Category: Telephony

2016, Independence Day: Introducing the Gotcha-Free Incredible PBX for XiVO

True story. What became a major career move for us began in the disco era about a block away from the Lincoln Memorial. It was during a meeting with another lawyer and a federal judge at my Watergate apartment that I altered my appellate legal career aspirations to focus more on technology in the legal marketplace. So it seems only fitting, after nearly a decade supporting the FreePBX® platform, to return to those roots once again to celebrate our independence as 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!

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:

  • Travelin’ Man 3 IPtables Firewall Whitelist
  • CallerID Superfecta
  • Festival Text-to-Speech for Asterisk
  • Voice Recognition for Asterisk
  • IVR Sample Template
  • Music on Hold
  • DISA Support
  • Telephone Reminders for XiVO
  • Web Reminders for XiVO
  • AsteriDex with Click-to-Dial
  • Conference Bridge Conferencing
  • Yahoo News by Phone
  • Weather Forecasts by Phone
  • NGINX Support for Web Apps
  • Speed Dialing by Voice or Code
  • DNS Manager for Asterisk
  • SQLite3 Support for XiVO
  • PBX Status Template
  • ODBC Support for SQLite3
  • NeoRouter VPN
  • Automatic Update Utility
  • SQLite3 CDR Logging
  • CSV CDR Logging
  • Blacklist Feature Set for XiVO
  • FCC Telemarketing BlackList Support

The 3 Flavors of Incredible PBX for XiVO

To kick off our Independence Day celebration, we are releasing three new Incredible PBX turnkey installers for XiVO. Because of the numerous platforms on which XiVO will run, we’ve chosen to build three separate installers to address different types of installations.

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). You choose an Incredible PBX installer 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. Here is a quick thumbnail explaining the suitability of each of the three Incredible PBX installers for XiVO:

IncrediblePBX13-XiVO1.sh – Debian 8 (64-bit) minimal platform where XiVO is not installed. Use with Cloud VMs.

IncrediblePBX13-XiVO2.sh – Debian 8 (64-bit) platform with XiVO installed but not configured. Use with XiVO ISO.

IncrediblePBX13-XiVO3.sh – Debian 8 (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

The installation procedure obviously depends upon your hardware platform. We’ve taken the guesswork out of this for seven platforms by providing detailed tutorials that you can follow. 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, choose the Incredible PBX flavor you need based upon the summaries in the preceding section. Then plug the filename of the Incredible PBX installer into the first line of the following script:

INSTALLER=IncrediblePBX13-XiVO1.sh
cd /root
wget http://incrediblepbx.com/$INSTALLER
chmod +x $INSTALLER
./$INSTALLER


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.

General Settings:SIP Protocol



General Settings:SIP Protocol:Signaling (Default 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.

General Settings:SIP Protocol:Signaling (DNS Manager/srvlookup)

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

General Settings:Advanced (Time Zone)

IPBX Settings:Users (Creating a Default User Account)

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)

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. Uncheck Delete message after notification to retrieve voicemail messages by dialing *98 from an extension. (7) Click Save.

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. 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.

Setting Up Trunks and Routes for Outbound and Incoming Calls

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. Start here.

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.

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 strongly recommend adding entries similar to the following in /etc/crontab to assure that you don’t inadvertently run out of disk space on your server:

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.

We use 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.

Taking the Sample Incredible PBX Apps for a Test Drive

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.

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

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)

Published: Monday, June 27, 2016





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


 
Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For our users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage. Any balance is refundable if you decide to discontinue service with Vitelity.


​​3CX is a software PBX that’s easy to install & manage. It includes integrated softphones, WebRTC conferencing and essential add-ons out of the box, at no additional cost. Try the free edition at www.3cx.com.

  • Run on Premise or in the Cloud, on Windows and soon Linux
  • Softphones for iOS, Android, Win & Mac
  • Easy install, backup & restore, version upgrades
  • Automatically configures IP Phones, SIP Trunks & Gateways

  • Some Recent Nerd Vittles Articles of Interest…

    As Easy As 1-2-3: Introducing Incredible PBX Telephone Reminders for XiVO

    We’re back in the saddle today after a terrific two-week break in Hawaii. Thanks for your patience while we recharged our batteries. Today we’re pleased to introduce Incredible PBX Telephone Reminders for the XiVO platform. It’s a 15-second installation on any XiVO server.

    How It Works. The original functionality of the application has been preserved. Dial 1-2-3 on a phone connected to your XiVO system and enter the password you assigned during installation. Then you can record a reminder message, specify the phone number to which the reminder should be delivered, schedule the date and time for delivery, and decide whether to enable recurring reminders of one of the flavors supported: single reminder, weekday reminder, daily reminder, monthly reminder, or annual reminder. There also a web interface to schedule reminders using the Festival TTS engine. Simply visit http://XiVO-ipaddress/reminders for access. If you were a pioneer and installed Telephone Reminders before the web component was available, just download the latest tarball below and run the installer again. As with all Nerd Vittles software, Telephone Reminders is pure (and free) GPL code with No Gotchas!

    Installing Telephone Reminders for XiVO. As mentioned, installation is as simple as downloading and running a script. Log into your XiVO server as root and issue the following commands to put everything in place:

    cd /root
    wget http://incrediblepbx.com/reminders-xivo.tar.gz
    tar zxvf reminders-xivo.tar.gz
    ./reminders-xivo.sh
    

    Scheduling Reminders by Phone with XiVO. Once you complete the install, pick up any telephone connected to your XiVO server and dial 1-2-3. When prompted, enter the numeric password that you assigned during installation. If you ever want to change it, just download the latest version and run the installation script again. Doesn’t get any easier than that to stay current. Keep in mind that anyone with access to extension 123 on your XiVO server can schedule reminders, and the reminders can be sent to any telephone number in the world that is supported by your dialplan and trunks. So use a secure password! It’s your phone bill.

    Scheduling Reminders via the Web with XiVO. As noted, there’s also a web interface to Telephone Reminders. Make certain you have first installed the Incredible PBX Firewall WhiteList so that the bad guys don’t schedule costly phone calls on your dime! To access Telephone Web Reminders, point your browser to the IP address of your server using standard http, e.g. http://12.34.56.78/reminders. The web component uses the Festival TTS engine to transcribe your reminders for delivery by phone.

    Firewall Installation on XiVO Platform. If you didn’t install XiVO using one of our tutorials, please add the Travelin’ Man 3 IPtables Firewall now!

    apt-get -y install dnsutils
    echo "*/10 5-22 * * * root /root/ipchecker > /dev/null 2>&1" >> /etc/crontab
    wget http://incrediblepbx.com/firewall-xivo.tar.gz
    tar zxvf firewall-xivo.tar.gz
    rm -f firewall-xivo.tar.gz
    ./tm3-xivo.sh
    

    Historical Overview and Tweaking. For the big picture of the Telephone Reminders project and to learn some of the additional settings you can configure, visit the Best of Nerd Vittles page. Enjoy!

