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VirtualBox Wonder: It’s Incredible PBX for Debian 11




If you’re new to the VoIP world and want to kick the tires to see what you’re missing, then today’s one minute setup is for you. You’ll get a $10 credit to try out some penny-a-minute calls and to purchase a $1 a month phone number in your choice of area codes. If you decide VoIP is not for you, you don’t have to buy anything ever. And you can use almost any desktop computer you already own to bring up the VirtualBox® edition of Incredible PBX® with Debian 11. Apple’s new ARM-based Macs unfortunately do not support VirtualBox.

If you’ve followed Nerd Vittles over the years, you already know that VirtualBox from Oracle® is one of our favorite platforms. Once VirtualBox is installed on your desktop computer, adding Incredible PBX is a snap. Download the latest Incredible PBX image for Debian 11 from SourceForge, double-click on the downloaded image and boom. In less than a minute, your PBX is ready to use with the very latest components of Asterisk® 18 and FreePBX® 15. There are no hidden fees or crippleware to hinder your use of Incredible PBX for as long as you like. Just set up an account with our Platinum provider, Skyetel, and you can start making calls in minutes. Of course, the Incredible PBX feature set is included as well which brings you nearly three dozen applications for Asterisk that will revolutionize your communications platform. Speech-to-text, voice recognition, and a Siri-like telephony interface are as close as your SIP phone.

Installing Oracle VM VirtualBox

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

NOTE: A VirtualBox 6.x platform is required. Adjust screen size in View -> Virtual Screen.

Installing the Incredible PBX for Debian 11 Image

To begin, download the Incredible PBX for Debian 11 image (3.7 GB) onto your desktop.

Next, double-click on the Incredible PBX .ova image on your desktop. Be sure to check the box to initialize the MAC address of the image if you’re using an older version of VirtualBox. Then click Import. Once the import is finished, you’ll see a new Incredible PBX for Debian 11 virtual machine in the VM List of the VirtualBox Manager Window. Let’s make a couple of one-time adjustments to the Incredible PBX configuration to account for possible differences in sound and network cards on different host machines.

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

Running Incredible PBX for Debian 11 in VirtualBox

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

Here’s what you need to know. To work in the Incredible PBX virtual machine, just left-click your mouse while it is positioned inside the VM window. To return to your host operating system desktop, press the right Option key on Windows machines or the left Command key on any Mac. On Linux desktops, press the right Ctrl key. For other operating systems, read the dialogue boxes for instructions on moving around. To access the Linux CLI, login as root with the default password: password. Change your root password when you are prompted to do so. Then update your admin password for web access: ./admin-pw-change. Also update your admin password for web applications: ./apache-pw-change. You’ll need these admin passwords to access the web GUI to manage your PBX as well as to use the AsteriDex and Reminders web apps. The above password updates are automatically requested when you first activate the virtual machine. You can update all of your other passwords using the scripts provided in /root.

Setting the Date and Time with VirtualBox

On some platforms, VirtualBox has a nasty habit of mangling the date and time of your virtual machine. Verify that you have enabled the Hardware Clock in UTC Time option for your virtual machine as documented above. If pbxstatus still shows an incorrect time, manually set the date and time and then update the hardware clock. Here’s how assuming 08130709 is the month (August), day (13), and correct time (7:09 a.m.) of your server:

date 08130709
clock -w

Configuring Skyetel for Incredible PBX

If you’d like to try out the Skyetel service at no charge, here’s the drill. Sign up for Skyetel service to take advantage of the Nerd Vittles specials. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request the $10 credit for your account by referencing the Nerd Vittles special offer. Once you are satisfied with the service, fund your account as desired, and Skyetel will match your deposit of up to $250 simply by opening another ticket. That gets you up to $500 of half-price calling. Credit is limited to one per person, company, and address. Effective 10/1/2023, $25/month minimum spend required.

