can find one. []

Raspberry Pi One-Minute Wonder: A Turnkey and Truly Incredible PBX for XiVO


Hard to believe it’s been 4½ years since the introduction of the original Raspberry Pi®. We love half-birthdays, and we’ve got a blockbuster gift for you today as we celebrate the fact that almost 10 million RasPi’s have been shipped. Yes, our love affair with the Raspberry Pi lives on. The sensational Raspberry Pi 3 sports a 1.2GHz 64-bit quad-core ARM Cortex-A53 CPU with ten times the performance of the original Raspberry Pi. Of particular interest to the VoIP community will be the RasPi 3’s integrated 802.11n wireless LAN and Bluetooth 4.1 hardware. And, of course, the RasPi 3 retains its compatibility with the Raspberry Pi 1 and 2. Did we mention it’s still just $35? Because we like to celebrate half birthdays, too, we’re pleased to introduce a brand new Incredible PBX™ for XiVO image for the Raspberry Pi 3 featuring Raspbian 8, the latest release of Asterisk® 13, and XiVO. This one installs in under a minute. And, yes, it’s still FREE with pure open source GPL code.

Special Thanks. First things first. We want to extend our extra special thanks to Iris-Network for their awesome Raspivo – XiVO build. Without it and their repositories, none of this would have been possible.

Raspberry Pi 3 Performance. Gone are the days of worrying about Raspberry Pi performance. Both the user interface and call quality now match what you’d expect to find on a $300-$500 VoIP server. For best results, we recommend 32GB Class 10 microSD cards which now are plentiful at the $10 price point.1

Raspberry Pi 3 Shopping List. Before you can install Incredible PBX for XiVO, you’ll need a compatible Raspberry Pi 3 platform. Here’s the short list that, when coupled with the Incredible PBX image, turns today’s adventure into kid’s play:

  • $35* Raspberry Pi 3 from MCM or Newark or Amazon
  • $10 Power Adapter (2.5 amps minimum!)
  • $10 32GB microSDHC Class 10 card (Don’t use SanDisk Ultra!)
  • £12.95 Pibow 3 case or $7.50 Official RasPi 3 case
  • About That Asterisk. We write about Asterisk® regularly, but the asterisk we’re talking about is the one accompanying the $35* price tag for the Raspberry Pi 3. Yes, that’s the advertised price. And, no, if you want one quickly, you may pay a bit more. Right now you can snag one on Amazon for $35.99 with two-day Prime shipping. We’re assuming you already own a USB keyboard and an HDMI-compatible monitor. If so, today’s going rate for all of our recommended pieces is under $65, not bad for a fully-equipped, quad-core computer. Did we mention that Incredible PBX for XiVO is FREE with NoGotchas!

    Incredible PBX Feature Set. Where to begin? Let’s start with the Alphabet Stew: IAX, SIP, SMS, FAX, SRTP, and OAuth functionality. Voice Recognition and Text-to-Speech VoIP application support using Festival and Google. Free calling with Google Voice, Simonics SIP gateway, or RingPlus cellular service. And all of your Nerd Vittles favorites: AsteriDex, Click-to-Dial, News, Weather, Reminders, and even an Alarm Clock. Plus hundreds of features that typically are found in commercial PBXs: Conferencing, IVRs and AutoAttendants, Simultaneous Ringing on your Smartphone, Email Delivery of Voicemail, Voicemail Blasting, Automatic Backups, High Availability Support, Automatic Phone Setups, and much more…

    Incredible PBX Network Security Model. Most phone calls cost money. Unlike many of the other "free" VoIP solutions, our most important criteria for VoIP is rock-solid security. If your free server ends up costing you thousands of dollars in phone bills due to fraud, it isn’t free at all. Once you plug in that network cable, you’ve painted a bullseye on your checkbook.

    No single network security system can protect you against zero-day vulnerabilities that no one has ever seen. Deploying multiple layers of security is not only smart, it’s essential with today’s Internet topology. It works much like the Bundle of Sticks from Aesop’s Fables. The more sticks there are in your bundle, the more difficult it is to break them apart. If a vulnerability suddenly appears in the Linux kernel, or in Asterisk, or in your web server, or in your favorite web GUI, you can continue to sleep well knowing that other layers of security have your back. No one else in the telecommunications industry has anything close. You can’t hack what you can’t see, and the Incredible PBX automatically configures a WhiteList as part of the one-minute setup. And it’s all open source GPL code that you can share with anybody and everybody unlike the so-called "freeware" products. Freeware with Asterisks is anything but free!

    Do your part and do your homework. Comparison shop as if your phone bill matters! 😉 Incredible PBX provides:

    1. Preconfigured IPtables Linux Firewall
    2. Preconfigured Travelin’ Man 3 WhiteLists
    3. Randomized Port Knocker for Remote Access
    4. Fail2Ban Log Monitoring for SSH, Apache, Asterisk
    5. Randomized Ultra-Secure Passwords
    6. Automatic Update Utility for Security & Bug Fixes
    7. Asterisk Manager Lockdown to localhost
    8. Security Alerts via the PIAF Forum

    Incredible PBX for XiVO Installation & Setup Tutorial

    Here’s everything need to know about installation and setup of Incredible PBX for XiVO. "Automatic" means you just watch.

    1. Download and unzip Incredible PBX for XiVO image from SourceForge (includes GV OAuth support)
    2. Transfer Incredible PBX image to microSD card
    3. Boot Raspberry Pi from new microSD card
    4. Login to RasPi console as root:password to initialize your server (Automatic) and expand image to match SD card
    5. Reboot after writing down your server IP address (Automatic)
    6. Login via SSH as root:password to set up passwords (You Pick ’em) & configure firewall (Automatic)
    7. Enjoy!

    Running Incredible PBX for XiVO on the Raspberry Pi

    The standard XiVO boot procedure will begin once you insert your microSD card into the Raspberry Pi 3 and apply power. Within a short time, you’ll get the familiar Linux login prompt. Login as root with a password of password.

    Once you log in, a startup script will briefly configure a few things and then advise you that it’s time to reboot. Write down the IP address provided because for Phase 2 of the setup, we need to use SSH or Putty on the desktop that you will actually be using to manage your server. The reason for this is that Incredible PBX automatically creates a whitelist of IP addresses that the firewall will allow to access your server. If the IP address isn’t in your whitelist, you may lock yourself out except from the RasPi’s console window.

    Once the console window shows that your server has rebooted by displaying the Linux login prompt, switch to SSH or Putty and login as root using the IP address you wrote down. You’ll then be prompted to change your root password for Linux as well as your root password for XiVO GUI access using a web browser. You’ll also need to set a PIN that will be used to authorize access to extension 123 to schedule Telephone Reminders on your server. This completes the configuration. You’ll get a final screen showing the credentials for the preconfigured extension 701 as well as a reminder that your PortKnocker credentials are stored in /root/knock.FAQ in the event you ever lock yourself out of your machine. It’s a good idea to leave this screen displayed while you install and configure a softphone since you can cut-and-paste your extension 701 credentials without having to type anything.

