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The Most Versatile VoIP Provider: FREE PORTING
JUST RELEASED: Visit the Incredible PBX WikiThe Next Best Thing to (formerly free) Google Voice
Today we want to once again shine the spotlight on LocalPhone, an oft-overlooked VoIP service that’s been around forever. You can call to and be called from any LocalPhone user at no cost. They also offer phone numbers (DIDs) of your choice almost anywhere in the world with free or almost free incoming calls. For those wanting a U.S. DID, the cost is 99¢ a month with a $3 setup fee. That gets you up to 100 free incoming calls a day to your PBX or any SIP phone. Additional calls are a penny per call. There are no limitations on the duration of the calls. If you prefer to forward the calls to your cellphone number in the contiguous U.S., there’s an additional fee of 0.5¢ per minute. But there’s little reason to do that when sending the calls to a SIP softphone on your Android device or iPhone is free. And now the mobile LocalPhone app supports PUSH Notifications. We’ll show you how.
FYI: Nerd Vittles receives a referral credit to keep the lights on when you sign up for service.
Deciphering Your SIP Credentials with LocalPhone
Once you have signed up for a LocalPhone account, the first thing you’ll want to do is make note of your Internet Phone credentials under My Account. These are what we typically refer to as SIP credentials consisting of a SIP ID, SIP password, and SIP server (localphone.com). That’s all you’ll need to configure an incoming LocalPhone trunk on any Incredible PBX® server. And these are the same settings you’d use to configure any SIP phone running on any Android or iOS device. As we noted, you and any other LocalPhone user can call any Internet Phone number worldwide at no cost without limitation. For world travelers, you’ll want to download the LocalPhone app for your smartphone (Android or iOS) and take advantage of their extremely competitive international calling rates.1
Ordering Incoming Numbers (DIDs) from LocalPhone
Begin by funding your account under My Account -> Add Credit. $10 will last you a long time.
The next step is to order one or more incoming phone numbers from LocalPhone.2 If you have friends in far away places that call you frequently, you can purchase DIDs in those locations to eliminate the cost of incoming calls both to them and to you. If you only want a dirt cheap U.S. DID for your home or small office, then LocalPhone is also a perfect fit. Navigate to My Account -> Incoming Numbers and choose the United States as the desired Country. Next, pick the State and City for the desired DID. For free incoming calls, set Call Forwarding and Caller ID for Internet Phone to your assigned Internet Phone SIP ID. You can also elect to forward calls to a SIP URI, if desired. Agree to the terms of use and make your purchase.
Configuring a LocalPhone Trunk with Incredible PBX
We’ve previously covered the LocalPhone trunk setup with Wazo. Most other releases of Incredible PBX include preconfigured LocalPhone trunks for incoming and outgoing calls. Login to the Incredible PBX GUI as admin using your favorite browser and navigate to Connectivity -> Trunks and edit the LocalPhone-In trunk. Set Disable Trunk to NO. Then click the sip-Settings tab. Insert your LocalPhone SIP ID in the username, fromuser, and authuser fields. Insert your LocalPhone SIP Password in the secret field. Change the context field entry to from-trunk. Click on the Incoming tab, and modify the Register String 9999999:yourpassword@localphone.com/9999999 replacing 9999999 with your LocalPhone SIP ID and yourpassword with your LocalPhone SIP Password. Click the Submit button and reload your dialplan when prompted.
Navigate to Connectivity -> Inbound Routes and create a new Inbound Route for LocalPhone using your SIP ID as the DID Number and choosing a desired Destination for incoming calls from your LocalPhone DID. Save your settings and reload the dialplan when prompted.
Configuring a LocalPhone Trunk with VitalPBX
Login to the VitalPBX GUI as admin using your favorite browser and navigate to PBX -> External -> Trunks. Create a new SIP trunk with the following settings replacing 999999 with your LocalPhone SIP ID and 1234 with the LocalPhone SIP Password. Leave the Device for Incoming Calls (User) section blank. Then click SAVE and reload your dialplan.
- Description: LocalPhone
- Codecs: ulaw,alaw
- Local Username: 999999
- Remote Host: localphone.com
- Remote Port: 5060
- Local Secret: 1234
- Insecure: Port,Invite
- Allow Inbound Calls: YES
- Username: [leave blank]
- Host: [leave blank]
- Local Secret: [leave blank]
- Remote Username: 999999
- Remote Secret: 1234
- From User: 999999
- From Domain: localphone.com
- Qualify: YES
- Insecure: [leave blank]
- IP Authentication: NO
- Qualify: [leave default]
- Register String: 999999:1234@localphone.com/999999
Navigate to PBX -> External -> Inbound Routes. Create a new Inbound Route for LocalPhone using your SIP ID as the DID Number and choosing a desired Destination for incoming calls from your LocalPhone DID. Save your settings and reload the dialplan when prompted.
Configuring a LocalPhone Trunk with FreePBX
Login to the FreePBX® GUI as admin using your favorite browser and navigate to Connectivity -> Trunks. Add a new chan_sip trunk named localphone. Then click on the sipSettings tab and enter the following replacing 999999 with your LocalPhone SIP ID and 1234 with the LocalPhone SIP Password.
- username=9999999
- type=friend
- secret=1234
- nat=no
- insecure=port,invite
- host=localphone.com
- fromuser=9999999
- fromdomain=localphone.com
- dtmfmode=rfc2833
- disallow=all
- context=from-trunk
- canreinvite=no
- authuser=9999999
- allow=ulaw&alaw
Next, click on the Incoming tab and enter the following Register String replacing 999999 with your LocalPhone SIP ID and 1234 with the LocalPhone SIP Password:
9999999:1234@localphone.com/9999999
Then click SUBMIT and reload your dialplan.
Navigate to Connectivity -> Inbound Routes and create a new Inbound Route for LocalPhone using your SIP ID as the DID Number and choosing a desired Destination for incoming calls from your LocalPhone DID. Save your settings and reload the dialplan when prompted.
Using Local Numbers for International Calls
LocalPhone has a unique feature that lets you dial a local number from a phone number you have whitelisted in your country and reach almost anyone in the world that you’ve added to your Contacts List. You only pay LocalPhone’s discounted international calling rate for the calls. For example, to call a landline in the U.K. from the U.S. using a LocalPhone-provided U.S. phone number, the calling rate is less than a penny a minute. A call to Cyprus by dialing a U.S. number assigned to your account for your whitelisted phone numbers is 4.5 cents per minute. To get started setting up your whitelisted phone numbers and contacts list, navigate to My Account -> Local Numbers in your LocalPhone account. In your Local Numbers list, first add and verify phone numbers you want to authorize to make calls on your nickel. Next, add the names and phone numbers of international destinations you wish to reach by dialing a local number. LocalPhone will immediately assign a local number for each destination. Simply add these local numbers to the contacts list on your smartphone, and you can call from anywhere in your country at the discounted LocalPhone international calling rates. There are no double-dialing or call menus to navigate. Dialing the assigned local number transparently connects you directly to your destination with no intermediate hurdles.
Using LocalPhone with Other Trunk Providers
So long as your PBX doesn’t have more than two incoming calls to a single DID at the same time, the most economical PBX design is to use LocalPhone DIDs as your published DIDs. This reduces the cost of incoming calls to less than a dollar a month per DID for up to 3,000 incoming calls of unlimited duration. Then use one of our Platinum Sponsors, Skyetel or our soon-to-be-available ClearlyIP SIP trunking service for outbound calls and spoof the outbound CallerID on those other trunks using your LocalPhone DID.