Skyetel by default does not use SIP registrations to make connections to your PBX. Instead, Skyetel utilizes Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. Here’s a typical setup for Incredible PBX:

  • Name: MyPBX
  • Priority: 1
  • IP Address: PBX-Public-IP-Address
  • Port: 5060
  • Protocol: UDP
  • Description: my.incrediblepbx.com

To receive incoming PSTN calls, you’ll need at least one DID. On the Skyetel site, you acquire DIDs under the Phone Numbers tab. You have the option of Porting in Existing Numbers (free for the first 60 days after you sign up for service) or purchasing new ones under the Buy Phone Numbers menu option.

Once you have acquired one or more DIDs, navigate to the Local Numbers or Toll Free Numbers tab and specify the desired SIP Format and Endpoint Group for each DID. Add SMS/MMS and E911 support, if desired. Call Forwarding and Failover are also supported. That completes the VoIP setup on the Skyetel side. System Status is always available here.

If VirtualBox is sitting behind a router or firewall on a private LAN, you’ll need to forward ports UDP 5060 and 10000-20000 in your router to the private LAN address of your Incredible PBX server. Also edit your extensions in the GUI and set NAT=YES in the Advanced tab of every extension. In Settings -> Asterisk SIP Settings, click the Detect Network Settings button and then Submit your changes and reload the Asterisk dialplan when prompted.

Configuring VoIP.ms for Incredible PBX

To sign up for VoIP.ms service, may we suggest you use our signup link so that Nerd Vittles gets a referral credit for your signup. Once your account is set up, you’ll need to set up a SIP SubAccount and, for Authentication Type, choose Static IP Authentication and enter your Incredible PBX server’s public IP address. For Transport, choose UDP. For Device Type, choose Asterisk, IP PBX, Gateway or VoIP Switch. Order a DID in their web panel, and then point the DID to the SubAccount you just created. Be sure to specify atlanta1.voip.ms as the POP from which to receive incoming calls.

Configuring Anveo Direct for Incredible PBX

To sign up for Anveo Direct service, sign up on their web site and then login. After adding funds to your account, purchase a DID under Inbound Service -> Order DID. Next, choose Configure Destination SIP Trunk. Give the Trunk a name. For the Primary SIP URI, enter $[E164]$@server-IP-address. For Call Options, select your new DID from the list. You also must whitelist your public IP address under Outbound Service -> Configure. Create a new Call Termination Trunk and name it to match your server. For Dialing Prefix, choose six alphanumeric characters beginning with a zero. In Authorized IP Addresses, enter the public IP address of your server. Set an appropriate rate cap. We like $0.01 per minute to be safe. Set a concurrent calls limit. We like 2. For the Call Routing Method, choose Least Cost unless you’re feeling extravagant. For Routes/Carriers, choose Standard Routes. Write down your Dialing Prefix and then click the Save button.

Before you can make outbound calls through Anveo Direct from your PBX, you first must configure the Dialing Prefix that you wrote down in the previous step. Log into the GUI as admin using a web browser and edit the Anveo-Out trunk in Connectivity -> Trunks. Click on the custom-Settings tab and replace anveo-pin with your actual Dialing Prefix. Click Submit and Apply Config to complete the setup.

By default, incoming Anveo Direct calls will be processed by the Default inbound route on your PBX. If you wish to redirect incoming Anveo Direct calls using DID-specific inbound routes, then you’ve got a bit more work to do. In addition to creating the inbound route using the 11-digit Anveo Direct DID, enter the following commands after logging into your server as root using SSH/Putty:

cd /etc/asterisk
echo "[from-anveo]" >> extensions_custom.conf
echo "exten => _.,1,Ringing" >> extensions_custom.conf
echo "exten => _.,n,Goto(from-trunk,\\${SIP_HEADER(X-anveo-e164)},1)" >> extensions_custom.conf
asterisk -rx "dialplan reload"

Configuring a Softphone for Incredible PBX

We’re in the home stretch now. You can connect virtually any kind of telephone to your new PBX. Plain Old Phones require an analog telephone adapter (ATA) which can be a separate board in your computer from a company such as Digium. Or it can be a standalone SIP device. SIP phones can be connected directly so long as they have an IP address. These could be hardware devices or software devices such as the YateClient softphone. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum when you’re ready to get serious about VoIP telephony.