    Published: Monday, June 20, 2016





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


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For our users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage. Any balance is refundable if you decide to discontinue service with Vitelity.


    ​​3CX is a software PBX that’s easy to install & manage. It includes integrated softphones, WebRTC conferencing and essential add-ons out of the box, at no additional cost. Try the free edition at www.3cx.com.

  • Run on Premise or in the Cloud, on Windows and soon Linux
  • Softphones for iOS, Android, Win & Mac
  • Easy install, backup & restore, version upgrades
  • Automatically configures IP Phones, SIP Trunks & Gateways

  • Some Recent Nerd Vittles Articles of Interest…

    2016: The Year of the May Bromance with XiVO, Asterisk 13, and the GPL

    Remember that girl that you never much noticed in high school that happened to cross your path years later and there was that instant attraction. It’s one of our favorite country songs.




    What a month it has been. 30 days ago we’d never met. And then it happened. We were introduced to XiVO by a single post on the PIAF Forum. It’s difficult to choose a single adjective that describes how feature-rich XiVO is as a real-time Asterisk platform. Ten years of development coupled with some really talented Canadian developers puts XiVO in a league of its own. Pure GPL code. No gotchas. No hidden agenda. Incredible documentation. No snake oil salesmen trying to peddle add-ons for money. And, thanks to the endless patience of Sylvain Boily and Pascal Cadotte Michaud, two of the key XiVO developers, we’ve had quite a joy ride this month. Today we want to share what we’ve learned and provide everything you need step-by-step to hit the ground running with XiVO. In less than an hour, you’ll be sitting in front of your own XiVO server, and you can judge for yourself. All we can ask is “Where have you been all these years?” XiVO really is THAT GOOD!

    Let’s Start with the Features. If you make frequent changes to your Asterisk platform, then you’ll really appreciate XiVO’s realtime implementation. Changes are loaded almost instantaneously. Contrast that with some other platforms with hundreds of users where simple changes require several minutes or more to reload the Asterisk dialplan. Here’s an example:



    Speaking of hundreds of users, consider the time required to generate accounts and phone configurations for hundreds of users. With XiVO, a simple spreadsheet can be used to build all the user accounts in seconds. And XiVO’s Endpoint Manager supports all of the major phone manufacturers and configures your phones in seconds. With other solutions, you’ve got a fee for the Endpoint Manager add-on itself and then another annual maintenance fee to assure that the software will continue to work. Contrast that with XIVO’s GPL alternative. Choosing Configuration:Provisioning:Plugins:Update generates templates for dozens of great SIP phones in seconds with just four button clicks. Adding your own new models is a breeze.



    We’ve already written about XiVO upgrades and backups, the twin-edged sword with most Asterisk implementations. Yes, you can go the proprietary route and stumble through dozens of menus and arcane commands to load upgrades and create backups. With XiVO, backups are automatic and the updates are fast and furious. Every morning a new backup arrives in /var/backups/xivo, and a new version of XiVO is released every three weeks. You can upgrade your server in under a minute with one simple command. Did we mention reported bugs are addressed in under 30 days!

    One of the real failures of the other GUI offerings is their lack of tools to perform most tasks in any way other than using the GUI. For developers and those that maintain numerous phone systems from afar, this becomes a nightmare. XiVO offers a better alternative. Yes, there’s an incredible GUI. But there’s also a robust collection of APIs that provide programmatic access to manage and query virtually every piece of the XiVO puzzle. As if that weren’t enough, there’s also real documentation and samples to show you how to use each piece.



    Have we got your attention yet? We haven’t even touched upon Call Centers and High Availability (HA) server deployments, but they’re included at zero cost. For today and just getting started, suffice it to say that XiVO covers all the bases with open source code. And all of these features already are incorporated into the product you’re about to install. So, if those two items have been on your Wish List and you’d prefer to avoid paying an arm and a leg for proprietary, commercial add-ons with recurring annual fees, today’s your lucky day. Break open the latest XiVO documentation, all 479 pages of it and enjoy!

    We’ve barely scratched the surface of what you can do with XiVO. Simply stated, anything you can do with the other GUI offerings, you can do better, quicker, and cheaper with XiVO. And, if there are features you need that aren’t there, all you have to do is ask. We’re porting applications to XiVO at the rate of about one new application a day. You can do the math on feature sets and measure where we’ll be when the summer is over.

    Getting Started with XiVO. Jump to the latest tutorial to install XiVO.

    Bookmark Getting Started with XiVO and you can follow our progress in coming months. Enjoy!

    For additional tips & tricks with XiVO, take a look at our previous articles:

    2016, The Year of (real) VoIP Choice: Meet XiVO, a UC Solution for Any Business
    The XiVO Adventure Continues: Adding Incredible PBX Goodies to Your Sandbox
    2016, Celebrating The Preakness: CallerID Superfecta Rides Again with XiVO
    Choosing Wisely: Mastering Asterisk IVR and AutoAttendant Design with XiVO

    Choosing a XiVO Hardware Platform

    XIVO Initial Setup Tutorial

    XiVO Initial Setup Tutorial: Getting Started with XiVO

    XIVO Trunk Implementation Tutorials

    XIVO Call Routing Tutorials


    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, tells us they soon will have a $20 a month platform specifically tailored to XiVO. And 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)

    Published: Tuesday, May 31, 2016





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


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For our users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage. Any balance is refundable if you decide to discontinue service with Vitelity.


    ​​3CX is a software PBX that’s easy to install & manage. It includes integrated softphones, WebRTC conferencing and essential add-ons out of the box, at no additional cost. Try the free edition at www.3cx.com.

  • Run on Premise or in the Cloud, on Windows and soon Linux
  • Softphones for iOS, Android, Win & Mac
  • Easy install, backup & restore, version upgrades
  • Automatically configures IP Phones, SIP Trunks & Gateways

  • Some Recent Nerd Vittles Articles of Interest…

    Choosing Wisely: Mastering Asterisk IVR and AutoAttendant Design with XiVO




    Today we want to talk a little about design choices and IVRs. First and foremost, we don’t want to leave anyone behind during our XiVO adventure. XiVO is a platform that adjectives really can’t describe. It’s that good and, frankly, we’re having a hard time believing it’s been around for almost a decade and nobody much talked about it. Leave it to the crazy Americans to only look at stuff from the U.S. of A. Funny thing is that the two major GUIs for Asterisk® now are both Canadian-based.


    One of our PIAF Forum readers posted a comment last week that said:

    The only downside I see is that XiVO does not have [a] GUI for building IVRs. To build [a] complex, nested IVR system, everything has to be thought about in great detail writing contexts and dial plans to suit your unique requirements. It would be nice if XiVO offered a GUI for building IVRs.

    This raises some issues about GUI design and development that are worth addressing. As with any GUI, the development cycle is lengthy and incredibly complex. This is especially true with XiVO where new versions are released every two weeks! In our second XiVO article, we showed how easy the upgrade procedure was. Those coming from other Asterisk platforms will appreciate this little shocker. XiVO doesn’t break stuff with their upgrades. Frankly, the only other company I can say that about is SONOS. If you don’t have their music platform, you’re missing a treat.