Enjoying the Best of All Worlds with LocalPhone
If you have an iPhone or Android smartphone in addition to a PBX, you can take advantage of LocalPhone’s ability to send incoming calls to multiple destinations. Just make sure your PBX isn’t routing the incoming calls to a destination that is automatically answered, e.g. an IVR. On your Android phone, download the VitalPBX Communicator from the Google Play Store and configure a SIP connection using your LocalPhone SIP credentials. Incoming calls from your LocalPhone DIDs and Internet Phone Number now will be sent to both destinations.
If you have followed one of our previous tutorials that document making SIP URI calls from either a PBX or a SIP client such as LinPhone on your smartphone, then you can take advantage of LocalPhone’s incoming SIP URI feature.3 Just dial 9999999@localphone.com where 9999999 is any LocalPhone SIP ID. You also can add Custom Extensions in Incredible PBX much like the Lenny extension using a Dial string of SIP/9999999@localphone.com to reach worldwide LocalPhone destinations from any PBX extension at no cost. Enjoy!
Originally published: Monday, December 9, 2019
Need help with Asterisk? Visit the VoIP-info Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.- Rates are based on the lowest pay as you go per-minute price to call a landline or a mobile. Skype is a registered trademark of Microsoft Corporation. [↩]
- LocalPhone advises that DID fulfillment can take up to 14 days although our orders always have been completed in less than an hour. [↩]
- LocalPhone offers call filtering for your Internet Phone number using either a blacklist or whitelist in addition to offering the option of blocking anonymous calls. [↩]
A New VPN for All Seasons: Introducing OpenVPN for Asterisk
This month marks our twentieth anniversary wrestling with virtual private networks. Here’s a quick walk down memory lane. Our adventure began with the Altiga 3000 series VPN concentrators which we introduced in the federal courts in 1999. It was a near perfect plug-and-play hardware solution for secure communications between remote sites using less than secure Windows PCs. Cisco quickly saw the potential, gobbled up the company, and promptly doubled the price of the rebranded concentrators. About 10 years ago, we introduced Hamachi® VPNs to interconnect Asterisk® and PBX in a Flash servers. At the time, Hamachi was free, but that was short-lived when they were subsequently acquired by LogMeIn®. What followed was a short stint with PPTP VPNs which worked great with Macs, Windows PCs, and many phones but suffered from an endless stream of security vulnerabilities. Finally, in April 2012, we introduced the free NeoRouter® VPN. Version 2 still is an integral component in every Incredible PBX® platform today, and PPTP still is available as well. While easy to set up and integrate into multi-site Asterisk deployments, the Achilles’ Heel of NeoRouter remains its inability to directly interconnect many smartphones and stand-alone SIP phones, some of which support the OpenVPN platform and nothing else.
The main reason we avoided OpenVPN® over the years was its complexity to configure and deploy.1 In addition, it was difficult to use with clients whose IP addresses were frequently changing. Thanks to the terrific work of Nyr, Stanislas Angristan, and more than a dozen contributors, OpenVPN now has been tamed. And the new server-based, star topology design makes it easy to deploy for those with changing or dynamic IP addresses. Today we’ll walk you through building an OpenVPN server as well as the one-minute client setup for almost any Asterisk deployment and most PCs, routers, smartphones, and VPN-compatible soft phones and SIP phones including Yealink, Grandstream, Snom, and many more. And the really great news is that OpenVPN clients can coexist with your current NeoRouter VPN.
Finally, a word about the OpenVPN Client installations below. We’ve tested all of these with current versions of Incredible PBX 13-13, 16-15, and Incredible PBX 2020. They should work equally well with other server platforms which have been properly configured. However, missing dependencies on other platforms are, of course, your responsibility.
Building an OpenVPN Server Platform
There are many ways to create an OpenVPN server platform. The major prerequisites are a supported operating system, a static IP address for your server, and a platform that is extremely reliable and always available. If the server is off line, all client connections will also fail. While we obviously have not tested all the permutations and combinations, we have identified a platform that just works™. It’s the CentOS 7, 64-bit cloud offering from Vultr. If you use our referral link at Vultr, you not only will be supporting Nerd Vittles through referral revenue, but you also will be able to take advantage of their $50 free credit for new customers. For home and small business deployments, we have found the $5/month platform more than adequate, and you can add automatic backups for an additional $1 a month. Cheap insurance!
To get started, create your CentOS 7 Vultr instance and login as root using SSH or Putty. Immediately change your password and update and install the necessary CentOS 7 packages:
passwd yum -y update yum -y install net-tools nano wget tar iptables-services systemctl stop firewalld systemctl disable firewalld systemctl enable iptables
We recommend keeping your OpenVPN server platform as barebones as possible to reduce the vulnerability risk. By default, this installer routes all client traffic through the VPN server which wastes considerable bandwidth. The sed commands below modify this design to only route client VPN traffic through the OpenVPN server.
cd /root curl -O https://raw.githubusercontent.com/Angristan/openvpn-install/master/openvpn-install.sh chmod +x openvpn-install.sh sed -i "s|\\techo 'push \\"redirect-gateway|#\\techo 'push \\"redirect-gateway|" openvpn-install.sh sed -i "s|push \\"redirect-gateway|#push \\"redirect-gateway|" openvpn-install.sh sed -i 's|tls-client|tls-client\\npull-filter ignore "redirect-gateway"|' openvpn-install.sh ./openvpn-install.sh
Here are the recommended entries in running the OpenVPN installer:
- Server IP Address: using FQDN strongly recommended to ease migration issues
- Enabled IPv6 (no): accept default
- Port (1194): accept default
- Protocol (UDP): accept default
- DNS (3): change to 9 (Google)
- Compression (no): accept default
- Custom encrypt(no): accept default
- Generate Server
- Client name: firstclient
- Passwordless (1): accept default
In the following steps, we will use IPtables to block all server access except via SSH or the VPN tunnel. Then we’ll start your OpenVPN server:
cd /etc/sysconfig wget http://incrediblepbx.com/iptables-openvpn.tar.gz tar zxvf iptables-openvpn.tar.gz rm -f iptables-openvpn.tar.gz echo "net.ipv4.ip_forward = 1" >> /etc/sysctl.conf sysctl -p systemctl -f enable openvpn@server.service systemctl start openvpn@server.service systemctl status openvpn@server.service systemctl enable openvpn@server.service systemctl restart iptables
Once OpenVPN is enabled, the server can be reached through the VPN at 10.8.0.1. OpenVPN clients will be assigned by DHCP in the range of 10.8.0.2 through 10.8.0.254. You can list your VPN clients like this: cat /etc/openvpn/ipp.txt. You can list active VPN clients like this: cat /var/log/openvpn/status.log | grep 10.8. And you can add new clients or delete old ones by rerunning /root/openvpn-install.sh.