We recommend YateClient for Windows which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You can find them by running /root/show-passwords. You’ll need the IP address of your server plus your extension 701 password. In the YateClient, fill in the blanks using the IP address of your Server, 701 for your Username, and whatever Password was assigned to the extension when you installed Incredible PBX. Click OK to save your entries.

Configuring Incredible PBX for VirtualBox

In order to take advantage of all the Incredible PBX applications, you’ll need to obtain IBM text-to-speech (TTS) and speech-to-text (STT) credentials as well as a (free) Application ID for Wolfram Alpha.

This Nerd Vittles tutorial will walk you through getting your IBM account set up and obtaining both your TTS and STT credentials. Be sure to write down BOTH sets of credentials which you’ll need in a minute. For home and SOHO use, IBM access and services are mostly FREE even though you must provide a credit card when signing up. The IBM signup process explains their pricing plans.

To use Wolfram Alpha, sign up for a free Wolfram Alpha API account. Just provide your email address and set up a password. It takes less than a minute. Log into your account and click on Get An App ID. Make up a name for your application and write down (and keep secret) your APP-ID code. That’s all there is to getting set up with Wolfram Alpha. If you want to explore costs for commercial use, there are links to let you get more information.

In addition to your Wolfram Alpha APPID, there are two sets of IBM credentials to plug into the Asterisk AGI scripts. Keep in mind that there are different usernames and passwords for the IBM Watson TTS and STT services. The TTS credentials will look like the following: $IBM_username and $IBM_password. The STT credentials look like this: $API_USERNAME and $API_PASSWORD. Don’t mix them up. 🙂

All of the scripts requiring credentials are located in /var/lib/asterisk/agi-bin so switch to that directory after logging into your server as root. Edit each of the following files and insert your TTS credentials in the variables already provided: nv-today2.php, ibmtts.php, and ibmtts2.php. Edit each of the following files and insert your STT credentials in the variables already provided: getquery.sh, getnumber.sh, and getnumber2.sh. Finally, edit 4747 and insert your Wolfram Alpha APPID.

Using AsteriDex with Incredible PBX

AsteriDex is a web-based dialer and address book application for Asterisk and Incredible PBX. It lets you store and manage phone numbers of all your friends and business associates in an easy-to-use SQLite3 database. You simply call up the application with your favorite web browser: http://pbx-ip-address/asteridex4/. When you click on a contact that you wish to call, AsteriDex first calls you at extension 701, and then AsteriDex connects you to your contact through another outbound call made using your default outbound trunk that supports numbers in the 1NXXNXXXXXX format.

Keeping FreePBX 15 Modules Current

We strongly recommend that you periodically update all of your FreePBX modules to eliminate bugs and to reduce security vulnerabilities. From the Linux CLI, log into your server as root and issue the following commands:

rm -f /tmp/*
fwconsole ma upgradeall
fwconsole reload
/root/sig-fix
systemctl restart apache2
/root/sig-fix

Taking Incredible PBX for a Test Drive

You can take Incredible PBX on a test drive by dialing D-E-M-O (3366) from any phone connected to your PBX.

With Allison’s Demo IVR, you can choose from the following options:

  • 0. Chat with Operator — connects to extension 701
  • 1. AsteriDex Voice Dialer – say "Delta Airlines" or "American Airlines" to connect
  • 2. Conferencing – log in using 1234 as the conference PIN
  • 3. Wolfram Alpha Almanac – say "What planes are flying overhead"
  • 4. Lenny – The Telemarketer’s Worst Nightmare
  • 5. Today’s News Headlines — courtesy of Yahoo! News
  • 6. Weather by ZIP Code – enter any 5-digit ZIP code for today’s weather
  • 7. Today in History — courtesy of OnThisDay.com
  • 8. Chat with Nerd Uno — courtesy of SIP URI connection to 3CX iPhone Client
  • 9. DISA Voice Dialer — say any 10-digit number to be connected
  • *. Current Date and Time — courtesy of Incredible PBX

Originally published: Monday, September 19, 2022



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.
 