    Once you complete the SIP softphone setup below, you can return to the SSH window and press ENTER to finish the install. The Incredible PBX Automatic Update Utility will run, and then you will be presented with the pbxstatus display. You can access the Asterisk CLI by typing: asterisk -rvvvvvvvvvv. Exit from the CLI by typing quit. As mentioned previously, always shut down your server gracefully by typing halt. When prompted for the hostname, type xivo. Once the shutdown procedure finishes, it’s safe to disconnect the power cord from your Raspberry Pi.

    Beginning with the September 1 release, many of the log files have been disabled to help prolong the life of microSD cards since XiVO tends to be very chatty. If you are running an earlier release, you can follow this tutorial to disable most logging on your Raspberry Pi.

    Enabling WiFi on the Raspberry Pi 3

    With the Raspberry Pi 3, wi-fi hardware is included. The next step is configuring it to connect to your WiFi router. Simply open /etc/wpa_supplicant/wpa_supplicant.conf with nano and (1) edit the SSID name and password fields to authorize access to your local, password-protected WiFi router as well as any open WiFi network. (2) Also update the country code for your WiFi region, e.g. country=US. Then (3) save your changes: Ctrl-X, Y, then press ENTER.

    network={
     ssid="YourSSID"
     psk="YourSSIDpassword"
     key_mgmt=WPA-PSK
     scan_ssid=1
     priority=5
    }
    
    network={
     key_mgmt=NONE
     priority=1
    }
    

    Next, enable automatic startup of the wlan0 network interface:

    sed -i 's|#allow-hotplug wlan0|allow-hotplug wlan0|' /etc/network/interfaces
    

    Finally, stop and restart the wlan0 interface, count to 15, and check pbxstatus to decipher the added private IP address for your WiFi connection:

    ifdown wlan0
    ifup wlan0
    pbxstatus
    

    If you want to run your Raspberry Pi exclusively off the WiFi connection going forward, simply unplug the network cable from your RasPi and reboot your server.

    Choosing a SIP Softphone for Incredible PBX for XiVO

    Softphones tend to be a matter of taste for most folks so we’ll keep our suggestions to a minimum. On the Windows platform, it’s hard to go wrong with X-Lite. It works out of the box by simply plugging in the IP address of your server and your SIP username and password. It also happens to be free. The only downside is that X-Lite has a nasty habit of embedding time bombs in their free software so you may have to reinstall it from time to time. If you know what you’re doing Zoiper is another alternative but be advised that it doesn’t work out of the box on servers behind NAT-based routers.

    On the Mac platform, our favorite free softphone is Telephone. It’s a barebones SIP client that just works. As with X-Lite, you plug in your server’s IP address and SIP credentials, and you’re in business.

    On the Linux or Solaris platforms, we assume that you know what you’re doing and that you are perfectly capable of choosing and installing a SIP phone that meets your requirements.

    Incredible PBX Application Quick Start Guide

    We’ve finished the basic Incredible PBX for XiVO setup. You now have a functioning PBX with dozens of applications for Asterisk that work out of the box. It’s probably a good idea to spend a little time getting acquainted with Incredible PBX for XiVO before you add trunks to communicate with the outside world.

    Here’s a handy cheat sheet for some of the Incredible PBX applications that have been installed or are available as add-ons. There’s also a link for more information. This remains a work-in-progress so expect more applications in coming weeks.

    How To Make Easily Compressed Backups of Incredible PBX

    MicroSD cards WILL wear out especially on XiVO servers with lots of activity. So it’s important to make regular backups of your media so you don’t get surprised when things come unglued down the road. After considerable discussion on the PIAF Forum, here’s the collective wisdom.

    You’ll need another machine (such as a Mac or Linux box) on which to plug in the microSD card in order to make a backup image of it since you can’t back up a card that is actually providing the live platform for your PBX. The recommended methodology goes like this. Before shutting down your PBX and removing the microSD card to make the backup, convert all of the unused space on the card to zeros so that the unused space can be easily compressed when you create the backup image. You do this by issuing the following command after logging into the Linux CLI as root on your RasPi 3. Be sure to do it during a period of inactivity on your PBX as it is processor intensive. Then halt the machine and remove the microSD card.

    xivo-service stop
    cat /dev/zero > wipe.it ; rm wipe.it
    halt
    

    Insert the card into an SD card slot on the machine you will use to make the backup image and issue the following commands after deciphering the correct device name for your card (/dev/disk4 in this example) using the df utility:

    sudo df -h
    sudo dd bs=1m if=/dev/disk4 | gzip -c > incrediblepbx-xivo.img.gz
    sudo sync
    sudo diskutil eject /dev/disk4s1
    echo "It's safe to remove the microSD card now."
    

    Now return the microSD card to your Raspberry Pi 3 and boot. Store your backups in a safe place!

    Configuring Trunks and Routes with Incredible PBX for XiVO

    The next step in your XiVO adventure is connecting your PBX to the outside world so that you can make and receive phone calls from anywhere in the world. For this you’ll need one or more trunks. Unlike the Ma Bell world, there’s no reason to put all your eggs in one basket. You can use one or more trunk providers for incoming calls with separate phone numbers for each. And you can use one or more trunk providers for outgoing calls and save money on calls to certain countries by choosing the best provider for where you want to call. And, of course, if you live in the United States, you can set up one or more Google Voice trunks and make calls to the U.S. and Canada for free. We’ve written a number of tutorials to make it easy to set up these trunks.

    To get started, point a web browser to the IP address of your PBX. Login as root with the XiVO GUI password you set up above. If you ever forget your password, you can run /root/admin-pw-change to reconfigure it.

    XIVO Trunk Implementation Tutorials

    Once you’ve added one or more trunks, you’ll need to tell XiVO how to route outgoing and incoming calls. Here are our step-by-step tutorials on setting up Outbound Calling Routes and Incoming Call Routes:

    XIVO Call Routing Tutorials

    Enabling Bluetooth & Proximity Detection on the Raspberry Pi


    Where To Go Next with Incredible PBX for XiVO

    Now you’re ready to explore. We recommend you pick up here in our Incredible PBX for XiVO tutorial. And be sure to check out the Last Minute Fixes that didn’t make it into the current build. Enjoy the ride!

    Originally published: Monday, August 29, 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. Many of our purchase links refer users to Amazon when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from Amazon to help cover the costs of our blog. We never recommend particular products solely to generate Amazon commissions. However, when pricing is comparable or availability is favorable, we support Amazon because Amazon supports us. []
    2. Vitelity is a platinum sponsor of Nerd Vittles, and they also happen to be the best in the business. You’ll find a discount coupon to get a great deal on a DID and 4-channel trunk toward the end of this article. []

    VirtualBox Magic: A Turnkey PBX in 5 Minutes Flat with XiVO

    We’ve sung the praises of VirtualBox for many years because it provides a wonderful platform for experimentation as well as production-ready systems using almost any hardware and any operating system. Versions of VirtualBox are available for Windows PCs, Macs, Linux desktops, and even Solaris machines. And, once you have VirtualBox in place, you can load gigabyte-sized turnkey virtual machines in under a minute. It literally transforms complex computer setups into child’s play.

    We’ve received dozens of emails about XiVO, and most of them go something like this:

    I’d love to experiment with XiVO as an Asterisk® platform, but I worry that the environment is just too different and the learning curve too steep. I just wish there were a simple way to get started so that I could learn the basics.