For better security, change the SSH access port replacing 1234 with desired port number:
PORT=1234 sed -i "s|#Port 22|Port $PORT|" /etc/ssh/sshd_config systemctl restart sshd sed -i "s|dport 22|dport $PORT|" /etc/sysconfig/iptables systemctl restart iptables
04/16 UPDATE: We’ve made changes in the Angristan script to adjust client routing. By default, all packets from every client flowed through the OpenVPN server which wasted considerable bandwidth. Our preference is to route client packets destined for the Internet directly to their destination rather than through the OpenVPN server. The sed commands added to the base install above do this; however, if you’ve already installed and run the original Angristan script, your existing clients will be configured differently. Our recommendation is to remove the existing clients, make the change below, and then recreate the clients again by rerunning the script. In the alternative, you can execute the command below to correct future client creations and then run it again on each existing client platform substituting the name of the /root/.ovpn client file for client-template.txt and then restart each OpenVPN client.
cd /etc/openvpn sed -i 's|tls-client|tls-client\\npull-filter ignore "redirect-gateway"|' client-template.txt
Creating OpenVPN Client Templates
In order to assign different private IP addresses to each of your OpenVPN client machines, you’ll need to create a separate client template for each computer. You do this by running /root/openvpn-install.sh again on the OpenVPN server. Choose option 1 to create a new .ovpn template. Give each client machine template a unique name and do NOT require a password for the template. Unless the client machine is running Windows, edit the new .ovpn template and comment out the setenv line: #setenv. Save the file and copy it to the /root folder of the client machine. Follow the instructions below to set up OpenVPN on the client machine and before starting up OpenVPN replace firstclient.ovpn in the command line with the name of .ovpn you created for the individual machine.
Renewing OpenVPN Server’s Expired Certificate
The server certificate will expire after 1080 days, and clients will no longer be able to connect. Here’s what to do next:
systemctl stop openvpn@server.service cd /etc/openvpn/easy-rsa ./easyrsa gen-crl cp /etc/openvpn/easy-rsa/pki/crl.pem /etc/openvpn/crl.pem systemctl start openvpn@server.service
Installing an OpenVPN Client on CentOS/RHEL
cd /root yum -y install epel-release yum --enablerepo=epel install openvpn -y # copy /root/firstclient.ovpn from server to client /root # and then start up the VPN client openvpn --config /root/firstclient.ovpn --daemon # adjust Incredible PBX 13-13 firewall below iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT cd /usr/local/sbin echo "iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT" >> iptables-custom
Running ifconfig should now show the VPN client in the list of network ports:
tun0 Link encap:UNSPEC HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
inet addr:10.8.0.2 P-t-P:10.8.0.2 Mask:255.255.255.0
UP POINTOPOINT RUNNING NOARP MULTICAST MTU:1500 Metric:1
RX packets:9 errors:0 dropped:0 overruns:0 frame:0
TX packets:39 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:100
RX bytes:855 (855.0 b) TX bytes:17254 (16.8 KiB)
And you should be able to login to the VPN server using its VPN IP address:
# enter actual SSH port replacing 1234 PORT=1234 ssh -p $PORT root@10.8.0.1
Installing an OpenVPN Client on Ubuntu 18.04.2
cd /root apt-get update apt-get install openvpn unzip dpkg-reconfigure tzdata # copy /root/firstclient.ovpn from server to client /root # and then start up the VPN client openvpn --config /root/firstclient.ovpn --daemon # adjust Incredible PBX 13-13 firewall below iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT cd /usr/local/sbin echo "iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT" >> iptables-custom
Running ifconfig should now show the VPN client in the list of network ports:
tun0 Link encap:UNSPEC HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
inet addr:10.8.0.2 P-t-P:10.8.0.2 Mask:255.255.255.0
UP POINTOPOINT RUNNING NOARP MULTICAST MTU:1500 Metric:1
RX packets:9 errors:0 dropped:0 overruns:0 frame:0
TX packets:39 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:100
RX bytes:855 (855.0 b) TX bytes:17254 (16.8 KiB)
And you should be able to login to the VPN server using its VPN IP address:
# enter actual SSH port replacing 1234 PORT=1234 ssh -p $PORT root@10.8.0.1
Installing an OpenVPN Client on Raspbian
Good news and bad news. First the bad news. Today’s OpenVPN server won’t work because of numerous unavailable encryption modules on the Raspberry Pi side. The good news is that NeoRouter is a perfect fit with Raspbian, and our upcoming article will show you how to securely interconnect a Raspberry Pi with any Asterisk server in the world… at no cost.
04/16 Update: We now have OpenVPN working with Incredible PBX for the Raspberry Pi. The trick is that you’ll need to build the latest version of OpenVPN from source before beginning the client install. Here’s how. Login to your Raspberry Pi as root and issue these commands:
apt-get remove openvpn apt-get update apt-get install libssl-dev liblzo2-dev libpam0g-dev build-essential -y cd /usr/src wget https://swupdate.openvpn.org/community/releases/openvpn-2.4.7.tar.gz tar zxvf openvpn-2.4.7.tar.gz cd openvpn-2.4.7 ./configure --prefix=/usr make make install openvpn --version
Now you should be ready to install a client config file, start up OpenVPN, and adjust firewall:
cd /root dpkg-reconfigure tzdata # copy /root/firstclient.ovpn from server to client /root # and then start up the VPN client openvpn --config /root/firstclient.ovpn --daemon # adjust Incredible PBX 13-13 firewall below iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT cd /usr/local/sbin echo "iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT" >> iptables-custom
Installing an OpenVPN Client on a Mac
While there are numerous OpenVPN clients for Mac OS X, none hold a candle to Tunnelblick in terms of ease of installation and use. First, create a new client config on your server and copy it (/root/*.ovpn) to a folder on your Mac where you can find it. Download Tunnelblick and install it. Run Tunnelblick and then open Finder. Click and drag your client config file to the Tunnelblick icon in the top toolbar. Choose Connect when prompted. Done.
Installing an OpenVPN Client for Windows 10
The installation procedure for Windows is similar to the Mac procedure above. Download the OpenVPN Client for Windows. Double-click on the downloaded file to install it. Create a new client config on your server and copy it (/root/*.ovpn) to a folder on your PC where you can find it. Start up the OpenVPN client and click on the OpenVPN client in the activity tray. Choose Import File and select the config file you downloaded from your OpenVPN Server. Right-click on the OpenVPN icon again and choose Connect. Done.
Installing an OpenVPN Client for Android
Our favorite OpenVPN client for Android is called OpenVPN for Android and is available in the Google Play Store. Download and install it as you would any other Android app. Upload a client config file from your OpenVPN server to your Google Drive. Run the app and click + to install a new profile. Navigate to your Google Drive and select the config file you uploaded.
Installing an OpenVPN Client for iOS Devices
The OpenVPN Connect client for iOS is available in the App Store. Download and install it as you would any other iOS app. Before uploading a client config file, open the OpenVPN Connect app and click the 4-bar Settings icon in the upper left corner of the screen. Click Settings and change the VPN Protocol to UDP and IPv6 to IPV4-ONLY Tunnel. Accept remaining defaults.
To upload a client config file, the easiest way is to use Gmail to send yourself an email with the config file as an attachment. Open the message with the Gmail app on your iPhone or iPad and click on the attachment. Then choose the Upload icon in the upper right corner of the dialog. Next, choose Copy to OpenVPN in the list of apps displayed. When the import listing displays in OpenVPN Connect, click Add to import the new profile. Click ADD again when the Profile has been successfully imported. You’ll be prompted for permission to Add VPN Configurations. Click Allow. Enter your iOS passcode when prompted. To connect, tap once on the OpenVPN Profile. To disconnect, tap on the Connected slider. When you reopen the OpenVPN Connect app, the OVPN Profiles menu will display by default. Simply tap once on your profile to connect thereafter.