 

  1. Many of our purchase links refer users to Amazon when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from Amazon to help cover the costs of our blog. We never recommend particular products solely to generate Amazon commissions. However, when pricing is comparable or availability is favorable, we support Amazon because Amazon supports us. []

Happy Fourth: Our Gift to You — 17+ Years of Nerd Vittles


Originally published: Monday, July 4, 2022



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.
 



Introducing Incredible PBX 2022 for the Raspberry Pi



It’s been a year and a half since our last Incredible PBX® release for the Raspberry Pi platform, and the RasPi enhancements just keep coming. The latest RasPi 4 supports dual 4K monitors, two USB 2.0 ports, two USB 3.0 ports, gigabit Ethernet, a USB-C power supply, a Broadcom BCM2711, Quad core Cortex-A72 (ARM v8) 64-bit SoC running at 1.5GHz, and POE connectivity with the addition of the POE Hat. You can read all about it here. Incredible PBX 2022 supports the latest RasPi boards and keyboard and is backwards compatible.

UPDATE: Download the latest Incredible PBX 2027 image for RasPi here.

We’ve preserved the Raspbian 10 platform in this build because the Debian 11 release has broken free faxing, at least for the short term. This build features Asterisk® 16 or 18 with the latest FreePBX® 15 GPL modules plus the feature set you know and love. We’ve added PJSIP support for Skyetel and the new gTTS release for terrific text-to-speech applications including our News Headlines (951) and Weather Forecasts by ZIP Code (947). It’s all rolled into one terrific bundle that can be installed in about a minute after you download the image from SourceForge and burn the image to a microSD card.

Unlike other aggregations, there’s nothing to compile with Incredible PBX/FAX 2022 for Raspbian. And, unlike the FreePBX Distro, we don’t rely on static packages which make it difficult to add future modifications on your own. Instead, Incredible PBX/Fax 2022 offers a snapshot image with a complete toolkit to make future modifications as desired. And, of course, Incredible PBX/Fax 2022 features the ClearlyIP module repository which protects you from proprietary modifications that limit or cripple your PBX moving forward.

What’s Included? Incredible PBX/Fax 2022 serves up a never before available VoIP powerhouse featuring Asterisk 16 or 18 and all FreePBX 15 GPL modules, an Apache web server, the latest MariaDB SQL server (formerly MySQL), Exim4 mail server, Incredible Fax with turnkey Hylafax and AvantFax, and most of the Incredible PBX feature set including SIP, PJSIP, SMS, voice recognition, AsteriDex, gTTS Text-to-Speech VoIP applications plus email delivery of faxes in PDF format, Click-to-Dial, News, Weather, Telephone Reminders, and hundreds of features that typically are found in commercial PBXs: Conferencing, IVRs and Email Delivery of transcribed voicemails, AutoAttendants, Voicemail Blasting, and more. We’ve also incorporated the Zero Trunk Configuration feature from the LITE build which lets you sign up with one of our VoIP providers and start making and receiving calls instantly. Or you can use the new ClearlyIP trunking module included in the GUI for seamless integration of SMS messaging into FreePBX and its User Control Panel.

Choosing a SIP Provider. As we mentioned, Incredible PBX/Fax 2022 comes preconfigured to support many of the major SIP providers including those that financially support Nerd Vittles and our open source projects: ClearlyIP, Skyetel, and VoIP.ms. As the old saying goes, they may not be the cheapest, but you get what you pay for. With all our providers, you only pay for minutes you use so signing up with more than one provider is a smart idea. For the full list of supported VoIP providers, visit the Incredible PBX Wiki.

>

Assembling the Required Raspberry Pi Components

Before you can deploy Incredible PBX 2022, you’ll first need the necessary Raspberry Pi hardware. To support the enhanced Incredible PBX/Fax platform, we strongly recommend either the Raspberry Pi 400 or the Raspberry Pi 4B with at least 2GB RAM for under $42. You can choose a reseller below for quicker delivery. Assuming you already own an HDMI-compatible monitor and a USB keyboard (only required if you don’t buy a RasPi 400)…

  • Raspberry Pi 4B or Raspberry Pi 400
  • $8 USB-C RasPi 4 (only) Power Supply
  • $11 32GB microSDHC Class 10 card (strongly recommended!)
  • $5 Official RasPi 4B Case or see above for our favorite
  • Getting Started with Incredible PBX 2022

    Here’s our 10-Step Guide to installation and setup. "Automatic" means just watch. Steps #1 and #2: follow the links. For the remaining steps, we’ll further document the procedures.