    Today, your prayers have been answered. You don’t have to buy any hardware. You can use the desktop computer you already have. We’ve taken the Incredible PBX for XiVO tutorial and turned it into a turnkey virtual machine for VirtualBox. You can load it in under a minute and be ready to go. It’s got all of the Incredible PBX bells and whistles, and an extension is already configured so that you can hit the ground running. Just install VirtualBox. Next, install Incredible PBX for XiVO. Install your favorite SIP phone. Plug in the SIP credentials provided. And you’re done in a few minutes. To make outgoing calls, you can add a SIP trunk using one of the numerous SIP provider tutorials we’ve provided. Or, if you live in the United States, you can add a Google Voice trunk in a couple minutes and make free calls in the U.S. and Canada. Let’s get started!

    Installing Oracle VM VirtualBox

    Oracle’s virtual machine platform inherited from Sun is amazing. It’s not only free, but it’s pure GPL2 code. VirtualBox gives you a virtual machine platform that runs on top of any desktop operating system. In terms of limitations, we haven’t found any. We even tested this on an Atom-based Windows 7 machine with 2GB of RAM, and it worked without a hiccup. So step #1 today is to download one or more of the VirtualBox installers from VirtualBox.org or Oracle.com. Our recommendation is to put all of the 100MB installers on a 4GB thumb drive.1 Then you’ll have everything in one place whenever and wherever you happen to need it. Once you’ve downloaded the software, simply install it onto your favorite desktop machine. Accept all of the default settings, and you’ll be good to go. For more details, here’s a link to the Oracle VM VirtualBox User Manual.

    Downloading & Installing Incredible PBX for XiVO Virtual Machine

    To begin, download Incredible PBX for XiVO .ova image (1.0 GB) to the computer on which you installed VirtualBox.

    When the download completes, double-click on the .ova file you downloaded to load it into VirtualBox. When prompted, be sure to check the Reinitialize the Mac address of all network cards box, agree to the license agreement, and then click the Import button. Once the import is finished, you’ll see a new (1) Incredible PBX for XiVO virtual machine in the VM List of the VirtualBox Manager Window. We need to make a couple of one-time adjustments to the Incredible PBX for XiVO configuration to account for differences in sound and network cards on different host machines.

    (1) Click once on the Incredible PBX for XiVO virtual machine in the VM List. Then (2) click the Settings button. In the Audio tab, check the Enable Audio option and choose your sound card. In the Network tab for Adapter 1, check the Enable Network Adapter option. From the Attached to pull-down menu, choose Bridged Adapter. Then select your network card from the Name list. Then click OK. That’s all the configuration that is necessary for your Incredible PBX for XiVO.

    Running Incredible PBX for XiVO in VirtualBox

    Once you’ve imported and configured the Incredible PBX for XiVO Virtual Machine, you’re ready to go. Highlight Incredible PBX for XiVO virtual machine in the VM List on the VirtualBox Manager Window and click the Start button. The standard XiVO boot procedure will begin and, within a short time, you’ll get the familiar Linux login prompt. During the bootstrap procedure, you’ll see a couple of dialogue boxes pop up that explain the keystrokes to move back and forth between your host operating system desktop and your virtual machine. Remember, you still have full access to your desktop computer. Incredible PBX for XiVO is merely running as a task in a VirtualBox window. Always gracefully halt Incredible PBX just as you would on a dedicated computer.

    Here’s what you need to know. To work in the Incredible PBX for XiVO virtual machine, just left-click your mouse while it is positioned inside the VM window. To return to your host operating system desktop, press the right Option key on Windows machines or the left Command key on any Mac. For other operating systems, read the dialogue boxes for instructions on moving around. To access the Linux CLI, login as root with the default password: password.

    Once you log into your virtual machine, a startup script will briefly configure a few things and then advise you that it’s time to reboot. Write down the IP address provided because for Phase 2 of the setup, we need to use SSH or Putty on the desktop that you will actually be using to manage your server. The reason for this is that Incredible PBX automatically creates a whitelist of IP addresses that the firewall will allow to access your server. If the IP address isn’t in your whitelist, you may lock yourself out except from the VirtualBox console window.

    Once the VirtualBox console window shows that your server has rebooted by displaying the Linux login prompt, switch to SHH or Putty and login as root using the IP address you wrote down. You’ll then be prompted to change your root password for Linux as well as your root password for XiVO GUI access using a web browser. You’ll also need to set a PIN that will be used to authorize access to extension 123 to schedule Telephone Reminders on your server. This completes the configuration. You’ll get a final screen showing the credentials for the preconfigured extension 701 as well as a reminder that your PortKnocker credentials are stored in /root/knock.FAQ in the event you ever lock yourself out of your machine. It’s a good idea to leave this screen displayed while you install and configure a softphone since you can cut-and-paste your extension 701 credentials without having to type anything.

    Once you complete the SIP softphone setup below, you can return to the SSH window and press ENTER to finish the install. The Incredible PBX Automatic Update Utility will run, and then you will be presented with the pbxstatus display. You can access the Asterisk CLI by typing: asterisk -rvvvvvvvvvv. Exit from the CLI by typing quit. As mentioned previously, always shut down your server gracefully by typing halt. When prompted for the hostname, type xivo. Once the shutdown procedure finishes, it’s save to turn off your virtual machine.

    Choosing a SIP Softphone for Incredible PBX for XiVO

    Softphones tend to be a matter of taste for most folks so we’ll keep our suggestions to a minimum. On the Windows platform, it’s hard to go wrong with X-Lite. It works out of the box by simply plugging in the IP address of your server and your SIP username and password. It also happens to be free. The only downside is that X-Lite has a nasty habit of embedding time bombs in their free software so you may have to reinstall it from time to time. If you know what you’re doing Zoiper is another alternative but be advised that it doesn’t work out of the box on servers behind NAT-based routers.

    On the Mac platform, our favorite free softphone is Telephone. It’s a barebones SIP client that just works. As with X-Lite, you plug in your server’s IP address and SIP credentials, and you’re in business.

    On the Linux or Solaris platforms, we assume that you know what you’re doing and that you are perfectly capable of choosing and installing a SIP phone that meets your requirements.

    Incredible PBX Application Quick Start Guide

    We’ve finished the basic Incredible PBX for XiVO setup. You now have a functioning PBX with dozens of applications for Asterisk that work out of the box. It’s probably a good idea to spend a little time getting acquainted with Incredible PBX for XiVO before you add trunks to communicate with the outside world.

    Here’s a handy cheat sheet for some of the Incredible PBX applications that have been installed or are available as add-ons. There’s also a link for more information. This remains a work-in-progress so expect more applications in coming weeks.

    Configuring Trunks and Routes with Incredible PBX for XiVO

    The next step in your XiVO adventure is connecting your PBX to the outside world so that you can make and receive phone calls from anywhere in the world. For this you’ll need one or more trunks. Unlike the Ma Bell world, there’s no reason to put all your eggs in one basket. You can use one or more trunk providers for incoming calls with separate phone numbers for each. And you can use one or more trunk providers for outgoing calls and save money on calls to certain countries by choosing the best provider for where you want to call. And, of course, if you live in the United States, you can set up one or more Google Voice trunks and make calls to the U.S. and Canada for free. We’ve written a number of tutorials to make it easy to set up these trunks.