    Introducing new components into any “main product” can cause all sorts of problems with the pieces that used to work. If you don’t believe it, look at some of the “other forums” and look at the number of message threads complaining that the new X Widget broke the Y widget and now nothing works. While we can’t speak for everyone, I think it’s safe to say nobody that depends upon their phone system wants to see it go up in flames regularly because some developer had a great new idea that didn’t quite do what it was supposed to do.

    To their credit, the XiVO developers were smarter than that. They’ve not only built a mighty mousetrap, but they’ve done it in a way that supports outside integration of additional components without breaking the main product. There are numerous “hooks” that allow anyone with any skill set to add missing pieces. Some of these hooks are exclusively for programmers, but many were designed to let anybody integrate almost anything into the XiVO platform.

    So, when a user says “I wish XiVO had an IVR Builder in the GUI,” our first inclination was to chuckle and respond with “You just don’t appreciate how lucky you are not to have an IVR Builder in the GUI.” What the commenter didn’t appreciate is that you don’t need to pre-build components with XiVO before developing an IVR. With the “other” GUI, you first had to create Custom Destinations and Custom Contexts and Miscellaneous Applications and Miscellaneous Destinations and Custom Recordings in the GUI before you could take advantage of the IVR GUI to build much of anything. Think about that for a minute. Yes, there was an IVR builder but, before you could use it, you first had to transform every component to be incorporated into the IVR using a large number of subcomponents to translate all of your Asterisk pieces into the GUI’s special lingo. Think of them as GUI pigeonholes, and you had to decipher which Asterisk square pegs went in which GUI round holes. We can’t count the number of times we’ve begun the IVR creation process only to have to stop and create missing components because the IVR builder simply wouldn’t recognize a feature as being part of our Asterisk dialplan.

    Building IVRs and AutoAttendants with XiVO

    The anatomy of an IVR in Asterisk could not be more straight-forward. You have a prerecorded message that plays to the caller giving them choices from which to choose from a menu of selections. The caller presses one of the 12 keys on their phone, and the IVR goes off and does some task: calls an extension, plays a recording, runs an Asterisk application, makes an outside call, or kicks off another IVR with another recording and more choices. Some options in the IVR may not be mentioned, and this is commonly referred to as the Stealth AutoAttendant. None of this is rocket science.

    To build an IVR, you need these components: (1) a prerecorded message, (2) a list of the choices you want to provide to the caller with the corresponding destinations on the PBX to execute those choices, and (3) a template to follow to create the IVR dialplan code in XiVO.

    Trust us when we say the major problem with IVRs is not that they’re difficult to build in XiVO. The real issue with most IVRs is that the person that implemented the IVR spent all their time worrying about the mechanics of PBX implementation and didn’t put sufficient thought into the IVR layout and the caller’s experience when actually interacting with the IVR. If you haven’t heard Allison Smith speak about IVR design, put it on your Bucket List for the next AstriCon or do some reading. That’s a long-winded way of saying that filling in the blanks of an IVR template is just as easy as point-and-click or drag-and-drop except for the eye candy. Just be thankful the XiVO platform gives you the flexibility to do it yourself without having to create imaginary destination hooks and recording linkages before they can be used in the product’s IVR GUI because the developers didn’t have the foresight to think outside their own GUI’s box. Every Windows user can appreciate that problem.

    For today, we’re assuming you’ve done your homework and have already sketched out the options you want to incorporate into your IVR or IVRs. No GUI can help with that! So we’ll pick up from there and show you how easy it is to incorporate your IVR design into XiVO.

    1. Incorporating Your Prerecorded Messages into XiVO

    For openers, you obviously need a recording to greet callers and tell them what their choices are when using your IVR or AutoAttendant. You can build these recordings yourself on the XiVO platform or, for a more professional IVR, you can send the text off to Allison Smith and let her record the voice prompts for you. Digium makes it easy. Visit their web site, type in the text, and you’ll have your recording in a couple of days. No, they’re not free, but they’re not expensive either.

    Since we’re just getting started, let’s assume you want to create a recording prototype on your own to work out the kinks in your IVR first. Here’s how. We’re assuming you’ve already read the Nerd Vittles XiVO tutorial and put the Festival TTS platform in place. Next, log into your XiVO server as root. To keep things simple, let’s put the recordings in WAV format in the /var/lib/xivo/sounds/playback directory which is reserved for our custom recordings:

    cd /var/lib/xivo/sounds/playback
    

    To actually generate the sound file that Asterisk can play back, execute the command below after placing your text between the quotation marks and giving the sound file a name, e.g. ivr-number1.wav:

    echo "Text goes here" | /usr/bin/text2wave -F 8000 -o ivr-number1.wav
    

    Here’s an example:

    echo "Thank you for calling. Press 1 for Tom, 2 for Dick, or 3 for Harry. Press 0 to be connected to the operator." | /usr/bin/text2wave -F 8000 -o ivr-number1.wav
    

    2. Marrying IVR Choices to PBX Destinations

    Whether you’re deploying an IVR using FreePBX® or XiVO, you still have to translate your Plain English options into code that the GUI understands so that calls get routed successfully to the intended destinations.

    Let’s begin with the FreePBX Way. Our previous IVR tutorial showed how it was done:




    As you can see from the above routing procedure, there were interim steps for every single option in this IVR menu except #8. What you may not appreciate is that you first had to create both a Misc Destination AND a Custom Extension before these options could be used in FreePBX. Otherwise, the options simply didn’t appear in the IVR GUI’s pull-down pick lists.




    If you wished to incorporate a custom context that wasn’t assigned an extension number on your PBX, there was a different GUI procedure. For something as simple as retrieving the time of day, you had to get the custom context registered with FreePBX before the dialplan code could be used in the IVR. According to the FreePBX developers, this functionality was considered an “advanced feature and should only be used by knowledgeable users.”



    Our purpose in documenting all of this is to demonstrate that building IVRs even in a GUI is much more than point-and-click. It requires mastery of some fundamental Asterisk dialplan concepts not to mention the GUI’s own labyrinth of secret pigeonholes. Once you’ve had to master all of that, we believe it’s simpler to build IVRs using simple commands rather than jumping through all of the convoluted hoops required just to make your IVR GUI platform happy.

    Let’s compare this methodology to the XiVO way of doing things by way of example. Then you can decide for yourself which approach is more complex. Would you know all of these on your own? Probably not. But now you can see how simple it really is. There really are only two words you need to learn: Dial and Goto. 🙂

    Call an Extension: Dial(Local/701@default)
    Call a Ring Group: Dial(Local/801@default)
    Call a PSTN Number: Dial(Local/8005551212@default)
    Call a SIP URI: Dial(SIP/2233435945@rentpbx.mundy.org)
    Access DISA with permission: Dial(Local/3472@default)
    Join a Conference: Dial(Local/2663@default)
    Playback Yahoo News: Dial(Local/951@default)
    Playback Weather Forecast: Dial(Local/947@default)
    Identify IVR Option as Invalid and Repeat Menu: Goto(i,1)
    Hangup on Caller for Choosing Invalid Option: Goto(t,1)
    Execute Time of Day Custom Context: Goto(new-time,s,1)
    Send Caller to a Second IVR and Play Second Recording: Goto(ivr-2,s,3)
    

    Building XiVO IVRs from an IVR Template

    We can’t speak for everyone, but we’ve always told folks not to write a book about how to do something. Just give us an example that’s easy to follow and we’ll take it from there. So here you go.