    1. Download and unzip latest Incredible PBX/FAX 2022.6 image from SourceForge
    2. Transfer Incredible PBX/FAX 2022 image to microSD card and Boot server
    3. Login to RasPi console as root:password to initialize your server (Automatic)
    4. In Localization Options, set Locale, TimeZone, Keyboard, & WiFi Country
    5. Reboot after writing down your server IP address (Automatic)
    6. Login via SSH or Putty as root:password to set passwords & setup firewall (Automatic)
    7. Enter an email address for receipt of incoming faxes in PDF format
    8. Run admin-pw-change to set the admin password for access to the web GUI
    9. Register for and configure at least one trunk provider for Incredible PBX 2022
    10. Set up and test your Exim mail server as documented below

    ALERT: Reportedly, the latest Raspberry Pi 4 board will not boot with earlier Incredible PBX images. Today’s updated image solves that, but you may wish to simply move your existing build to the latest RasPi hardware and preserve your data. If you have an older (working) Raspberry Pi, simply issue the following commands on the old platform. Following shutdown, insert the new microSD card into your new RasPi 4.

    apt update
    apt dist-upgrade
    halt
    

    First Boot of Incredible PBX 2022 with Wi-Fi

    Incredible PBX 2022 requires Internet connectivity to complete its automated install. If you’re using a wired network connection, you can skip to the next section. With the Raspberry Pi 3B, 4B and 400, WiFi is built into the hardware. But you still have to insert your SSID name and SSID password to make a connection to your WiFi network. To do so, follow these next steps carefully. Insert the Incredible PBX 2022 microSD card into your Raspberry Pi and apply power to the hardware. When the bootup procedure finishes, login as root with the default password: password. At the first prompt, DO NOT PRESS THE ENTER KEY! Instead, press Ctrl-C to break out of the setup script. At the command prompt, issue the following commands to bring up the WiFi config file:

    cd /etc/wpa_supplicant
    nano -w wpa_supplicant.conf
    

    If your WiFi network does not require a password, uncomment or insert the four lines below and save the file: Ctrl-X, Y, then Enter. Now restart your server: reboot. When the reboot finishes, you now should have network connectivity.

    network={
     key_mgmt=NONE
     priority=1
     country=US
    }
    

    If your WiFi network requires a password, uncomment or insert the following into wpa_supplicant.conf:

    ctrl_interface=DIR=/var/run/wpa_supplicant GROUP=netdev
    update_config=1
    country=US
    
    network={
     ssid="YourSSID"
     psk="YourSSIDpassword"
     key_mgmt=WPA-PSK
     scan_ssid=1
     priority=7
    }
    

     
    Then scroll down to the SSID entry and replace YourSSID with the actual SSID of your WiFi network. Make sure you preserve the entry with the quotes as shown. Next, replace YourSSIDpassword with the SSID password of your WiFi network. Save the file: Ctrl-X, Y, then Enter. Now restart your server: reboot. When the reboot finishes, you now should have network connectivity.

    Once the reboot process finishes, you should see an entry on about the middle line displayed on your monitor which reads: "My IP address is…". Write down the IP address shown. You’ll need it in a minute. Skip the next section since you are using a WiFi connection.

    If you don’t see an IP address assigned to your server, then correct the network deficiency (invalid WiFi credentials, DHCP not working, Internet down), and reboot until you see an IP address assigned to your server. DO NOT PROCEED WITHOUT AN ASSIGNED IP ADDRESS. NOTE: The Raspberry Pi 400 requires the latest Incredible PBX image for Wi-Fi connectivity.