Installing a Web Interface to Display Available Clients
One advantage of NeoRouter is a simple way for any VPN client to display a listing of all VPN clients that are online at any given time. While that’s not possible with OpenVPN, we can do the next best thing and create a simple web page that can be accessed using a browser but only from a connected OpenVPN client pointing to http://10.8.0.1.
To set this up, log in to your OpenVPN server as root and issue the following commands:
yum --enablerepo=epel install lighttpd -y systemctl start lighttpd.service systemctl enable lighttpd.service chown root:lighttpd /var/log/openvpn/status.log chmod 640 /var/log/openvpn/status.log cd /var/www rm -rf lighttpd wget http://incrediblepbx.com/lighttpd.tar.gz tar zxvf lighttpd.tar.gz ln -s /var/log/openvpn/status.log /var/www/lighttpd/status.log sed -i 's|#server.bind = "localhost"|server.bind = "10.8.0.1"|' /etc/lighttpd/lighttpd.conf systemctl restart lighttpd.service
Latest VPN Security Alerts
https://nakedsecurity.sophos.com/2019/04/16/security-weakness-in-popular-vpn-clients/
Originally published: Monday, April 15, 2019 Updated: Saturday, February 29, 2020
Need help with Asterisk? Visit the VoIP-info Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
- Our discussion today is focused on the free, MIT-licensed version of OpenVPN. For details on their commercial offerings, follow this link. [↩]
R.I.P. GVSIP: A Final Farewell to Google Voice
It’s been a death by a thousand cuts, but today marks the end of the Google Voice era with Asterisk®. Since Google removed XMPP support and transitioned to their new GVSIP platform, many have held out hope that Google hadn’t moved to a purely commercial platform with their ObiHai deal. Yesterday, the head of the Google Voice project requested that all Asterisk GVSIP implementations be discontinued citing Google’s Terms of Service. We hinted this was coming back in July and have reproduced our tweet below. We have since removed all of our articles pertaining to GVSIP, and we would encourage all of our readers to honor Google’s wishes and move on. We’ve made it easy with a $50 gift certificate from Skyetel (expires March 31, 2019). It will buy you many months of free VoIP service.
You still have several options with your Google Voice trunks. First, you can forward all incoming calls to Google Voice to another phone or DID of your choosing. This costs you nothing other than a minute to set it up. Second, you can port out your Google Voice number to another provider. Skyetel will cover your porting expense at their end during your first 60 days of service. Google charges $3 to port out your number unless you originally ported it into Google in which case it is free. Here’s how. Although we’re not big fans, a third option is to purchase an OBi200 device and continue to use your Google Voice trunk with Asterisk. Our tutorial from last May will show you how. Effective 10/1/2023, $25/month minimum spend at Skyetel is required.
As we’ve mentioned often, the beauty of VoIP is not having to put all of your telephony eggs in a single basket. Google’s latest move reinforces how important it actually is to configure several VoIP trunks on your server. While Skyetel and Vitelity are both excellent primary trunks and rarely experience an outage, it’s still a good idea to have a backup. VoIP.ms (free iNUM), CircleNet, CallCentric ($1/mo. DID and iNUM), LocalPhone (25¢/mo. iNUM), Future-Nine, AnveoDirect, and V1VoIP are excellent options. Most don’t cost you anything unless you make calls. Review our complete SIP tutorial here: Developing a Cost-Effective SIP Strategy.
Dale Carnegie Award: ObiHai Man of the Year
Originally published: Friday, November 16, 2018
Need help with Asterisk? Join our new MeWe Support Site.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Free Asterisk Voicemail Transcription with IBM Watson STT
There are many commercial voicemail transcription services for Asterisk® PBXs, but none hold a candle to the speech-to-text (STT) quality of the IBM Cloud offering known as Watson® STT, formerly known as Bluemix TTS. Despite a recent price increase that takes effect in December, the pricing remains competitive. On the Standard Pricing Plan, voicemail transcription is 2¢ per minute. Or you can try things out on the LITE plan which offers 100 minutes a month at no cost. When the messages are delivered by email, you get the voicemail recording in MP3 format AND transcribed text courtesy of Watson TTS. With IBM services, there no longer are username:password credentials. Instead, you will have a new apikey.
Those with existing configurations can update your credentials by inserting a new apikey using the following commands, or you can simply insert apikey as your $API_USERNAME and enter your actual APIkey as your $API_PASSWORD.
cd /usr/local/sbin sed -i 's|$API_USERNAME:$API_PASSWORD|"apikey:x-yy-zzz"|' sendmailmp3 sed -i 's|$API_USERNAME:$API_PASSWORD|"apikey:x-yy-zzz"|' bluemix-test
IBM Cloud’s STT solution is a real game-changer for one simple reason. Their STT API performs more accurately than any speech recognition engine in the world. As an added bonus, you won’t have to worry about Google breaking our middleware every month. It’s worth noting that IBM doesn’t round up minutes. Transcribing two 30-second messages counts as one minute.
https://youtu.be/JWnLgZ58zsw
Overview. What we’ve done today is integrate the Watson STT API directly into existing Asterisk voicemail systems. We started with Nicolas Bernaerts’ terrific sendmailmp3 script. It works on both the Wazo and FreePBX® platforms. If you have deployed Incredible PBX®, then the setup takes a couple of minutes. For everyone else, there’s an additional configuration step using your favorite GUI. To get started, you’ll sign up for an IBM Cloud account and obtain your credentials. Next, you download today’s script for your platform and insert your credentials. Finally, you set up voicemail on the extensions desired and insert an email address for each voicemail account. On generic FreePBX systems, you’ll need to add the name of our script to manage your voicemail recordings. And, regardless of your PBX platform, you obviously need outgoing SMTP email working reliably.
Start by sending yourself a test email and get that working first:
echo "test" | mail -s testmessage yourname@your-email-domain.com
What About the Quality? Here’s the bottom line. Speech recognition isn’t all that useful if it fails miserably in recognizing everyday speech. The good news is that IBM Watson’s speech recognition engine is now the best in the business. If you want more details, read the article below which will walk you through IBM’s latest speech recognition breakthrough:
Why IBM's speech recognition breakthrough matters for AI and IoT. Via @techrepublic https://t.co/AJi8MA3E20
— IBM Developer (@IBMDeveloper) March 15, 2017
Obtaining IBM Cloud Speech to Text Credentials
Follow this link to set up your IBM account and obtain credentials for both Speech to Text (STT) and Text to Speech (TTS) services. Please note that your STT and TTS API keys will NOT be the same. So don’t accidentally use the wrong one.