    In the XiVO world, IVRs are nothing more than custom contexts. They have a name in [brackets], and they’re stored in config files saved in /etc/asterisk/extensions_extra.d. A config file can include multiple contexts or only one. For IVRs, we recommend you save each one in a single configuration file that houses a single context.

    We’re going to give you a template to follow in creating all of the IVRs you can dream up. All you need is a custom recording for each one and your list of choices and destinations for those choices. The examples above tell you everything you need to know to build awesome IVRs.

    After downloading the template, we recommend that you not edit it directly. Make a copy with a new file name and change the context name in the template to match your new file name. We also do one other little trick with all of our custom contexts. They always begin and end with comment lines like this using the context name:

    ;# // BEGIN ivr-template
    ;# // END ivr-template
    

    The reason for this is it makes it incredibly easy to remove the entire context with a single command:

    sed -i '\:// BEGIN ivr-template:,\:// END ivr-template:d' ivr-template.conf
    

    This doesn’t matter so much when you only have a single context in a single file. But it is immensely helpful when you’ve stored dozens of contexts within the same file. Some may prefer to store all of the related IVR contexts for their entire IVR tree in a single file. And then you’ll appreciate this tip when it’s time to make major changes in your IVR.

    Let’s begin by putting your template in place and then cloning it to ivr-number1:

    cd /etc/asterisk/extensions_extra.d
    wget http://incrediblepbx.com/ivr-template.tar.gz
    tar zxvf ivr-template.tar.gz
    rm -f ivr-template.tar.gz
    cp -p ivr-template.conf ivr-number1.conf
    sed -i 's|ivr-template|ivr-number1|' ivr-number1.conf
    

    The rest of today’s exercise can be performed in the XiVO GUI using its built-in editor. Open the GUI with your browser and navigate to Services -> iPBX -> Configuration files and then open ivr-number1.conf by clicking on the pencil icon beside it.

    Anatomy of the XiVO IVR Template

    First things first. Change the sound recording in line s,3 to match the recording you made above without the .wav extension: ivr-number1. Leave the directory path just as it is. So your line should now look like this:

    exten => s,3(skip),Set(IVR_MSG=/var/lib/xivo/sounds/playback/ivr-number1)
    

    Next, take a look at the structure of the file. You’ll note that there are options labeled exten => 0,1, through exten => 9,1,. These match the numeric keys on a telephone obviously. In the IVR world, it’s called a phone tree. All you need to change is what comes after the second comma on each line. This destination should be one of the XiVO commands we documented above telling XiVO how to process the call. For option 0, let’s assume you wanted to route the call to extension 701. Your 0 branch would look like this:

    exten => 0,1,Dial(Local/701@default)
    

    The remaining dial options should be obvious. If you want to designate a particular option to be invalid, make the option look like this:

    exten => 9,1,Goto(i,1)
    

    Another alternative is to remove the line entirely; however, we prefer the above approach because it makes it easy to change things down the road if you decide to use option 9 as a call destination.

    Two other options warrant a brief explanation. The i option tells XiVO how to process the call if the caller chooses an invalid option. The t option tells XiVO what to do if the 3-second timeout occurs without the caller pressing a key. You can modify these to meet your own requirements. As configured, an invalid option sends the caller back to the recording to start over. And the timeout option hangs up the call.




    Finally, phone trees can get quite complex. A GUI can’t fix that either. Pressing option 2 might trigger phone tree 2 while pressing 3 might trigger phone tree 3. Programmers could obviously rewrite the dialplan to handle all of these separate phone trees with their separate branches in one giant, convoluted chunk of dialplan code. But why? Just make each phone tree a separate IVR housed in its own file with its own context. And navigate between the IVRs using simple Goto commands such as Goto(ivr-number2,s,3). To return to the main IVR, do the same thing pointing to the line number to which the call should be redirected, e.g. Goto(ivr-number1,s,3). You obviously don’t need to answer each call but once so skip those lines in the IVR dialplan when choosing the line number to which to redirect processing.

    Routing Incoming Calls to Your IVR

    If you’ve already set up one or more DIDs on your PBX, then you probably routed those Incoming Calls to a user or ring group. Changing the routing to send the calls to your IVR is easy. Just edit the DID entry for the Incoming Calls you wish to redirect and set the Destination to Customized and the destination Command to the context of your IVR: Goto(ivr-number1,s,1). Save your change and you’re all set. Remember, XiVO is a real-time Asterisk server so all of your changes take effect immediately. There’s no rewriting of the entire Asterisk dialplan. 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, tells us they soon will have a $20 a month platform specifically tailored to XiVO. And 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)

    Published: Thursday, May 26, 2016





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


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For our users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage. Any balance is refundable if you decide to discontinue service with Vitelity.


    ​​3CX is a software PBX that’s easy to install & manage. It includes integrated softphones, WebRTC conferencing and essential add-ons out of the box, at no additional cost. Try the free edition at www.3cx.com.

  • Run on Premise or in the Cloud, on Windows and soon Linux
  • Softphones for iOS, Android, Win & Mac
  • Easy install, backup & restore, version upgrades
  • Automatically configures IP Phones, SIP Trunks & Gateways

  • Some Recent Nerd Vittles Articles of Interest…

    2016, Celebrating The Preakness: CallerID Superfecta Rides Again with XiVO


    If you missed The Preakness Saturday, another Triple Crown bit the dust… or the mud in this case. But, if you had the winning Superfecta ticket, you made a $316 profit on your $1 bet. For the rest of us, there’s still a Superfecta win to celebrate, and this one’s free. We’ve begun porting CallerID Superfecta to the XiVO platform and today we’ll share that code with you together with lots of other goodies in our third roundup of Incredible PBX add-ons for the XiVO PBX. If you’re just joining the party, start with the first and second articles on XiVO, and then you’ll be ready to roll up your sleeves for Chapter 3.

    Installing CallerID Superfecta for XiVO

    As we mentioned in April, it’s always nice to see your baby grow up. Nearly a decade ago, we introduced an AGI script for Asterisk@Home known as CallerID Trifecta for FreePBX® 2.2.0. As sources of CNAM lookups expanded, a number of other individuals contributed code to support those lookups. When we added a fourth CNAM lookup source, the original application morphed into CallerID Superfecta. Then we gave up. The source lookups became too numerous to mention.