    You’ll also need to change the default PortKnocker setting to your wireless LAN connection:

    sed -i 's|eth0|wlan0|' /etc/default/knockd
    service knockd restart
    

     

    First Boot of Incredible PBX Using Wired Connection

    Incredible PBX 2022 requires Internet connectivity to complete its automated install. After connecting your server to your local network with a network cable, insert the Incredible PBX 2022 microSD card into your Raspberry Pi and apply power to the hardware. When the bootup procedure finishes, you should see an entry on about the middle line displayed on your monitor which reads: "My IP address is…". Write down the IP address shown. You’ll need it in the next step.

    If you don’t see an IP address assigned to your server, then correct the network deficiency (cable not connected, DHCP not working, Internet down), and reboot until you see an IP address assigned to your server. DO NOT PROCEED WITHOUT AN ASSIGNED IP ADDRESS.

    Completing the Incredible PBX Initialization Procedure

    Unless your desktop PC and RasPi are both on the same private LAN, the remainder of the install procedure should be completed from a desktop PC using SSH or Putty. This will assure that your desktop PC is also whitelisted in the Incredible PBX firewall. Using the console to complete the install is NOT recommended as your desktop PC will not be whitelisted in the firewall. This may result in your not being able to log in to your server. Once you have network connectivity, log in to your server as root from a desktop PC using the default password: password. Accept the license agreement by pressing ENTER. You then will be redirected to raspi-config. This is the utility used to expand your Incredible PBX 2022 image to use your entire microSD card; however, this new build does this for you so you can skip this step. Next, choose Localization Options and set Locale, TimeZone, Keyboard, & WiFi Country. Review the other items and then exit and reboot.

    Once your server reboots and you log back in as root, you’ll first be prompted to enter an email address for delivery of incoming faxes in PDF format. All of your passwords then will be randomly assigned with the exception of the root user Linux password and your admin passwords for access to the web GUI and AvantFax. You can set the root password by issuing the command: passwd. Set the admin password for access to the web GUI with this command: /root/admin-pw-change. Set the admin password for access to AvantFax with this command: /root/avantfax-pw-change. With the exception of these passwords, the remaining passwords can be displayed using the command: /root/show-passwords.

    Finally, if your PBX is sitting behind a NAT-based router, you’ll need to redirect incoming UDP 5060-5061 and UDP 10000-20000 traffic to the private IP address of your RasPi. This is required for all of the SIP providers included in the Incredible PBX 2022 build. Otherwise, all inbound calls will fail.

    Configuring Skyetel for Incredible PBX 2022

    If you’ve decided to go with Skyetel, here’s the drill. Sign up for Skyetel service and take advantage of the Nerd Vittles Free $10 credit and BOGO special. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request the $10 credit for your account by referencing the Nerd Vittles special offer. Once you are happy with the service, open another ticket after funding your account and request that Skyetel match your deposit of up to $250. That gets you up to $500 of helf-price calling. Credit is limited to one per person/company/address/location. If you have numbers to port in, you can do it at no cost after funding your account. Effective 10/1/2023, $25/month minimum spend required.

    Skyetel typically does not require SIP registrations to make connections to your PBX. Instead, Skyetel utilizes Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. Here’s a typical setup for Incredible PBX 2022:

    • Name: MyPBX
    • Priority: 1
    • IP Address: PBX-Public-IP-Address
    • Port: 5061
    • Protocol: UDP
    • Description: 2022.incrediblepbx.com

    To receive incoming PSTN calls, you’ll need at least one DID. On the Skyetel site, you acquire DIDs under the Phone Numbers tab. You have the option of Porting in Existing Numbers (free for the first 60 days after you fund your account) or purchasing new ones under the Buy Phone Numbers menu option.

    Once you have acquired one or more DIDs, navigate to the Local Numbers or Toll Free Numbers tab and specify the desired SIP Format and Endpoint Group for each DID. Add SMS/MMS and E911 support, if desired. Call Forwarding and Failover are also supported. That completes the VoIP setup on the Skyetel side. System Status is always available here.