    To get started, point a web browser to the IP address of your PBX. Login as root with the XiVO GUI password you set up above. If you ever forget your password, you can run /root/admin-pw-change to reconfigure it.

    XIVO Trunk Implementation Tutorials

    Once you’ve added one or more trunks, you’ll need to tell XiVO how to route outgoing and incoming calls. Here are our step-by-step tutorials on setting up Outbound Calling Routes and Incoming Call Routes:

    XIVO Call Routing Tutorials

    Now you’re ready to explore. We recommend you pick up here in our Incredible PBX for XiVO tutorial. And be sure to check out the Last Minute Fixes that didn’t make it into the current build. Enjoy the ride!

    Originally published: Monday, August 22, 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. Many of our purchase links refer users to Amazon when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from Amazon to help cover the costs of our blog. We never recommend particular products solely to generate Amazon commissions. However, when pricing is comparable or availability is favorable, we support Amazon because Amazon supports us. []
    2. Vitelity is a platinum sponsor of Nerd Vittles, and they also happen to be the best in the business. You’ll find a discount coupon to get a great deal on a DID and 4-channel trunk toward the end of this article. []

    Google Voice with OAuth 2 Comes to Incredible PBX for XiVO


    Since we began our XiVO adventure a couple months ago, the most requested feature has been direct support for Google Voice. For those in the United States, it remains the cheapest VoIP solution on the planet (when it works) with unlimited free calls throughout the U.S. and Canada. While we’ve had Google Voice functionality in XiVO through the Simonics SIP to Google Voice Gateway since the outset, there still were some who preferred to keep their credentials and tokens to themselves. And then there were those that found the $4.99 per line Simonics setup fee too rich for their blood.

    To celebrate the new school year, today we’re pleased to provide a new tutorial and script that bolts Google Voice with OAuth authentication onto Incredible PBX for XiVO. Our extra special thanks goes to Sylvain Boily, the father of XiVO, for his selfless work in bringing this to fruition in less than a day. That tells you just how adaptable the XiVO platform really is. We’ve simply added a little window dressing to ease the pain for those just getting started with XiVO and Incredible PBX.

    Overview. If you’re new to Google Voice, here’s how the installation scenario goes. First, you set up a Gmail account at gmail.com. Next, you create a Google Voice account. Then, you configure Google Voice for use with Asterisk®. Next, you obtain your Google Voice OAuth 2 Refresh Token which becomes your password to use in configuring Google Voice on the XiVO platform. Next, using SSH or Putty, you log into your XiVO server as root and download and run the installation script to get your Google Voice credentials set up in XiVO. Finally, you log into the XiVO GUI with a browser and set up a custom trunk as well as an outgoing and incoming route for Google Voice calls. To add more Google Voice trunks, you simply repeat the drill. You now should have a perfectly functioning, free VoIP platform compliments of Google and Sylvain Boily and his development team. Let’s get started.

    Configuring Google Voice for Use with XiVO

    If you’re one of the five people on Earth that does not yet have a Gmail account, start there. Once you’ve set up your Gmail account and logged in, open a new browser tab to access the Google Voice site. Accept the Google Terms and Privacy Policy. Then choose a new Phone Number in your favorite area code. NOTE: Before Google will assign you a number, you must enter an existing U.S. phone number to verify your identity and location as well as to use for initially forwarding calls. Once your account is set up, you will get an email asking that you verify your email address. Once you’ve done that, you’ll be prompted to login to your Google Voice account again. When you do so, you’ll be prompted to Install the Hangouts Dialer app to make VoIP calls from Android. Do NOT install the dialer, or you may break the ability to use your Google Voice number with Asterisk. Instead, click X to close the dialog box.

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

    Next, click on the Calls tab. Make sure your settings match these:

    • Call ScreeningOFF
    • Call PresentationOFF
    • Caller ID (In)Display Caller’s Number
    • Caller ID (Out)Don’t Change Anything
    • Do Not DisturbOFF
    • Call Options (Enable Recording)OFF
    • Global Spam FilteringON

    Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Then click Save Settings. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.

    One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in any provider passwords, or nothing will work!

    Now it’s time to obtain your OAuth 2 credentials. Even though it’s a bit more work on the front end, the good news is you won’t have to worry about your Google Voice trunks failing when Google phases out plain-text passwords. The other good news is you won’t be passing your plain-text Google Voice credentials across the Internet for everyone in the world to see.

    Obtaining Your Google Voice OAuth 2 Credentials for XiVO

    While you’re still logged into your Google Voice account, you need to obtain a refresh_token which is what you’ll use instead of a password when setting up your Google Voice account with XiVO. Here’s how.

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

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

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

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

    466295438629-prpknsovs0b8gjfcrs0sn04s9hgn8j3d.apps.googleusercontent.com
    

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

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

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

    5. Click Step 2: Exchange authorization code for tokens

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

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

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

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

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

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

    Downloading and Installing Google Voice with OAuth 2 for XiVO

    Now it’s time to reconfigure XiVO to use Google Voice with OAuth 2. Before you begin, write down your 10-digit Google Voice phone number, your Google account name without @gmail.com, and your Refresh Token from the previous step.

    Log into your server as root using SSH or Putty. Then execute the following commands to kick off the install:

    cd /root
    apt-get update
    apt-get -y install build-essential libssl-dev
    wget http://incrediblepbx.com/gvoauth-xivo.tar.gz
    tar zxvf gvoauth-xivo.tar.gz
    rm -f gvoauth-xivo.tar.gz
    ./add-gvtrunk
    

    Plug in your Google Voice phone number and credentials when prompted. Then check your work carefully. When the install finishes, fire up your favorite browser to finish the setup using the settings that were provided.

    Configuring XiVO for Google Voice OAuth

    From a browser pointed to the IP address of your server, log in to XiVO as root with your GUI password.

    Choose Services.IPBX.Trunk Management.Customized. Click on + Add to create a new custom trunk. Configure the trunk using the settings provided by the installer and click the Save button. The Trunk Name will be your actual gmailname (without @gmail.com). Interface will be Motif/gmailname (using your actual Gmail name). Interface suffix will be @voice.google.com. And the Context will be Outcalls (to-extern).

    Next, choose Services.IPBX.Call Management.Incoming Calls. Click on + Add to create a new inbound route for your Google Voice DID. This is where you tell XiVO how to route calls placed to your Google Voice number. For your DID, enter your 10-digit Google Voice number. For the Context, choose Incalls (from-extern). Then choose from the pick lists to select a Destination and Redirect option for the incoming calls. It could be an extension, a ring group, a conference room, or an IVR. Click Save when you’re finished.

    Finally, choose Services.IPBX.Call Management.Outgoing Calls. Click on + Add to create a new outbound route. Under the General tab, configure the route as shown below using a Name of out_gmailname (using your actual Gmail name). The Context should be Outcalls (to-extern). The Preprocess subroutine should be subr-gv-outcall. And the desired Custom Trunk should be dragged left to the selected column:

    Under the Exten tab, add the desired 10-digit Exten string and trim off any prefix using Stripnum. Then click the Save button.