    For today, we’ve changed the design a bit to better accommodate the XiVO platform. There’s a single AGI script that houses the various CNAM lookup sources and the code to extract CallerID names from those sources. And there’s a dialplan script that let’s you specify which CNAM sources to use and in which order. As with the original release, CallerID lookups take the phone number of the caller and walk through your CNAM lookup sources in the order you specify until a CallerID name match is found. Then the result is returned to the PBX for use with the incoming call. The reason for all of this is historical. The Bell Sisters decided it was more profitable to dump CallerID name information in the bit bucket rather than passing it along with incoming calls. In that way, they could charge folks for looking up the matching name in their proprietary databases. A few CallerID lookup sources remain free, but many now are pay-as-you-go platforms with a typical lookup costing about half a cent. Unfortunately, all providers consider “WIRELESS CALLER” a successful lookup. Ka-Ching! We’ve documented the procedure to add additional CNAM lookup sources on the PIAF Forum. Please share your work!

    This release of CallerID Superfecta provides four lookup sources. That’s what a Superfecta is all about, picking four winners:

    0 - AsteriDex SQLite3 database
    1 - OpenCNAM (free from cache or commercial)
    2 - BulkCNAM (commercial only with free trial)
    3 - TelcoData (provider, city, and state of caller)

    There are three simple steps to putting everything in place. First, run the scripted commands below. Second, specify which CNAM sources you wish to use and in what order. Third, register with the commercial providers you’d like to use and plug your credentials into the CallerID Superfecta script.

    To install CallerID Superfecta, log into your server as root and issue the following commands:

    cd /
    apt-get -y install php5-xmlrpc
    wget http://incrediblepbx.com/cid-superfecta.tar.gz
    tar zxvf cid-superfecta.tar.gz
    rm -f cid-superfecta.tar.gz
    /etc/init.d/asterisk restart
    

    By default, CallerID Superfecta will attempt to use all four of the providers in the order shown to retrieve a CNAM match. If you have migrated your AsteriDex database to XiVO as we covered in last week’s article, then CallerID names will be provided for your most frequent incoming calls without ever accessing external sources. You won’t break anything by leaving all four CNAM sources activated. But, without signing up for service with OpenCNAM or BulkCNAM, your CNAM results will be diminished considerably. And a result of “WIRELESS CHARLESTON SC” from TelcoData doesn’t provide much of a clue as to who is calling. But at least you don’t get charged for that one.

    In the next release, we will add an optional feature that will populate entries in AsteriDex from CNAM data returned from OpenCNAM and BulkCNAM. The good news is, if you leave AsteriDex at the top of the CallerID Superfecta search list, you’ll never pay for the CNAM lookup of the same number twice. The bad news is, to keep the bad guys from self-populating your database with expensive phone numbers, you’ll need to password-protect the Voice Dialing application if it is part of your inbound IVR.

    To change the source list or sequence of CNAM lookups, open the XiVO GUI and navigate to IPX configuration -> Configuration files. Then edit cid-superfecta.conf. Find the line that looks like the following and specify the sources you wish to use and the sequence in which they should be searched using the source numbers listed above to replace 0-1-2-3. Separate your entries with hyphens. Then SAVE the file.

    same = n,AGI(nv-cid-superfecta.php,${XIVO_SRCNUM},0-1-2-3)
    

    To use the commercial CNAM services of either OpenCNAM or BulkCNAM, you first must register with them and provide a credit card. You then will be provided credentials to use for your CNAM lookups. These need to be inserted at the top of /var/lib/asterisk/agi-bin/nv-cid-superfecta.php. Then SAVE the file.



    Activating Traditional Asterisk Call Detail Recordings

    If you want to preserve the numbers AND names of those that call your PBX, you’ll need to activate the traditional CDR reporting mechanisms in Asterisk®.

    To activate SQLite3 logging of calls:

    cd /etc/asterisk
    sed -i 's|no|yes|' cdr.conf
    echo "[master]" >  cdr_sqlite3_custom.conf
    echo "table = cdr" >>  cdr_sqlite3_custom.conf
    echo "columns => calldate, clid, dcontext, channel, dstchannel, lastapp, lastdata, duration, billsec, disposition, amaflags, accountcode, uniqueid, userfield"  >>  cdr_sqlite3_custom.conf
    echo "values => '${CDR(start)}','${CDR(clid)}','${CDR(dcontext)}','${CDR(channel)}', '${CDR(dstchannel)}','${CDR(lastapp)}','${CDR(lastdata)}','${CDR(duration)}', '${CDR(billsec)}','${CDR(disposition)}','${CDR(amaflags)}', '${CDR(accountcode)}','${CDR(uniqueid)}','${CDR(userfield)}'" >>  cdr_sqlite3_custom.conf
    chown asterisk:www-data cdr_sqlite3_custom.conf
    chmod 660 cdr_sqlite3_custom.conf
    sed -i 's|noload => app_cdr.so|;noload => app_cdr.so|' modules.conf
    sed -i 's|noload => cdr_sqlite3_custom.so|;noload => cdr_sqlite3_custom.so|' modules.conf
    sed -i 's|noload => func_cdr.so|;noload => func_cdr.so.so|' modules.conf
    touch /var/log/asterisk/master.db
    chown asterisk:asterisk /var/log/asterisk/master.db
    chmod 640 /var/log/asterisk/master.db
    /etc/init.d/asterisk restart
    

    To also activate CSV logging of calls:

    cd /etc/asterisk
    echo "[csv]" >> cdr.conf
    echo "loguniqueid=yes" >> cdr.conf
    echo "loguserfield=yes" >> cdr.conf
    echo "accountlogs=yes" >> cdr.conf
    sed -i 's|noload => cdr_csv.so|;noload => cdr_csv.so|' modules.conf
    /etc/init.d/asterisk restart
    

    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
    

    Adding Asterisk ULAW Sound Files to Your XiVO PBX

    At least for us, the default sound files distributed with XiVO didn’t work. Here’s how to add the ulaw versions of all the files to your server:

    cd /usr/share/asterisk/sounds/en
    wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-extra-sounds-en-ulaw-current.tar.gz
    wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-en-ulaw-current.tar.gz
    tar zxvf asterisk-extra-sounds-en-ulaw-current.tar.gz
    tar zxvf asterisk-core-sounds-en-ulaw-current.tar.gz
    rm -f *.tar.gz
    chown asterisk:asterisk *.ulaw
    

    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.

    We use 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. Download the DISA dialplan script into your /root folder where it can be edited:

    cd /root
    wget http://incrediblepbx.com/disa-xivo.tar.gz
    tar zxvf disa-xivo.tar.gz
    rm -f disa-xivo.tar.gz
    nano -w disa-xivo.txt
    

    2. 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
    

    3. 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.

    4. 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)
    

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

    6. 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 restart
    

    7. 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)
    

    8. 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:



    9. 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.