Installing STT Engine with Incredible PBX for Wazo
1. After logging into your Incredible PBX for Wazo server as root using SSH/Putty:
cd /usr/sbin wget http://incrediblepbx.com/sendmailibm.tar.gz tar zxvf sendmailibm.tar.gz rm -f sendmailibm.tar.gz
2. Edit sendmailibm and insert IBM STT API_KEY and URL.
3. Edit bluemix-test and insert IBM STT API_KEY and URL.
4. Apply the patch documented above if using LITE plan using sendmail filename instead of sendmailmp3.
5. Copy the updated sendmailibm file to sendmail:
cd /usr/sbin cp -p sendmailibm sendmail
6. Test your Bluemix STT setup: bluemix-test
7. Result should be: please record your message after the beep
8. Set up voicemail account for a Wazo extension with your email address.
9. Place a test call to the extension and record a voicemail when prompted.
10. Your message will be transcribed and delivered via email.
Installing STT Engine with Incredible PBX for RasPi
1. After logging into your Raspberry Pi server as root using SSH/Putty:
cd /usr/sbin wget http://incrediblepbx.com/sendmailibm-raspi.tar.gz tar zxvf sendmailibm-raspi.tar.gz rm -f sendmailibm-raspi.tar.gz
2. Edit sendmailmp3.ibm and insert your Bluemix STT API_KEY and URL. Save file.
3. Edit bluemix-test and insert your Bluemix STT API_KEY and URL. Save the file.
4. Copy the updated sendmailmp3.ibm file to sendmailmp3:
cd /usr/sbin cp -p sendmailmp3.ibm sendmailmp3
5. Apply the patch documented above if using LITE plan.
6. Test your Bluemix STT setup: bluemix-test
7. Result should be: your dictation is now being processed and emailed please wait
8. Set up voicemail for a RasPi extension with your email address.
9. Place a test call to the extension and record a voicemail when prompted.
10. Your message will be transcribed and delivered via email.
Installing STT Engine with Incredible PBX 13-13
1. After logging into your Incredible PBX 13 server as root using SSH/Putty:
cd /usr/local/sbin wget http://incrediblepbx.com/sendmailibm-13.tar.gz tar zxvf sendmailibm-13.tar.gz rm -f sendmailibm-13.tar.gz
2. Edit sendmailmp3.ibm and insert your IBM STT API_KEY and URL. Save file.
3. Edit bluemix-test and insert your IBM STT API_KEY and URL. Save the file.
4. Copy the updated sendmailmp3.ibm file to sendmailmp3:
cd /usr/local/sbin cp -p sendmailmp3.ibm sendmailmp3
5. Test your Bluemix STT setup: bluemix-test
6. Result should be: we are now transferring you out of the company directory…
7. Set up voicemail for an extension and include your email address.
8. Place a test call to the extension and record a voicemail when prompted.
9. Your message will be transcribed and delivered via email.
Installing STT Engine with VitalPBX
For those using VitalPBX with or without Incredible PBX, we’ve written a new tutorial to walk you through the procedure to get voicemail transcription with IBM Watson STT up and running. Here’s the link.
Installing STT Engine with Legacy FreePBX Servers
1. Follow steps #1 through #8 from the Incredible PBX 13 tutorial above.
2. Choose Settings -> Voicemail Admin -> Settings in the GUI.
3. In the format field, insert: wav|wav49
4. In the mailcmd field, insert: /usr/local/sbin/sendmailmp3
5. Click Submit to save your settings and then Reload the FreePBX Dialplan.
6. Place a test call to the extension and record a voicemail when prompted.
7. Your message will be transcribed and delivered via email.
Update: Matt Darnell reports that, depending upon your existing setup, you may need to add the unix2dos and lame packages with legacy FreePBX servers to get MP3 messages delivered correctly.
Originally published: Monday, March 12, 2018 Updated: Monday, November 12, 2018
Got Friends? 7 Countries Have Never Visited Nerd Vittles. 2018 Is Calling! https://t.co/wMfmlhAr16 #asterisk #freepbx #wazo #issabel #IncrediblePBX #3CX pic.twitter.com/kAmAEnwVIw
— Ward Mundy (@NerdUno) January 9, 2018
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
VoIP 101: Developing a Cost-Effective SIP Strategy
In the lead up to the demise of Google Voice XMPP service next week, we wanted to offer what we have found to be a cost-effective SIP strategy which takes advantage of the best of all worlds. We would divide SIP offerings into five broad categories: business-class unlimited SIP trunks, Old Faithful SIP providers, Mom-and-Pop SIP services, dirt-cheap termination services, and Gee Whiz SIP providers. As we have said many times, the beauty of setting up an Asterisk® PBX such as Incredible PBX® is you don’t have to put all your VoIP eggs in one basket. In our particular case, that has included a mix of Google Voice trunks plus all five of the SIP categories above. Today we want to document why we’ve personally made the selections we’ve made and hope that it provides a roadmap for your own VoIP setup while encouraging you to venture out of your safe zone and try some new VoIP options.
The all-you-can-eat business plans, which we previously have covered, make little sense for most home and small business users. Then there are the rock-solid, long term pay-as-you-go providers such as Vitelity and CallCentric that make perfect sense as your primary DID and SIP provider. While they may not always be the cheapest VoIP providers, the tradeoff is dependability and long-term reliability for your VoIP platform. In the case of Vitelity, it turns out the Nerd Vittles DID special (detailed below) from our Platinum Sponsor is perhaps one of the best VoIP deals on the planet.
The third category of SIP providers and our personal favorite is what we would call the mom-and-pop providers. These are typically one or two-person operations that offer incredible deals on all-you-can-eat VoIP plans for home users. Included in this category are Vestalink (available to existing customers only), Future-Nine and CircleNet. VestaLink originally began as OBiVoice and morphed over trademark issues. While the service is no longer available to new customers, it remains the best bargain at $72 for two years of unlimited inbound and outbound residential calling services. A close second goes to Future-Nine and their "Future 5 Grey" plan which provides 1,500 inbound and 1,500 outbound minutes a month for only $5. You can sign up here. Be sure to read the Terms of Services carefully, especially item #18. The New Kid on the Block is CircleNet. In addition to very attractive pay-by-the-minute offerings of $.005 per minute to most of the U.S. and Canada, they also have an $8 a month all-you-can-eat plan for residential customers that includes a very reasonable 5,000 minutes a month for calls to the following countries: United States, Canada, Australia, Bangladesh, Belgium, Brazil, Chile, Cyprus, Denmark, Finland, France, Germany, Greece , Guam, Hungary, India,Ireland, Italy, Japan, Latvia, Mexico, Netherlands, New Zealand, Norway, Poland, Puerto Rico, Singapore, Spain, Sweden, Taiwan, Thailand, United Kingdom, and Vatican City. Just let them know that you plan to use it with an Asterisk-based PBX. CircleNet also is offering Nerd Vittles readers a free month of the $8/month service to kick the tires. Simply send an email to sales@circlenet.us with your valid email address to take advantage of the offer. One free trial per customer/email address. CircleNet also offers a $15 a month business plan with even more minutes.
A fourth class of VoIP providers is the dirt-cheap termination services including Anveo Direct, TelecomsXchange, V1VoIP and the Betamax companies for low-cost international calling. These providers make terrific additions for supplementing your other VoIP services. TelecomsXchange is our personal favorite because of the special deal they have extended to Incredible PBX users. You get access to 300 VoIP wholesalers and can read about their services in this Nerd Vittles article. V1VoIP also has some terrific deals with 15¢/mo. DIDs from 13,000 Rate Centers and incoming and outgoing U.S. call pricing as low as $.003 per minute (not a typo!). Anveo Direct was perhaps the first provider to offer wholesale pricing to consumers, and they remain a terrific service both for DID and origination services with T.38 fax support as well as many of the lowest cost SIP terminations worldwide featuring user-configurable least-cost routing. Check out their pricing and rates here.