    For example, for a first trunk, you might choose NXXNXXXXXX as the Exten with a Stripnum of 0. This would tell XiVO to route all 10-digit calls to this Custom GV Trunk. For a second Google Voice trunk, you might choose 9NXXNXXXXXX with a Stripnum of 1. This would tell XiVO to route 11-digit calls with a 9 prefix to this Custom Trunk AND to strip off the first digit (9) before sending the 10-digit call to Google Voice.

    Taking Google Voice for a Test Drive with XiVO

    That completes the Google Voice setup. You now should be able to place a call using your Google Voice trunk by dialing any 10-digit number. And calls placed to your Google Voice number should ring at the inbound destination you chose above.

    If you have additional Google Voice trunks, simply run /root/add-gvtrunk again and insert the new credentials.

    Should you ever need to delete a Google Voice account from your server, just run /root/del-gvtrunk with the name of the Google Voice trunk to delete. Enjoy your free phone service!

    Originally published: Tuesday, August 16, 2016   Updated: Thursday, August 18, 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…

    Take the XiVO Plunge: 4 Months of Free Cloud Hosting


    Nobody has to tell us how painful change can be. We oversaw the deployment of over 30,000 IBM PCs only to switch horses and become a dedicated Mac lover. And we’ve invested almost 10 years in another Asterisk® GUI only to be disappointed by the direction of that project. That led to our New Year’s Resolution to find a better mousetrap for unified communications open source development. And, boy, did we find one. So here’s the deal. You either believe in the open source community and want to foster free and open development of software, or you don’t. And, if you don’t, that’s perfectly fine. There are lots of commercial PBX alternatives including the terrific 3CX products from our platinum sponsor. But don’t wrap yourself in the open source flag, brag about free and freedom, and then market a product that is none of the above. If your distro’s license agreement prohibits redistribution thereby discouraging sharing which is the lynchpin of the GPL, then the product has little if anything to do with free and freedom.

    The good news is we’ve now found an awesome alternative that is pure open source code with an actual GPL3 license. So come join the party and lend a hand with your suggestions and/or your code contributions. We’ll put your name in bright lights, and the open source community will be forever in your debt. Our challenge is to get you as excited about XiVO as we are. There’s nothing with VoIP and Unified Communications that you can’t do better, cheaper, and faster using XiVO. And XiVO’s Asterisk RealTime implementation has no competition, period. Instead of lengthy delays to process changes, rewrite Asterisk config files, and reload the entire Asterisk dial plan, Asterisk RealTime brings instantaneous configuration updates.

    We can think of no better way to introduce you to this terrific platform than offering up a free cloud platform until 2017 to let you kick the tires. It won’t impact your production servers while letting you explore the possibilities offered by a state-of-the-art Asterisk 13 platform with no equal. Believe me. We know every wart and pimple in the old GUI platform, and you won’t have to wrestle with any of the traditional problems that we all assumed were native to Asterisk. Guess what? They weren’t. No, your server won’t blow up when you add a new module. No, Asterisk won’t refuse to start because you chose to upgrade an existing component. No, you won’t be Nickle and Dimed into buying critical platform enhancements. And, no, you won’t be charged hundreds of dollars for "support" only to be told that you need to switch to a more proprietary platform. Yes, the XiVO development team releases seamless upgrades every three weeks at no cost. Yes, uncrippled endpoint provisioning for dozens of phones is provided in XiVO at no cost. Yes, powerful call center and High Availability technology is included at no cost. And, yes, backups of your server are made every night for free.

    There’s more good news. VULTR is a relatively new cloud provider that now hosts virtual machines in over a dozen cities around the world. For new subscribers, they are offering a $20 credit when you sign up using our referral link. And, yes, your registration provides a few shekels to Nerd Vittles to keep the lights on. The great news is that $20 buys you a full four months of XiVO cloud hosting service, and you won’t find a better do-it-yourself platform at any price, let alone free.

    Building the Debian 8 Platform at Vultr for XiVO

    The first step in your XiVO adventure is to sign up for a Vultr account with your $20 credit using the Nerd Vittles referral link. Once you’ve done that, it’s time to build your Debian 8 virtual machine to host XiVO in the Cloud. (1) Choose your favorite city to host your server, (2) pick the Debian 8 64-bit platform, and (3) choose the $5/month server size.

    IMPORTANT: Leave the Server Hostname & Label blank!

    Once your virtual machine is up and running, log in with SSH or Putty using the root password provided. Do NOT install XiVO from the console, or the firewall will lock you out of your own machine! Change your root password immediately: passwd.

    Next, set up a swap file on your virtual machine, or the XiVO install will fail on the $5 platform:

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

    Installing Incredible PBX for XiVO in the Vultr Cloud

    While still logged into your server as root using SSH/Putty, issue the following commands to kick off the install:

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

    The initial setup brings your Debian 8 server up to current specs, and then the virtual machine will reboot. After rebooting, log into your server again as root with your new root password. Issue the following command to complete the XiVO and Incredible PBX installation and configuration:

    ./IncrediblePBX13-XiVO.sh
    

    You’ll be prompted to set your time zone, passwords, and choose the optional features of Incredible PBX you wish to install. We strongly recommend you install ALL of the Incredible PBX feature set. Many cannot be added later.

    Verify that the XiVO install completed successfully when prompted. Then verify that the XiVO initial configuration completed successfully by once again pressing ENTER. The firewall and Incredible PBX install will then proceed without further prompting. Total setup time: under 10 minutes.

    There still are some setup steps required, and these are performed within the XiVO GUI using a web browser. For step-by-step instructions on the Incredible PBX Initial Configuration Procedure, click here. Enjoy your adventure!

    Originally published: Monday, July 25, 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…

    The Definitive Quick Start Guide: Introducing Incredible PBX for XiVO



    Today we kick off a new Asterisk® adventure with the introduction of Incredible PBX™ for XiVO®. This pure GPL implementation of Asterisk has no strings, no gotchas, no hidden agenda, and no primadonnas. It’s open source code with no prohibitions on redistribution. The XiVO developers actively participate in the XiVO and PBX in a Flash™ communities and actually listen to constructive suggestions to improve their product. Changes happen in days, not years. Today we celebrate the return of true GPL project development and the end of closed-source ISOs and commercial modules with costly annual support contracts. Join us!

    UPDATE: This article has been superseded. For the latest tutorial, go here.

    If you’ve been following Nerd Vittles these past two months, then you already know there is literally nothing in the open source Unified Communications world that you can’t do faster, better, and cheaper with XiVO: automatic backups every night, seamless upgrades every three weeks, uncrippled endpoint provisioning for dozens of phones, powerful call centers, high availability redundant servers, real-time Asterisk technology out of the box, flexible SDK and APIs, and much more.

    XiVO Installation Methodology

    There are two ways to build XiVO servers. You can start with a minimal install of Debian 8 (64-bit), or you can use the 64-bit XiVO ISO. The advantage of the XiVO ISO is that building a system from the ISO gets you BOTH Debian 8 AND the basic XiVO install. However, you can only use the XiVO ISO on platforms that you own, not on virtual machines controlled by somebody else. Stated another way, if you plan to use dedicated hardware or VirtualBox or VMware ESXi, use the XiVO ISO. Otherwise, install a minimal Debian 8 (64-bit) operating system and nothing else on your platform of choice. Now you’re ready to choose your Incredible PBX installer. Install time: about 5-20 minutes depending upon the platform.