    Adding Vitelity to XiVO for Flawless VoIP Calling

    We already have shown you several ways to take advantage of free VoIP calling in the U.S. and Canada as well as internationally. But, the old adage still holds true. You get what you pay for. And, if you’re using XiVO for your business or if you like a good night’s sleep without worrying about whether your spouse is going to stab you because of lousy phone connections, then splurge and spend a penny and a half a minute for outbound calls while getting unlimited incoming calls (4 at a time!) for only $3.99 a month. You’re worth it. The signup link for Vitelity is at the end of today’s article. Once you have your credentials, create a subaccount on the Vitelity site and then you’re ready to set up your Vitelity trunks with XiVO. We’ll use one trunk for incoming calls and a second trunk for outbound calls. The setup procedure for both trunks is already documented on the PIAF Forum. Make that your next stop!


    Simultaneous Cellphone Ringing for Inbound Calls with XiVO

    Speaking of incoming calls, wouldn’t it be nice if your cellphone also rang when XiVO calls arrived on your main extension. Then you don’t have to worry about missing a call just because you stepped out of the office.

    If you took our earlier advice and purchased a RingPlus phone with free monthly service, then you’re already covered. Setting up the RingPlus SIP trunk last week covered all the bases. And, there’s more good news from RingPlus. Now you can buy a phone in their Classifieds section without previously owning a phone. So you can hit the ground running with a phone AND a free calling plan. For example, $149 currently buys a brand new Moto E with 3,000 4G/LTE and SIP minutes, 3,000 SMS messages, and 3,000 MB of LTE data every month. And the monthly cost: ZERO!

    But, let’s assume you’re not the sharpest tool in the shed, and you still want your cellphone to ring when extension 701 rings on your PBX. Here’s how.

    In the User setup for your extension:

    1. Enter your cellphone number in the Mobile Phone Number field. Be sure it includes any necessary dial prefix so that it’s routed out through the correct trunk.

    2. On the same screen, you’ll find a Preprocess subroutine field. Enter the following there: pre-mobility

    3. SAVE your changes.

    Keep in mind that outbound calls in XiVO are routed out using dialing prefixes. If you have set up a trunk with a provider that allows CallerID spoofing such as Vitelity, Anveo Direct, or VoIP.ms, then you can preserve the caller’s original CallerID number on the forwarded call to your mobile phone provided the dial string for your cellphone number matches the format you set up for the trunk you wish to use. For example, if Exten for Vitelity is 8NXXNXXXXXX, then you would enter the number for your cellphone with an 8 prefix: 89991234567.

    Munin Makes XiVO Shine

    If you look under the Services tab and choose Graphics, the World of Munin will suddenly appear. There are literally dozens of gorgeous charts to tell you anything and everything you’d ever want to know about your server’s performance. Enjoy!



    Endpoint Management on Steroids… and It’s FREE

    If you’ve longed for an endpoint manager that would automatically configure your phones, the wait is over. XiVO supports literally dozens of phones out of the box. And the setup is integrated into the setup procedure for the users and devices. To get started, choose the Configuration tab and click Plugins. Next click on the + icon to load the default endpoint config files. We couldn’t do justice to this topic in a blog. That’s what tutorials are for. And XiVO has a 700+ page reference guide that will tell you everything you ever wanted to know about endpoint management.



    Adding NeoRouter VPN to XiVO

    We’ll finish up for this week by showing you how easy it is to add the NeoRouter Client to XiVO. In less than five minutes, you’ll be able to use XiVO’s NeoRouter private IP address to access your server securely from anywhere in the world. Start by reading our last introduction to NeoRouter.

    If you’re running XiVO on a 64-bit platform, issue the following commands to install the free NeoRouter client:

    cd /root
    wget http://download.neorouter.com/Downloads/NRFree/Update_2.3.1.4360/Linux/Ubuntu/nrclient-2.3.1.4360-free-ubuntu-amd64.deb
    dpkg -i nrclient-2.3.1.4360-free-ubuntu-amd64.deb
    

    If you’re running XiVO on a 32-bit platform, do this instead:

    cd /root
    wget http://download.neorouter.com/Downloads/NRFree/Update_2.3.1.4360/Linux/Ubuntu/nrclient-2.3.1.4360-free-ubuntu-i386.deb
    dpkg -i nrclient-2.3.1.4360-free-ubuntu-i386.deb
    

    Unless you want your server identified in NeoRouter as localhost, we recommend changing your hostname and rebooting your server at this juncture. Just edit /etc/hostname and give it a name, e.g. xivo. Then reboot.

    Now log back into your server as root and then log into your NeoRouter client. This will assign a private IP address to your XiVO server. The nrtap entry running ifconfig will tell you what that address actually is.

    nrclientcmd
    ifconfig
    

    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, tells us they soon will have a $20 a month platform specifically tailored to XiVO. And 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)

    Published: Monday, May 23, 2016





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


     
    Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For our users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage. Any balance is refundable if you decide to discontinue service with Vitelity.


    ​​3CX is a software PBX that’s easy to install & manage. It includes integrated softphones, WebRTC conferencing and essential add-ons out of the box, at no additional cost. Try the free edition at www.3cx.com.

  • Run on Premise or in the Cloud, on Windows and soon Linux
  • Softphones for iOS, Android, Win & Mac
  • Easy install, backup & restore, version upgrades
  • Automatically configures IP Phones, SIP Trunks & Gateways

  • Some Recent Nerd Vittles Articles of Interest…

    The XiVO Adventure Continues: Adding Incredible PBX Goodies to Your Sandbox

    We began our XiVO adventure last week by introducing a terrific new communications platform for both businesses and hobbyists. This week we begin the task of incorporating the Incredible PBX Goody Bag into an already amazing PBX, and we’ll cover about a dozen new topics. We’ll also address a few XiVO basics such as where to find and how to use the backups that XiVO makes every morning while many of us are still sleeping. Since a new XiVO release is imminent, we also want to show you how easy it is to upgrade your server. Before we get to the good stuff, we want to take a moment and document a fourth platform for XiVO that will appeal to many large organizations and perhaps some of our pioneers. It’s our platform of choice for development of new applications.

    Installing XiVO as a VMware Virtual Machine

    If your organization runs VMware, you may not need to worry about finding your own platform for XiVO. You can get your IT guys to build you a XiVO VM using XiVO’s Debian-based ISO. Then again, you might have followed our tutorial and chosen to run your own VMware ESXi server. In either case, a quick refresher on getting XiVO installed may be helpful. Begin by downloading XiVO to your Windows desktop. Then log into VMware vSphere Client on your Windows machine to access ESXi.

    First, you’ll want to upload the XiVO ISO as a VMware guest operating system so that it can be used to create virtual machines at any time. From your inventory, click on the Configuration tab. Then click Storage under the Hardware listing. When your Datastore appears, right-click on datastore1 and choose Browse Datastore. Finally, click the Upload Files to this Datastore icon in your Datastore Browser and choose Upload File option. Choose the XiVO ISO from the Upload Items menu to upload it into your Datastore.