Finally, there are the SIP providers such as VoIP.ms that offer a rich collection of special features that you won’t find in many places and certainly not under the same roof. These features include SMS messaging, SIP URI proxying and iNUM for free worldwide calling, and fax support. Every one of these features is free when you sign up for an account at VoIP.ms. We encourage you to take advantage of these little known free services to enhance your PBX.
Putting It All Together. Now that we’ve covered the options, let’s go over how we would actually implement this. For the inbound trunk and primary DID, we’d recommend a SIP trunk from either Vitelity, VoIP.ms, or CallCentric. If you have multiple, simultaneous inbound calls, then the Nerd Vittles Vitelity special below can’t be beat because it provides four call paths. In addition, you get SMS support on the same trunk. Many people now assume your primary number supports SMS. We actually get dozens of unsolicited SMS messages on our home number from schools, churches, and political groups. If incoming call volume isn’t an issue, then VoIP.ms and CallCentric also offer a free iNUM number for your account. And VoIP.ms throws in a SIP URI as well.
For outbound calling for home and SOHO deployments, we recommend at least one of the mom-and-pop, all-you-can-eat providers: Future-Nine or CircleNet. If international calling is a requirement, you can’t beat the CircleNet offering. In addition to using your primary incoming provider, we also recommend you set up SIP accounts with a couple of the dirt-cheap termination providers. These don’t cost you anything other than a modest deposit unless you actually use them to place calls. And, when your primary outbound service has an outage, your PBX will never miss a beat.
The icing on the cake always has been several Google Voice trunks which work well for IVRs, Stealth AutoAttendants with DISA support, and faxing. While this may change with the demise of XMPP support, it appears that Bill Simon’s SIP Gateway to Google Voice will live on. With the Nerd Vittles sign-up link, you can migrate your existing Google Voice XMPP connections to the Simonics gateway for $4.99 each should the need arise. Enjoy!
Originally published: Monday, June 11, 2018
CircleNet SIP Setup for FreePBX/IncrediblePBX/VitalPBX/Issabel:
username=acct-id type=friend trustrpid=yes sendrpid=yes secret=acct-pword qualify=yes nat=yes insecure=port,invite host=sip.circlenet.biz fromuser=acct-id context=from-trunk disallow=all allow=ulaw Registration String: acct-id:acct-pword@sip.circlenet.biz:5060/did-num
Future-Nine SIP Setup for FreePBX/IncrediblePBX/VitalPBX/Issabel:
username=acct-num type=friend trustrpid=yes sendrpid=yes secret=acct-pword qualify=yes nat=yes insecure=port,invite host=incoming.future-nine.com fromuser=acct-num context=from-trunk canreinvite=no disallow=all allow=ulaw Registration String: acct-num:acct-pword@incoming.future-nine.com/acct-num
Need help with Asterisk? Visit the PIAF Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
Dare to Compare: The Best (free) VoIP Offerings for 2018
Last week we showed you how to get 10 months of free hosting for your Incredible PBX® in the Cloud. And today we present our semi-annual survey of the latest and greatest VoIP offerings for 2018. The beauty of the cloud platform is you can try all of them for less than a penny an hour and decide for yourself which free offering best meets your needs. This year we’ve ushered in new Asterisk® 13 LTS releases of Incredible PBX® on the CentOS, Ubuntu, and Raspberry Pi platforms as well as new versions for Issabel 4 and VitalPBX. To sweeten the pot even further, we nailed down a new Cloud-based offering for $10 a year that makes a perfect VOIP sandbox for our CentOS platform. For 2018, we also secured new (free) DID offerings in the U.S. and announced a Nerd Vittles exclusive providing access to 300+ VoIP providers worldwide, all at wholesale prices. And, last but not least, we introduced Digium’s newest IP phones for Asterisk including a $59 model that makes a perfect VoIP companion.
Choosing the Best VoIP Platform for Your Needs
Choosing a VoIP platform is partially a subjective decision, but there also are some glaring red flags to consider. We suggest you begin by deciding whether your preferences include any must-have’s. Do your requirements mandate an open source solution? Do you need text-to-speech and voice recognition? Does the operating system have to be Linux-based and, if so, must it be CentOS, Debian, or Ubuntu? If you’ll be using SIP phones, must the platform include phone provisioning software for your phones, or is the ability to purchase it as an add-on sufficient? Is paid support important in making your platform decision and how much are you prepared to pay? Are automatic or pain-free software updates critical in making your selection? Is migration from an existing platform a factor? Does a preconfigured, secure firewall matter, or are you prepared to do it yourself or take your chances? Before choosing to ignore security, read this RIPS analysis of FreePBX®. Here’s a snippet from the article. Read it carefully. It’s your phone bill.
Since FreePBX is written completely in PHP, we decided to throw it into our code analysis tool RIPS. The results were more than surprising and should tell you why a rock-solid firewall is absolutely essential.
The total amount of detected vulnerabilities is very high. Luckily, the majority of the detected vulnerabilities are inside the administration control panel, such that attackers either need to steal a valid account or they have to trick an administrator into visiting a malicious website that triggers one of the critical vulnerabilities. For example, a remote command execution vulnerability could be triggered by a less critical cross-site scripting vulnerability. By chaining both vulnerabilities, the severity is increased drastically and can lead to full server compromise.
In choosing which platforms to include today, we eliminated platforms which we considered too complicated for the average new user to configure. We also eliminated any platform that did not offer at least a free tier of service with a reasonably complete feature set as part of their offering. So here’s our Pick of the Litter.
We must confess that we are partial to the Incredible PBX offerings because they provide a turnkey GPL platform with minimal configuration required on your part. Regardless of platform, all come standard with a preconfigured firewall and about three dozen applications for Asterisk that will help you learn everything there is to know about VoIP telephony.
VoIP Platform Feature Summary
Aggregation: Incredible PBX 13-13 for CentOS/SL
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 13 GPL modules
O/S: CentOS/SL 6.9 or 7
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic Update Utility included
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall Whitelist
Security Rating (as delivered): Secure
Comments: Lean & Mean or Whole Enchilada installers as well as ISO available
Aggregation: Incredible PBX 13-13 for Raspbian
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 13 GPL modules
O/S: Raspbian 7
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic Update Utility included
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall Whitelist
Security Rating (as delivered): Secure
Aggregation: Incredible PBX 13-13 for Ubuntu
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 13 GPL modules
O/S: Ubuntu 18.04
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic Update Utility included
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall Whitelist
Security Rating (as delivered): Secure
Comments: Lean & Mean or Whole Enchilada installers
Aggregation: VitalPBX
License: Closed Source
VoIP Platform: Asterisk 13
GUI: Free and Commercial modules
O/S: CentOS 7
Phone Provisioning: Free
Text-to-Speech/Voice Recognition: Optional/Optional
Software Updates: Automatic
Migration Tools: Yes
Security: Fail2Ban + User-Configurable Firewall
Security Rating (as delivered): Insecure
Comments: Incredible PBX add-on now available including TM3 firewall.