    IMPORTANT: When you build your Debian 8 platform on either stand-alone hardware or as a virtual machine, use a fully-qualified domain name for your server’s hostname, e.g. xivo.incrediblepbx.com, NOT xivo. Disaster awaits if you forget this! But, don’t worry. If you do forget, the install will blow up, and you’ll get to start over. But you’ll remember the next time. 😉

    Incredible PBX Feature Set

    If you’ve been sleeping under a rock for the last few years, you may be wondering what the Incredible PBX offering includes. We’ve tried to preserve much of the functionality of prior releases in the XiVO implementation, and there is still more to come. Here’s a quick summary of two dozen features and applications that Incredible PBX offers for XiVO today:

    Recent Additions: Skype Connect, Port Knocker, PPTP VPN, Pico TTS, A La Carte installer, Telephone Alarms.

    The 3 Flavors of Incredible PBX for XiVO

    To kick off our Independence Day celebration, we introduced three new Incredible PBX turnkey installers for XiVO because of the numerous platforms on which XiVO will run. We’ve now combined all three of the original installers into a single script for ease of use.

    For those new to XiVO, there are three steps in getting a XiVO PBX up and running: (1) Debian 8 OS installation, (2) XiVO installation, (3) and XiVO basic configuration (typically using a web browser). The Incredible PBX installer has different tasks based upon how far along in this installation process you happen to be on a particular platform. Our special thanks to Sylvain Boily for his Python wizard to assist us in providing turnkey installs to the greatest extent possible. So here’s the new installer, but you are well advised to actually follow the platform tutorial (below) for your provider because of special quirks that are provider-specific:

    IncrediblePBX13-XiVO.sh – Suitable for Debian 8 (32-bit or 64-bit) minimal platform where XiVO is not installed. Use with Cloud VMs. Also works with Debian 8 (32-bit or 64-bit) platform with XiVO installed but not configured. This is typically the situation if you built your server using the XiVO ISO. And the new installer works with Debian 8 (32-bit and 64-bit) platform with XiVO installed and configured.

    WARNING: Incredible PBX erases and replaces stuff as part of its installation procedure. NEVER install Incredible PBX over the top of an existing production server!

    Incredible PBX Installation Procedure


    We’ve taken the guesswork out of this for a number of platforms by providing detailed tutorials that you can follow:

    Choosing a XiVO Hardware 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. For turnkey cloud servers such as RentPBX, simply choose the VM option that already has Debian 8 and XiVO preinstalled.

    Once you have your platform up and running, simply download and run the Incredible PBX installer:

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


    Incredible PBX Initial Configuration

    Here are the first steps to complete after you have finished your initial XiVO and Incredible PBX installation. Log into the web interface at the IP address of your server using username root and the web password you created during installation.

    All of this initial setup will be completed under the IPBX option of the Services tab as shown below. For each of the categories below, click on the matching section and tab in XiVO’s IPBX toolbar and fill in the properties as indicated.

    UPDATE: The latest Incredible PBX for XiVO installer automatically configures SIP defaults and a dozen SIP trunks for you using XiVO Snapshots if you elect to install all of the Incredible PBX features when you run the installer. If so, you can skip through the next few sections of this tutorial.

    General Settings:SIP Protocol


    WARNING: If your XiVO server is running as a virtual machine behind a hardware-based NAT router and the virtual host also is sitting behind the same router, you may experience failed calls by setting the external IP address and local network addresses in the following screen. Try calls first without these settings, and add them only if you experience calling issues such as failed calls or one-way audio.

    Genl Settings:SIP Protocol:Signaling:Codecs

    In order of priority, move desired Codecs from right to left by clicking on + icons. If you plan to use the IAX or SCCP protocol for phones and/or trunks, also select Default Codecs under General Settings:IAX Protocol:Default and General Settings:SCCP Protocol tabs, respectively.

    Genl Settings:SIP Protocol:Signaling:DNS

    For DNS Manager and Server Lookup support (required for some SIP providers), enable the DNS Request field:

    IPBX Configuration:Contexts

    XiVO differs from some other Asterisk implementations in the way it manages the routing of calls. XiVO uses Contexts to define what constitute Internal calls (Default), External calls (Outcalls), and Incoming calls (Incalls). Think of these contexts as dialing rules. They define how the three categories of calls are managed internally by the XiVO PBX and determine which callers can do what with your PBX resources. XiVO uses dial strings and ranges of phone numbers to manage and constrain how various classes of calls are routed. The reason for these call specifications is pretty simple. You don’t want outside callers dialing into your PBX and making outbound calls using your PBX trunks on your nickel.

    Some basic settings to enable internal calls and allow creation of user accounts were configured when you set up your XiVO PBX by running the configuration script. However, before anyone can make or receive calls to/from outside the XiVO PBX, you’ll need some additional specifications.

    Edit the from-extern (Incalls) context and click Incoming Calls tab then the + icon. Add a range of DID numbers for incoming calls that will be allowed. These are the phone numbers assigned to SIP and IAX trunks that were acquired through commercial providers such as Vitelity. Note that the example below assumes that your incoming DID trunks deliver calls with 10-digit numbers. If you’re using a service such as Google Voice that delivers calls with 11-digit numbers starting with a 1, then add an additional range of numbers starting with a 1. If the provider delivers calls with +44, then you’d add an additional range with that prefix. Click Save once you’ve entered your settings.

    Let’s also modify the Default context to support MeetMe conferencing for your server. Edit the default context and click Conference Rooms tab then + icon. For the extension range, enter 2663-2665. 2663 spells C-O-N-F by the way. Then click Save. If you have a DAHDI timing source on your server, you then can add conferences: IPBX Setting:Conference Rooms. If you don’t have a DAHDI timing source or you don’t know what any of this means, keep reading. There’s an easier way to set up a conference room for your users.

    While you’re still in the (2) Default context, click on the (3) General tab and (4) move all of the sub-contexts to the left (Selected) column. (5) Then click the Save button.

    General Settings:Advanced (Time Zone)

    IPBX Settings:Users:Add User

    Before you can actually make or receive calls with XiVO PBX, you’ll first need at least one User, Extension, and Line. So click on the (1) Users tab and then (2) the + icon and Add option (as shown below) to get started.

    Use the General tab entries below as a guide to create your first user account. You only need to fill in options (1) and (2) if you would like this user to receive a simultaneous call on a mobile phone whenever this user’s internal phone rings.

    In the Lines tab, assign an internal phone number for this user. By default, the initial configuration script created a range of extension numbers for you: 701-799. This can be changed in the next section to meet your specific requirements.

    Once you’ve chosen an extension, click the Save button and a Line will automatically be generated to associate with your new User account.

    Next, goto IPX Settings:Lines and click the pencil icon to obtain your SIP username and password credentials. You’ll need these to connect a SIP phone or softphone to your user account.

    While you’re obtaining your username and password SIP credentials, fill in the blanks for the Line and click Save:

    IPX Settings:Users (Voicemail Setup)

    There are two steps to setting up voice mailboxes correctly. First, you need to configure the voicemail system defaults to accommodate your required time zones. The system only comes with support for Europe/Paris.