    Now we’re ready to create a Virtual Machine. Right-click on the IP address of your VMware server and choose New Virtual Machine. Leave the Typical Configuration option selected and click Next. Give the virtual machine a name and click Next. Select the Destination Storage device and click Next. For the Operating System, choose Linux and pick Debian 8 (64-bit) then Next. Choose the NIC to use for the VM and click Next. Choose your Virtual Disk Size and Thin Provision option then Next. Check the box to Edit Virtual Machine Settings Before Completion and click Continue. Click the Options tab in Virtual Machine Properties and click Boot Options. Check the Force BIOS Setup option on next boot. Click Finish.

    Starting your virtual machine the first time is not exactly intuitive so follow these steps carefully and in order. Keep in mind that, on the initial bootup of your virtual machine, what we want to do is run the XiVO ISO installer just as if we had booted a standalone machine using a CD on which we had burned the XiVO ISO. To begin the boot process correctly, first highlight your new VM by clicking on it and then choose Power on Virtual Machine. Next, click on the CD/DVD icon in the toolbar, choose CD/DVD Drive 1, choose connect to ISO image on Datastore. Double-click on datastore1 and then double-click on the XiVO ISO we uploaded previously. Now click on Launch Virtual Machine Console icon in the toolbar. When the BIOS setup utility appears, click in the window and use the Right Arrow key to move to the Boot tab. Move the CD-ROM option to the top of the list by highlighting it and pressing the + key to move it up. Press F10 to Save and Exit from the BIOS Setup and boot into your XiVO ISO. Click Install option to begin the regular XiVO installation procedure. When you finish the install, log into your server as root and obtain your IP address: ifconfig. You then can exit from the Console window by pressing Ctrl-Alt and use a browser to complete the install by pointing to the IP address of your virtual machine. Don’t forget that root SSH access is disabled by default. Our original tutorial will show you how to fix it AND install the Travelin’ Man 3 firewall whitelist to protect your server.

    Adding a RingPlus SIP Trunk for Unified Communications with Sprint

    Last week we began the XiVO adventure by turning on free Google Voice calling in the U.S. and Canada. Today we want to integrate smartphones into the mix by providing an incredibly simple and dirt cheap way to expand your XiVO communications platform while transparently meshing it with a RingPlus smartphone and the Sprint cellular network. When we’re finished, calls to your smartphone will also ring on one or more XiVO extensions. And designated users of your XiVO PBX will be able to place free calls to U.S. destinations using a SIP trunk tied directly to your RingPlus cellular account. These calls won’t be cellular. They’ll be pure VoIP calls using Sprint’s Internet backbone so listen for that pin to drop. If you have a (free) unlimited calling plan with RingPlus, then you’ll inherit a (free) unlimited calling plan for your XiVO PBX. Stated another way, whatever calling minutes you have with RingPlus can be shared on your XiVO PBX as inbound and outbound VoIP calls. The silver lining is that voicemails left on RingPlus get transcribed and delivered to your email address in seconds. So you get the best of both worlds. That’s what Unified Communications is all about!

    Don’t worry if you’re late to the party and not yet a RingPlus user. They announce new deals every week so just check every few days until you find a plan that meets your needs. You won’t have to wait long. Here’s a list of all the previously announced PROMOS to give you a good handle on the scope of the RingPlus offerings. Deals don’t last but a couple hours or days so check often or sign up for RingPlus Alerts on SlickDeals and you’ll be the first to know! There’s a terrific deal tonight only from 8 p.m. until midnight.



    We’ve already documented the XiVO setup procedure on the PIAF Forum so hop over there to see how easy this is. Keep in mind that XiVO differs a bit from FreePBX® in the way Outbound Calls are managed. In FreePBX, you prioritized the routes by arranging them in a hierarchical list. In XiVO, you use unique dial strings, e.g. NXXNXXXXXX, for every Outbound Route. So, if you’re adding RingPlus to an existing XiVO server that already is using the NXXNXXXXXX dial string, then you’d need to use a different dial string to route calls out through the RingPlus trunk, e.g. 77NXXNXXXXXX with Stripnum=2. That tells XiVO that your users will dial calls to be handled by RingPlus with a prefix of 77 (RP), and then we want XiVO to strip off the first two digits before passing the call to the RingPlus SIP trunk for processing.

    If you’re new to RingPlus, start with the original Nerd Vittles article for some background and then follow the RingPlus threads on the PIAF Forum and DSL Reports for the latest tips and tricks.

    Adding a FreeVoipDeal (Betamax) SIP Trunk for Free International Calling

    Before deploying a SIP trunk from one of the Betamax companies, read our latest article about Betamax for tips and tricks and land mines to watch out for. Then click the link below when you’re ready to deploy FreeVoipDeal as a trunk on your XiVO PBX:


    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
    

    Update: There’s a great tip from one of the XiVO developers on a better way to do this. See the first comment below.

    Prerequisites for Today’s XiVO Adventure

    If you’re just getting started with XiVO, DON’T START HERE. Read our first article. Be sure you have completed the following 8 steps before proceeding:

    1. Set Up Root SSH Access to Your XiVO PBX
    2. Set Up the Travelin’ Man 3 IPtables Firewall Using an SSH/Putty Connection
    3. Complete the XiVO Setup Using a Web Browser
    4. Create At Least One User with a 701 Extension
    5. Create At Least One SIP Trunk to Use for Outbound Calls
    6. Configure Outbound Call Settings for Your Trunk Using NXXNXXXXXX
    7. Configure an Inbound Route for Trunk Pointing to Your User Account
    8. If Behind NAT Firewall, Set externip and local network in General Settings -> SIP Protocol -> Network

    Creating a MeetMe Conference Room for XiVO

    There are just two steps to setting up a conference room. First, you need to add the extensions you will use for your conferences in the Default context. Then you add the Conference Room under IPBX Settings. Let’s set up a conference room extension 2663 (C-O-N-F). In your Default context, click on the Conference Rooms tab and enter an extension range of 2663-2664 and click Save. Then, in the Conference Rooms tab, click the + icon to add a new CONF conference room at extension 2663 in the Default context. You can experiment with the other settings when you have some spare time. The entries are pretty much self-explanatory. Click Save to activate your conference room. You won’t have music on hold for the first participant just yet. We’ll do that next.

    Adding Music on Hold to XiVO

    By default, XiVO doesn’t come with any music on hold. Fortunately, Digium has negotiated a music on hold license that you can use to add it to your PBX at no cost. While logged into your XiVO PBX as root, issue the following commands:

    cd /
    wget http://incrediblepbx.com/moh-xivo.tar.gz
    tar zxvf moh-xivo.tar.gz
    /etc/init.d/asterisk restart
    

    Asterisk Application Development with XiVO

    For those coming from the FreePBX world, here’s a quick introduction to Asterisk application development on the XiVO platform. First and foremost, there are more similarities than differences. In the FreePBX environment, custom dialplan code was stored in /etc/asterisk/extensions_custom.conf. For custom extensions that you wanted to add, that code had to appear in the [from-internal-custom] context. For custom dialplan contexts, those appeared immediately below the last entry in the [from-internal-custom] context. If your custom code appeared anywhere else, there was always the risk that it might be overwritten with your next FreePBX reload.