    Configuring VoIP.ms for Incredible PBX 2022

    To sign up for VoIP.ms service, may we suggest you use our signup link so that Nerd Vittles gets a referral credit for your signup. Once your account is set up, you’ll need to set up a SIP SubAccount and, for Authentication Type, choose Static IP Authentication and enter your Incredible PBX 2022 server’s public IP address. For Transport, choose UDP. For Device Type, choose Asterisk, IP PBX, Gateway or VoIP Switch. Order a DID in their web panel, and then point the DID to the SubAccount you just created. Be sure to specify atlanta1.voip.ms as the POP from which to receive incoming calls. On the Incredible PBX side, simply Enable the VoIPms trunk and save your update.

    Adding a Bootable SSD to Raspberry Pi

    Shown below are the two components that make up the 256GB storage solution for the Raspberry Pi. These include the M.2 SSD SATA drive and the M.2 enclosure which provides a USB connector that’s compatible with your RasPi. Assembly of the components takes less than a minute as shown in the steps below:




    You can order the M.2 SSD SATA drive and the UGREEN M.2 enclosure using our Amazon referral links which help support Nerd Vittles and the Incredible PBX open source project.

    Once you have assembled your SSD in the sleeve, log back in as root using SSH or Putty. For best performance, insert the SSD drive into one of the blue USB 3.0 ports and verify that /dev/sda device is shown when you issue the command: fdisk -l

    Now proceed with the following steps to copy the image from your microSD card to the new SSD SATA drive:

    rpi-clone -l -e sda -f sda
    # answer prompts with yes and incred2022
    # once the image is copied, dismount the drive when prompted
    mount /dev/sda2 /mnt/clone
    cd /mnt/clone/boot
    cp -p -r /boot/* .
    sed -i 's|sda2|mmcblk0p2|' /boot/cmdline.txt
    cd /
    umount /mnt/clone
    halt
    

     
    Now you’re ready to restart your Raspberry Pi from the SSD SATA drive. Remove the microSD card and reboot your server.



    Configuring a Softphone for Incredible PBX 2022

    We’re in the home stretch now. You can connect virtually any kind of telephone to your new PBX. Plain Old Phones require an analog telephone adapter (ATA) which can be a separate board in your computer from a company such as Digium. Or it can be a standalone SIP device such as ObiHai’s OBi100 or OBi110 (if you have a phone line from Ma Bell to hook up as well). SIP phones can be connected directly so long as they have an IP address. These could be hardware devices or software devices such as the YateClient softphone. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum when you’re ready to get serious about VoIP telephony.

    We recommend YateClient for Windows which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You can find them by running /root/show-passwords. You’ll need the IP address of your server plus your extension 701 password. In the YateClient, fill in the blanks using the IP address of your Server, 701 for your Username, and whatever Password was assigned to the extension when you installed Incredible PBX. Click OK to save your entries.

    Once you are registered to extension 701, close the Account window. Then click on YATE’s Telephony Tab and place some test calls to the numerous apps that are preconfigured on Incredible PBX. Dial a few of these to get started:

    DEMO - Apps Demo
    123 - Reminders
    947 - Weather by ZIP Code
    951 - Yahoo News
    TODAY - Today in History
    LENNY - The Telemarketer's Worst Nightmare
    

    If you are a Mac user, another great no-frills softphone is Telephone. Just download and install it from the Mac App Store.

    Audio Issues with Incredible PBX 2022

    Only if you experience one-way or no audio on some calls, add your external IP address and LAN subnet in the GUI by navigating to Settings -> Asterisk SIP Settings. In the NAT Settings section, click Detect Network Settings. Click Submit and Apply Settings to save your changes.