    Go to (1) IPX General Settings:Voicemails and (2) click Time Zones tab and then (3) + Add. (4) Name your new time zone, (5) select the correct Time Zone from the pull-down list, and (6) add the following under Options and (7) Save your entry:

    'vm-received' q 'digits/at' kM
    

    Go to (1) IPX Settings:Users, edit your (2) User account, and click the (3) Voicemail tab. (4) Click the + icon to Add a new Voicemail account. (5) Check Enable Voicemail. (6) Fill in the form using the sample below. Be sure to choose the correct Time Zone for your voicemails. Uncheck Delete message after notification to retrieve voicemail messages by dialing *98 from an extension. (7) Click Save.


    Setting Up a Ring Group in XiVO

    A ring group is a collection of extensions to which calls can be routed. In XiVO terminology, they’re known as Groups. Extensions in a Group can be set to ring simultaneously or in one of six round-robin configurations based upon factors such as previous call volume. Before you can create a ring group, you first have to enable a range of extensions to dedicate to Groups. Edit the Default context, click the Groups tab, and then click the + Add icon to add a range of extension numbers:

    To create a new ring group, choose IPX Setttings -> Groups and click the + Add icon. A typical setup to ring all extensions simultaneous and play a ring tone to the caller would look like this:

    Next, click on the Users tab and move the desired extensions to the the selected side of the window. Then click Save.

    Setting Up Trunks and Routes for XiVO Calling

    Before you can make calls to phones outside your PBX or receive calls from outside your PBX, you’ll need one or more trunks. We’ve simplified the process of setting these up by providing step-by-step tutorials for the leading trunk providers. They are reproduced below for ease of reference:

    XIVO Trunk Implementation Tutorials

    Once you’ve added one or more trunks, you’ll need to tell XiVO how to route outgoing and incoming calls. Here are our step-by-step tutorials on setting up Outbound Calling Routes and Incoming Call Routes:

    XIVO Call Routing Tutorials

    Deploying Google Voice with OAuth on XiVO PBX

    Beginning in mid-August, 2016, native Google Voice with OAuth support became available on the Incredible PBX for XiVO platform. It supports deployment of multiple Google Voice trunks on any XiVO server. This new Nerd Vittles tutorial will walk you through implementation.

    Using an SMTP Mail RelayHost with Postfix

    To cut down on spam, many ISPs no longer allow SMTP mail traffic that originates from downstream mail servers. If your server is connected to an ISP such as Comcast, that would be you. Here’s how to reconfigure the Postfix mail server included with XiVO to process your outgoing emails using your ISP as a mail relay.

    First, edit /etc/postfix/main.cf and search for relayhost. Replace it with the entries below. If it’s not in the file, then just add the following entries to the end of the file:

    relayhost = smtp.comcast.net:587
    smtp_sasl_auth_enable = yes
    smtp_sasl_password_maps = hash:/etc/postfix/sasldb
    smtp_sasl_security_options = noanonymous
    

    Next, create /etc/postfix/sasldb and add the following entries: your ISP (smtp.comcast.net) followed by a TAB and then your full comcast login name, a colon, and your Comcast password. No spaces! Save the file.

    Next, create a hashed version of the file: postmap sasldb

    Then restart Postfix: /etc/init.d/postfix restart

    Now send yourself a test email like this:

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

    Getting Started with SQLite3 on the XiVO Platform

    Here are a couple SQLite3 queries to get you started with syntax:

    sqlite3 /var/lib/asterisk/agi-bin/zipcodes.sqlite "select zip,city,state from zipcodes where zip=29401;"
    sqlite3 /var/lib/asterisk/agi-bin/asteridex.sqlite 'select name,out from user1 where name LIKE "%Airlines%";'
    

    A bonus script in /root will let you convert existing MySQL databases to SQLite3. For example, if you’re currently using AsteriDex on another Incredible PBX platform, it only takes a couple seconds to convert your MySQL database to SQLite3. The syntax to run the script looks like this:

    ./mysql2sqlite3.sh -u root -ppassw0rd yourdatabase | sqlite3 yourdatabase.sqlite
    

    Move the script to the server on which your existing MySQL databases are stored and run it there using the above syntax. Then copy the asteridex.sqlite file to your XiVO server and save it in /var/lib/asterisk/agi-bin.

    Getting Started with Incredible PBX Call Logs

    To retrieve SQLite3 call log data, here are a few examples to get you started:

    ALL: sqlite3 /var/log/asterisk/master.db "select * from cdr"
    DATE: sqlite3 /var/log/asterisk/master.db "select * from cdr where calldate >= '2016-05-22'"
    NPA: sqlite3 /var/log/asterisk/master.db "SELECT * from cdr WHERE clid LIKE '%<843%'"
    DEST: sqlite3 /var/log/asterisk/master.db "SELECT * from cdr WHERE dstchannel LIKE '%411%'"
    FLDS: sqlite3 /var/log/asterisk/master.db "PRAGMA table_info(cdr)"

    To retrieve the CDR log in CSV format suitable for spreadsheets, download:

    /var/log/asterisk/cdr-csv/Master.csv
    

    Managing Your Logs with XiVO

    XiVO is a busy place especially on a busy PBX. Call logs and traditional Asterisk and Linux logs grow like crazy. We have added the following entries to /etc/crontab to assure that you don’t inadvertently run out of disk space on your server. Modify them to meet your own requirements.

    10 1    * * *  root    rm -f /tmp/tts* > /dev/null 2>&1
    11 1    * * *  root    rm -f /var/log/asterisk/*.gz > /dev/null 2>&1
    11 2    * * *  root    rm -f /var/log/asterisk/*.1.gz > /dev/null 2>&1
    12 1    * * *  root    rm -f /var/log/*.gz > /dev/null 2>&1
    12 2    * * *  root    rm -f /var/log/*.1.gz > /dev/null 2>&1
    

    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.

    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 we installed last week, 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)
    

    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.


    Setting Up a Softphone or 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.

    Prefer to use WebRTC from your browser as a softphone? XiVO has you covered. Complete setup instructions available here.

    For everyone else, we recommend YateClient 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.

    Test Drive of Sample Incredible PBX Apps

    Once your softphone is registered, you can try out some of the Incredible PBX sample applications:

    • 4871 (IVR1) – Allison’s Demo IVR
    • 411 (Voice Dialing) – Call by Name (try "Delta Airlines")
    • 2663 (CONF) – Conference Room with Music on Hold
    • 951 – Yahoo! News Headlines (TTS)
    • 947 (ZIP) – NWS Weather by ZIP Code
    • 53669 (LENNY) – The Telemarketer’s Worst Nightmare

    You can review the Dialplan code in the GUI by choosing IPBX Configuration:Configuration Files and clicking xivo-extrafeatures.conf. The sample IVR code is in ivr-1.conf. This Nerd Vittles tutorial will walk you through building your own IVRs for XiVO.

    Using PBX Status with XiVO

    For those that like to see how things are going from the Linux CLI, a modified version of pbxstatus is available for XiVO. From the Linux CLI, type: pbxstatus.