    The XiVO design is quite different. As we noted last week, it is not an Asterisk code generator at all, unlike FreePBX. Instead, it has a realtime interface to Asterisk using its PostGreSQL database engine. Updates are nearly instantaneous without reloading Asterisk modules from disk.



    The other advantage is you won’t have to worry about XiVO stepping on your custom code as long as you leave PostGreSQL alone. HINT! The good news is there still are hooks to add your own custom dialplan extensions and code as well as PHP/AGI scripts. And it’s easy. In XiVO, custom extensions are stored in xivo-extrafeatures.conf which you’ll find in the /etc/asterisk/extensions_extra.d directory. Don’t edit files in /etc/asterisk/extensions_extra.d from the Linux command prompt! Instead, use the editor built into the XiVO GUI by selecting Configuration Files under IPBX configuration. This will automatically assure that realtime updates are posted correctly. To add additional contexts to your dialplan, create separate files for each context and store them in this same directory. Again, the easy way to make certain that Asterisk is updated automatically when you add new code snippets is to create and edit them within the XiVO GUI. These files all will appear under IPBX Configuration -> Configuration Files as well.

    In order to better mimic the FreePBX way of doing things so that your PHP/AGI scripts work in either environment, we recommend issuing the following symlink while logged into XiVO. We’ll do it as part of the SQLite3 install below.

    ln -s /var/lib/asterisk/agi-bin /usr/share/asterisk/agi-bin
    

    Once you’ve established the symlink, PHP/AGI scripts can be migrated from FreePBX to XiVO directly using the same directory structure for storage: /var/lib/asterisk/agi-bin. As with FreePBX, all files in this directory should be owned by asterisk with 775 permissions:

    chown asterisk:asterisk /var/lib/asterisk/agi-bin/*
    chmod 775 /var/lib/asterisk/agi-bin/*
    

    There are many other powerful features in XiVO that weren’t available at all in FreePBX. We’ll cover some of them in coming months. In the meantime, this brief overview of the dialplan environment should be sufficient to let you start building.

    Installing SQLite3 to Support Incredible PBX Applications

    There’s one other difference between XiVO and FreePBX that we’ve already touched upon. But it bears repeating here. XiVO doesn’t use MySQL or MariaDB for its database management tasks. Instead, the XiVO development team chose PostGreSQL which is equally powerful, but different. For the Incredible PBX application suite, we’ve chosen to rewrite the ones that depend upon MySQL so that they can run under SQLite3 which is considerably less processor intensive than running both PostGreSQL and MySQL 24/7. We also didn’t want to interfere with the PostGreSQL setup of XiVO since it is an integral component of the product and will get upgraded automatically as part of the regular XiVO upgrade cycle.

    Here’s how to put the SQLite3 and corresponding ODBC components in place on your new server. While logged into your server as root, simply issue the following commands:

    cd /
    wget http://incrediblepbx.com/sqlite3-xivo.tar.gz
    tar zxvf sqlite3-xivo*
    rm -f sqlite3-xivo.tar.gz
    cd /root
    ./sqlite3-xivo.sh
    

    Running a couple SQLite3 queries using the ZIPCODES and ASTERIDEX databases will give you a feel for the performance you can expect from SQLite3. The queries might look like this:

    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%";'
    

    And here are the results of the two queries:

    29401|CHARLESTON|SC
    --------------------------------
    American Airlines|8004337300
    Continental Airlines|8005250280
    Delta AirLines|8002211212
    Frontier Airlines|8004321359
    Iberia AirLines|8007724642
    Midway Airlines|8004464392
    Northwest Airlines|8002252525
    Southwest Airlines|8004359792
    Ted Airlines|8002255833
    United Airlines|8002416522
    WestJet Airlines|8005385696
    Yemen Airlines|8009368300
    

    We’ve included a bonus script in /root that 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 should look like this:

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

    You obviously cannot run the script on your XiVO server because your MySQL databases and MySQL itself are missing. So move the script to the server on which your 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.

    Installing and Activating the Festival TTS Engine with Asterisk

    We’ve got a couple more building blocks to put in place to support Incredible PBX applications. Then we’ll be ready to kick the tires with a few applications to get you started. In coming weeks, we’ll finish up the conversion of the remaining apps, and then we’ll publish an Incredible PBX installer for XiVO with all the pieces. But why wait? Finish up installing the remaining pieces today, and you’ll have something to play with. And, as we said, it will also provide you with simple scripts so you can actually see how Incredible PBX is put together.

    Many of the Incredible PBX applications rely upon text-to-speech and/or voice recognition (speech-to-text) to work their magic. Neither comes installed with XiVO by default, but Asterisk was properly configured to support Festival so let’s work with that. Festival is the Big Brother of FLITE and includes some additional voices of fairly good quality. The XiVO Demo IVR will give you an idea of the TTS voice quality you can expect:

    To get Festival installed and activated for use with Asterisk, issue these commands:

    cd /
    wget http://incrediblepbx.com/festival-xivo.tar.gz
    tar zxvf festival-xivo.tar.gz
    cd /root
    ./festival-xivo.sh
    

    Installing Dial Plan Code for Sample Incredible PBX Applications

    Now we’re ready to put today’s Dial Plan Code and IVR in place and load the PHP/AGI components necessary to make the sample applications work. Here’s how:

    cd /
    wget http://incrediblepbx.com/ivr-xivo.tar.gz
    tar zxvf ivr-xivo.tar.gz
    chown asterisk:www-data /etc/asterisk/extensions_extra.d
    chmod 775 /etc/asterisk/extensions_extra.d
    chmod g+s /etc/asterisk/extensions_extra.d
    /etc/init.d/asterisk restart
    

    Installing and 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.

    Taking XiVO on a Test Drive with the Incredible PBX Apps

    Now set up a softphone using the IP address of your XiVO server and the Line credentials for Extension 701. When you obtain your credentials, double-check to make sure all of the fields for the Line are filled in correctly as shown below:

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

    • 4871 (IVR1) – Allison’s Demo IVR
    • 411 (Voice Dialing) – Call by Name (try “Delta Airlines”)
    • 2663 (CONF) – MeetMe Conference 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 Configuration Files and clicking xivo-extrafeatures.conf. The sample IVR code is in ivr-1.conf.

    Taking Nerd Vittles’ XiVO IVR for a Test Drive

    There’s also a new Demo IVR running at www.pacificnx.com on their XenServer virtualization platform. Scott McCarthy, a leading outside XiVO developer and a principal at PacificNX, tells us they soon will have a $20 a month platform specifically tailored to XiVO. And that’s what you’ll be hearing when you call the Nerd Vittles IVR: 1-843-606-0555. Setup at PacificNX took less than a minute. Enjoy!

    Published: Thursday, May 12, 2016





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


     
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  • Softphones for iOS, Android, Win & Mac
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  • Automatically configures IP Phones, SIP Trunks & Gateways

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