Aggregation: Incredible PBX for Issabel 4
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 11 GPL modules
O/S: CentOS 7
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: No/No
Software Updates: Semi-Automatic
Migration Tools: No
Security: Fail2Ban + Unconfigured Firewall
Security Rating (as delivered): Secure with Incredible PBX add-on
Comments: Incredible PBX add-on provides secure platform
Aggregation: FusionPBX for FreeSWITCH
License: Open Source MPL 1.1
VoIP Platform: FreeSWITCH 1.6
GUI: FusionPBX
O/S: Debian 8
Phone Provisioning: Free
Text-to-Speech/Voice Recognition: Optional/Optional
Software Updates: Automatic
Security: Fail2Ban + User-Configurable Firewall
Security Rating (as delivered): Secure with mods below
Comments: Incredible PBX firewall add-on now available .
Aggregation: Incredible PBX for Wazo
License: GPL3 Open Source
VoIP Platform: Asterisk 15 RealTime
GUI: Wazo GPL3 modules
O/S: Debian 9
Phone Provisioning: Extensive Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic or 2-minute Manual
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall
Security Rating (as delivered): Secure WhiteList with Incredible PBX add-on
Comments: High Availability & Call Center GPL3 Modules
Aggregation: FreePBX Distro a.k.a. AsteriskNOW
License: Closed Source
VoIP Platform: Asterisk 13/14/15
GUI: FreePBX GPL and Commercial modules
O/S: Closed-source CentOS fork
Phone Provisioning: Open Source (minimal) or Commercial
Text-to-Speech/Voice Recognition: Optional/No
Software Updates: Manual from Hidden Repo
Migration Tools: Yes
Security: Fail2Ban + User-Configurable Firewall
Security Rating (as delivered): Insecure
Comments: Extensive commercial NagWare preinstalled
Deploying a Local Server vs. Cloud Platform
We’ve always been big fans of local servers because you have almost total control of your own destiny. This was especially true when the Raspberry Pi came along to take the financial pain out of the server equation. But the price of Cloud-based servers has continued to plummet. For 2018, you can run any of our favorites on the least expensive platform at Vultr or Digital Ocean for $2.50 a month. And, if you hurry, your first 10 months are free at Vultr. Spending another 50 cents buys you automatic backups.1 And, for the Incredible PBX 13-13 build with CentOS 6.9 (64-bit), we’ve found a deal at HiFormance that offers a high-performance OpenVZ platform at an annual cost of just $10. The technical specs are impressive (even better if you sign up for 3 years), and we don’t think you’ll find a comparable deal with anything near comparable performance and specs anywhere, period. You get your choice of hosting sites including New York, Chicago, Los Angeles, Buffalo, Atlanta, and Dallas. Complete tutorial available here.
NOTE: OpenVZ/SolusVM platforms not suitable for CentOS 7, Debian 9, or Ubuntu 18 implementations, and some providers do not yet support Ubuntu 18.04 platform although Vultr and Digital Ocean both do.
Available Free Trunks for VoIP Servers
For many years, we’ve offered free Google Voice connectivity with our VoIP platforms. And that remains true at least for a few more weeks. On all of the Incredible PBX platforms, Google Voice trunks can be set up to make free calls in the U.S. and Canada provided you have a U.S. residence and a U.S. cellphone number to verify that you are who you say you are. There’s even a ray of hope that the Simonics gateway may allow you to continue using Google Voice after Google Voice’s mid-June drop-dead date for XMPP. Details here. But what about the rest of the world. For 2018, we solved the problem by offering free DID trunks for inbound calls and a collection of 300 wholesale VoIP carriers worldwide to make outbound calls at the same wholesale rates offered to the very largest resellers. Simply pay a 13% surcharge in lieu of the $650 annual fee, and TelecomsXchange (TCXC) will provide you access to their entire suite of wholesale carriers together with state-of-the-art tools to manage all of the services.2 The Nerd Vittles setup tutorial is available here. Enjoy!
Published: Monday, March 5, 2018 Updated: Sunday, May 27, 2018
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
- On the Vultr and Digital Ocean $2.50 platforms, be sure to (1) create a 1GB swapfile once you’ve chosen your operating system. (2) Then, for Vultr, issue the following command before beginning the Incredible PBX install: apt-get install cloud-init.
(3) Now complete the steps outlined in your preferred Nerd Vittles tutorial, and you’ll be all set in about 15 minutes. [↩] - Our special thanks to TelecomsXchange. They have generously offered to contribute a portion of the wholesale surcharge to support the Incredible PBX open source project. [↩]
Cloud 9: Free Incredible PBX in the Cloud Hosting until 2019
These deals don’t come along every day so we’re interrupting our regular programming to alert you to a terrific, limited time cloud hosting offer for first-time users of Vultr. If you hurry, you can take advantage of a $25 credit on Vultr which translates into 10 free months of cloud hosting service. We can’t say enough about Vultr. They’ve been one of our key resources for development and testing of new releases of Incredible PBX for many years. Historically, they’ve supported our open source projects through generous referral revenue although that does not apply with this special offer. If you’ve always wondered whether cloud hosting was a viable alternative to on-premise solutions, now’s your chance to kick the tires at zero cost. And the other good news is you have your choice of the following Incredible PBX offerings. Simply load the required OS or upload the ISO for the platform of your choice and follow the linked tutorials below. Enjoy!
- Incredible PBX for CentOS 6 or 7
- Incredible PBX with Incredible PBX 13-13 ISO
- Incredible PBX for Ubuntu 18.04
- Incredible PBX with VitalPBX 2.0 ISO
- Incredible PBX with Jan. 2018 Issabel ISO
- Incredible PBX with Wazo 17.17 (Debian 8)
Originally published: Friday, May 25, 2018
Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
Beginner’s Navigation Guide to VoIP PBXs and Nerd Vittles
Here at Nerd Vittles, we cover a lot of VoIP territory over the course of a year. To kick off the new year, we thought it might be helpful for those just beginning their VoIP adventure to sketch out the VoIP lay of the land for you. We’re assuming that you came to our site because you wanted a VoIP solution that gives you something to play with and to learn from. That’s not for everybody, and there are less flexible, turnkey VoIP solutions that function pretty much like a toaster. At the top of that short list would be the Ooma Telo and OBi200. Both offer (almost) free calling in the U.S. and Canada.
Keep in mind that all of us started as beginners so there’s no reason to be intimidated if you choose to deploy your own PBX. We’ve gotten a dozen years of enjoyment out of our adventures with VoIP telephony, and there’s no reason you can’t do the same. Let’s begin.
Choosing a Hardware Platform for Your VoIP PBX
First, you’ll need to choose a platform for your VoIP-based PBX: dedicated hardware, virtual machine, or cloud-based PBX. In no small part, this choice depends upon the target audience for your PBX. If it’s for home use or a SOHO business, a $35 Raspberry Pi may suffice. On the other hand, if your PBX will be supporting more than a dozen users or more than a handful of simultaneous calls, we’d look elsewhere. Many of Intel’s Atom-based PCs work very well. And a VirtualBox virtual PBX running atop an iMac or beefy Dell PC can support dozens of users if you have the necessary Internet bandwidth to handle your call volume. Cloud-based servers come in all shapes and sizes as well. As prices have plummeted, cloud solutions have become our favorite. For $3 to $6 a month, you now can host your PBX in the cloud with automatic image backups of your entire server every week. If you’re willing to forfeit backups, here is a cloud solution that will only set you back about a dollar a month. If your server is primarily for business use, we strongly recommend our Platinum Sponsor, RentPBX, that offers dozens of VoIP choices for $14.99 a month with coupon code: NOGOTCHAS.