    Using FQDNs with the Travelin’ Man 3 Firewall

    If you plan to use FQDNs with your IPtables firewall or if your remote users will be using a Dynamic DNS provider to keep their IP addresses fresh, be sure to review Step #5 in the Travelin’ Man 3 tutorial which explains how to configure your firewall to automatically refresh IP addresses based upon changes in dynamic addresses. All of the necessary components already have been activated. Simply insert your FQDN entries using /root/add-fqdn and modify /root/ipchecker.

    PortKnocker for XiVO: Your Firewall Safety Net

    If you use a dynamic IP address for your local PC and that address changes, you may find yourself locked out of your own server unless you have heeded the advice in the preceding section. But there’s still hope. Incredible PBX for XiVO now includes the PortKnocker utility which lets you ping three predefined TCP ports in sequence to regain access to your server. You can read all about PortKnocker in this Nerd Vittles article. Unfortunately, PortKnocker doesn’t do you a bit of good if you haven’t deciphered what the three-port secret handshake is for your server. Before you forget, review /root/knock.FAQ and put the information in a safe place where you can retrieve it if the need should ever arise.

    Adding a PPTP VPN to XiVO

    Microsoft introduced the Point-to-Point-Tunneling-Protocol (PPTP) with Windows 95. Back then we knew it as Dial-Up Networking. Suffice it to say that, in those days, PPTP was anything but secure. Unfortunately, the bad name kinda stuck. For the most part, the security issues have been addressed with the possible exception of man-in-the-middle attacks which are incredibly difficult to pull off unless you are a service provider or have access to the wiring closets of your employer. You can read the long history of PPTP VPNs on Wikipedia for more background. If you’re traveling to China or other democracy-challenged destinations, you probably shouldn’t rely upon PPTP for network security. If these security considerations aren’t applicable in your situation, keep reading because PPTP VPNs are incredibly useful and extremely easy to deploy for an extra layer of VoIP and network security in most countries that have severe wiretapping penalties in place.

    PPTP VPNs also provide home-away-from-home transparency to home office network services. Simply stated, with a PPTP VPN, you get a private IP address on the XiVO PBX that lets you do almost anything you could have done sitting at a desk in the home office. PPTP VPNs probably won’t work on most OpenVZ platforms such as Wable and ImpactVPS. But they work great on virtual machines such as CloudAtCost and Digital Ocean. For a quick-and-dirty back door into your server, a PPTP VPN is hard to beat. Here’s how to set one up on your XiVO PBX using 128-bit encryption. Make up a very obscure username and password in the first two lines below:

    PPTPUSER=somebodyspecial
    PPTPPASS=someverysecurepassword
    apt-get -y update
    apt-get -y install pptpd
    sed -i 's|#ms-dns 10.0.0.1|ms-dns 8.8.8.8|' /etc/ppp/pptpd-options
    sed -i 's|#ms-dns 10.0.0.2|ms-dns 8.8.4.4|' /etc/ppp/pptpd-options
    echo "localip 172.16.16.100" >> /etc/pptpd.conf
    echo "remoteip 172.16.16.101-199" >> /etc/pptpd.conf
    echo "$PPTPUSER pptpd $PPTPPASS *" >> /etc/ppp/chap-secrets
    /etc/init.d/pptpd restart
    # show logged in PPTP users
    last | grep ppp
    

    Connect to your PPTP server from a Windows or Mac in the usual PPTP way. Once connected, you will be assigned an IP address in the range of 172.16.16.101-199. You then can access your XiVO PBX on the following IP address: 172.16.16.100.

    Everything You Need to Know About XiVO Backups

    Another feature of XiVO that separates the men from the boys is its documentation. In the case of backups, you’ll find everything you need to know here. All backups are stored on your XiVO server’s local drive in /var/backups/xivo. Be sure you have ample storage space available and, if you’re smart, you’ll copy both data.tgz and db.tgz from the local drive to a safe remote location periodically just in case disaster strikes. The documentation shows you how to quickly restore a backup should that ever become necessary.

    Upgrading XiVO to the Latest Release

    The XiVO development cycle is nothing short of miraculous. A new version is released every three weeks! The average time to close a bug has dropped from 315 days in 2009 to 28 days in 2012! You’ll probably want to keep your system current. 🙂

    Upgrading XiVO is even easier than restoring a backup. Upgrade documentation is available here. Because we’ve added the Travelin’ Man 3 firewall, we recommend stopping IPtables during an upgrade and then restarting it when you’re finished. Your phone system is disabled during the upgrade. When upgrading XiVO, remember to also upgrade all associated XiVO Clients. Be sure to verify that things are back to normal once the upgrade procedure is completed: xivo-service status.

    The commands to upgrade your XiVO PBX are as follows:

    /etc/init.d/netfilter-persistent stop
    xivo-upgrade
    iptables-restart
    # restore Incredible PBX module and ODBC configuration
    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
    xivo-service restart
    # code below reactivates Incredible PBX web apps
    cd /
    wget http://incrediblepbx.com/incredible-nginx.tar.gz
    tar zxvf incredible-nginx.tar.gz
    rm -f incredible-nginx.tar.gz
    /etc/init.d/nginx restart
    

    Google Voice CLI and SMS Messaging Support

    Thanks to Nick Pettazzoni, beginning with the August 29, 2016 release of Incredible PBX for XiVO, you now can take advantage of the pygooglevoice implementation of gvoice as well as Nerd Vittles’ SMS messaging and message blasting utilities. If you’re using an earlier release, it’s easy to add this functionality to your server as well:

    cd /root
    wget http://incrediblepbx.com/install-gv-cli
    chmod +x install-gv-cli
    ./install-gv-cli
    

    Be advised that the Google Voice CLI interface (gvoice) uses plain-text Google Voice passwords, not OAuth. Before most Google Voice accounts will work with gvoice and smsblast, you’ll need to do the following and then immediately login to gvoice from the Linux CLI at least once to mark your account as safe for access from this location. Here are the steps:

    1. Log in to the Gmail account you plan to use with gvoice
    2. While logged in, open a new browser tab to this site and enable Less Secure Apps
    3. Open another browser tab and enable the Google Reset procedure here
    4. Return immediately to the Linux CLI and login to gvoice

    Creating an SMS Message Blast with XiVO

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

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

    Incredible PBX Application Quick Start Guide

    Here’s a quick refresher on some of the Incredible PBX applications that have been installed. There’s also a link for more information. This remains a work-in-progress so expect more applications in coming weeks.

    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. See also the previous 7 Nerd Vittles XiVO tutorials, all of which are listed below. Enjoy!


    Taking Nerd Vittles’ XiVO IVR for a Test Drive

    There’s a Demo IVR running at www.pacificnx.com on their XenServer virtualization platform. Scott McCarthy, a leading outside XiVO developer and a principal at PacificNX, advises they have a $50 a month GOLD platform specifically tailored to XiVO for those needing 99.999% reliability, 24/7 support with nightly backups and enterprise level firewalls that have intelligence to stop attacks and look for viruses, spyware and more. That’s what you’ll be hearing when you call the Nerd Vittles Demo IVR:

    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)

    Don’t Stop Reading Just Yet. We’ve been busy since this article was first published in June, 2016. Continue reading about the latest developments including XiVO Snapshots.

    Published: Monday, June 27, 2016  Updated: Regularly



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