Choosing the Best PBX to Meet Your Requirements
Once you’ve nailed down your hardware platform, the next step is choosing an operating system and PBX to support your individual requirements. As you might have guessed, there are dozens from which to choose. In both the open source and commercial PBX world, most systems require a specific version of Linux so your operating system choice typically is dependent upon the PBX you choose. In the open source world, the PBX learning curve is often related to the feature set being offered. More sophisticated feature sets typically have a steeper learning curve. If you’re just getting started with VoIP and you want a platform for learning, experimenting, or home use, you can’t beat Incredible PBX 13-13 Whole Enchilada. It was designed by us to be a turnkey PBX for first-time users with rock-solid security and all of the features you will ever need. It includes 31 applications for Asterisk® that cover every imaginable function that can be performed with a telephone including faxing, voice dialing, SMS messaging, wakeup calls and telephone reminders, free calling, conferencing, text-to-speech applications such as News Headlines and Weather Forecasts, Wolfram Alpha for Siri-like queries, plus all the usual PBX features: blacklists, call forwarding, call waiting, call transfer, call parking, call recording, intercom, voicemail including voicemail transcription with email delivery, IVRs, paging, AutoAttendants, DISA, and many more.
If you’re an experienced Asterisk developer that just wants a lean PBX where you can customize it to meet individual customer’s requirements, then Incredible PBX 13-13 Lean should be just the ticket. All of its components are configurable including Asterisk which can be recompiled from the included source code.
At the sophisticated end of the spectrum is Incredible PBX for Wazo which is based upon the Wazo PBX, an Asterisk 15 realtime implementation with full support for High Availability redundancy, multi-party videoconferencing, WebRTC, and automatic nightly backups. It includes API libraries from which you literally can build your own customized PBX from the ground up. The Incredible PBX feature set provides a platform with virtually identical applications to those found in Incredible PBX 13-13.
Sandwiched in between Incredible PBX 13-13 and Incredible PBX for Wazo is Incredible PBX 13 for Issabel. Issabel is an enhanced fork of the previous Elastix 4.0 PBX. The 2018 release includes Asterisk 13, the LTS version of the Asterisk platform. With the new Incredible PBX 13 add-on, you get the best of all worlds with Google Voice support and dozens of applications for Asterisk. Issabel provides a Unified Communications platform that is second to none in the open source world.
Thus far, all of our recommendations have been to open source, GPL-licensed PBX platforms. But you’d be making a mistake to limit your search for business telephony platforms to open source offerings. Our corporate sponsor, 3CX, offers a full year of their commercial PBX running in the Google Cloud at no cost. It’s incredibly simple to install and configure. And the beauty of the 3CX commercial platform is it can scale to any size as your business grows. And the 3CX feature set can be expanded geometrically as your business requirements mature. We added free text-to-speech applications for News and Weather reports just last week. Our favorite open source deployment strategy is to install a 3CX PBX alongside Incredible PBX which yields literally the best of both worlds. The 3CX clients for Windows and Macs, Android, and iOS make VoIP telephony available from anywhere with a couple of button clicks, and 3CX users experience none of the traditional communications problems that invariably crop up on platforms deployed by novice VoIP users running Asterisk.
Getting Started with Extensions, Trunks, and Routes
The Big 3 when it comes to PBX configuration are extensions, trunks, and routes. Extensions carry calls between phones on the PBX and other phones either inside or outside your home or office. Trunks actually provide the links between your PBX and the outside telephony world. Inbound routes tell your PBX where to send incoming calls while Outbound routes tell your PBX which trunk to use when calls are made to numbers outside your PBX. We’ve covered this in more detail including dozens of trunk setups in this Nerd Vittles tutorial.
Making Free U.S./Canada Calls within the United States
There are three ways to make free calls using your PBX. If you’re in the United States, you can use Google Voice to make free calls to the U.S. and Canada if your PBX supports Google Voice trunks, e.g. Incredible PBX 13-13 Whole Enchilada and Incredible PBX for Issabel. An alternative, if your PBX does not directly support Google Voice trunks, e.g. Incredible PBX for Wazo and 3CX, is to use the Simonics SIP to Google Voice Gateway service. For Nerd Vittles users, there is a one-time $4.99 signup fee with no additional charges ever. Whether you live in the U.S or not, all the PBXs we’ve covered today can make free SIP calls to anyone in the world that has a SIP URI address and a SIP phone. Most SIP softphones are free.
Mastering the Incredible PBX Feature Set
Configuring the Travelin’ Man 3 Firewall
All Incredible PBX servers include a firewall that is configured automatically as part of the installation process. On the 3CX platform, you’ll need to add the Travelin’ Man 3 firewall after installing your 3CX PBX. Here’s how:
Configuring a Firewall WhiteList:
WhiteListing Users with Travelin’ Man 3 and IPtables Firewall
Learning to Build Effective IVRs
Interactive Voice Response (IVR) systems and AutoAttendants are the bread-and-butter applications for businesses. If you’ve ever called a business and actually spoken to a live person without encountering an IVR, lucky you! But, believe it or not, IVRs can actually be a useful tool including our Stealth AutoAttendant which lets you intercept incoming calls with a greeting which provides a slight delay to allow the caller (or you) to reroute the call to a specific destination before the default destination kicks in. Nerd Vittles and the Incredible PBX offerings provide all of the tools you’ll need to build any type of IVR imaginable. Mastering Allison Smith’s Top 15 is an excellent starting point.
Harnessing Nerd Vittles Resources
Google is your friend when it comes to finding tutorials of interest in the VoIP world. To narrow searches to just Nerd Vittles, use the following syntax:
stealth autoattendant site:nerdvittles.com
And the Nerd Vittles site itself provides several powerful ways to drill down into topics of interest. In the upper right column of any article, you’ll find a search function which will return a list of matching articles to peruse. At the bottom of every article, check out the all-new Articles of Interest section of Nerd Vittles arranged by topic. Also in the right column of Nerd Vittles, you’ll find a listing of Categories with Nerd Vittles articles conveniently grouped by topic. And, finally, you can quickly jump to the lead article on every major Incredible PBX implementation in the color-coded tabs labeled: GPL VOIP SOLUTIONS FOR ALL.
Happy New Year!
Originally published: Monday, January 1, 2018
Support Issues. With any application as sophisticated as a VoIP PBX, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk and 3CX gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.
NEW YEAR’S TREAT: If you could use one or more free DIDs in the U.S. with unlimited inbound calls and unlimited simultaneous channels, then today’s your lucky day. TelecomsXChange and Bluebird Communications have a few hundred thousand DIDs to give away so you better hurry. You have your choice of DID locations including New York, New Jersey, California, Texas, and Iowa. The DIDs support Voice, Fax, Video, and even Text Messaging (by request). The only requirement at your end is a dedicated IP address for your VoIP server. Once you receive your welcome email with your number, be sure to whitelist the provider’s IP address in your firewall. For Incredible PBX servers, use add-ip to whitelist the UDP SIP port, 5060, using the IP address provided in your welcoming email.
Here’s the link to order your DIDs.
Your DID Trunk Setup in your favorite GUI should look like this:
Trunk Name: IPC
Peer Details:
type=friend
qualify=yes
host={IP address provided in welcome email}
context=from-trunk
Your Inbound Route should specify the 11-digit DID beginning with a 1. Enjoy!
Need help with Asterisk or 3CX? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
Free Asterisk Solutions