    Configuring Gmail as Exim Smart Relay Host

    Most Raspberry Pi implementations will be on networks managed by companies like Comcast, Spectrum, and AT&T that block downstream mail servers (that’s you) from sending email. The solution is to use Gmail or your local ISP as a smart relay host to send mail from your server. You’ll need this to deliver voicemails via email. Here’s how to set it up using a Gmail account without two-step authentication. Log into your server as root and run dpkg-reconfigure exim4-config. Choose "mail sent by smarthost; received via SMTP or fetchmail." Accept all the defaults until you get to Outgoing Smarthost prompt. Enter: smtp.gmail.com::587. At the following prompts, choose NO, NO, mbox, and NO. When the setup completes, edit /etc/exim4/passwd.client and insert the following line using your Gmail AcctName and AcctPW. NOTE: Because insecure Gmail access is going away, you MUST use a Gmail App Password instead of your Gmail account password.

    smtp.gmail.com:AcctName@gmail.com:AcctPW
    

    Save the file and then issue the following commands to complete the setup:

    update-exim4.conf
    systemctl restart exim4
    exim4 -qff
    

    Now send yourself a test email message to make sure things are working properly:

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

    Once you have email messages flowing, incoming faxes automatically will be delivered to the email address you assigned when setting up your PBX. You can change this email address with the command: avantfax-email-change.

    Fixing Corrupted rc.local File & More

    Some prefer an email notification whenever your server is booted. Also fixes the corrupted rc.local file. Once you have configured a relay host above, you can add the feature by editing /etc/rc.local and making the file look like this replacing name@domain.com with your actual email address:

    #!/bin/sh -e
    
    # Print the IP address
    _IP=$(hostname -I) || true
    if [ "$_IP" ]; then
      printf "My IP address is %s\n" "$_IP"
    fi
    
    _PRIVATE="Private IP: `cat /etc/hostip | cut -f1-2 -d " "`"
    _PUBLIC="Public: $(dig TXT +short o-o.myaddr.l.google.com @ns1.google.com | sed 's|"||g')"
    echo "$_PRIVATE   $_PUBLIC" | mail -s "Incredible PBX 2022.6 has booted" name@domain.com
    
    sleep 5
    
    service knockd start
    sleep 30
    chmod -R 777 /var/www/html/avantfax
    exit 0
    

    Configuring Inbound Routes for Fax Detection

    Not all VoIP trunks support fax transmission, e.g. Vitelity. Assuming yours do and you’ll only know by trial and error, here’s how to configure FreePBX to automatically detect incoming faxes and process them for PDF delivery by email. The default inbound route is preconfigured to support email delivery of your faxes. So, any trunks using that default route require no further configuration. If you add additional Inbound Routes, here’s how to enable fax detection on those routes.

    Under the Fax tab of each new Inbound Route, enter the following settings:

    Detect Faxes: YES
    Fax Detection Type: SIP
    Fax Ring: YES
    Fax Detect Time: 4
    Fax Destination: Custom Destinations -> Fax (Hylafax)
    



    Managing Faxes with AvantFax

    You can manage your incoming and outgoing faxes using AvantFax. Click on the AvantFax tab in FreePBX to access it. The default credentials are admin:password. When you first access AvantFax with a browser, you may get a missing page error. Just press the back arrow key in your browser and the AvantFax main page will appear.

    If you want to change the admin password for AvantFax, log into your server as root with SSH/Putty and issue the command: /root/avantfax-pw-change.

    Send yourself a fax at no cost in the United States from FaxZero.

    Building the Incredible PBX Demo IVR

    If you’d like to try your hand at building an IVR, here are the steps to build the Incredible PBX Demo IVR. From the FreePBX Dashboard, choose Applications -> IVR -> Add IVR. Then fill in the template using the entries shown below. Then click Submit and Reload Dialplan.



    Building the Incredible PBX Stealth AutoAttendant

    Many users prefer to play an announcement to incoming callers with a brief pause thereafter which indicates that the call is being connected. If configured properly, this lets you embed several dial codes which can be entered while the announcement is playing and the call is being transferred. For example, you might wish to route incoming calls to Lenny if a caller presses 0. Or you might wish to immediately route an incoming call to a Ring Group if the caller presses 1. Here’s a sample IVR setup to get you started.


    Incredible PBX 2022 Administration

    We’ve eased the pain of administering your new PBX with a collection of scripts which you will find in the /root folder after logging in with SSH or Putty. Here’s a quick summary of what each of the scripts does.

    admin-pw-change lets you update the admin password for web browser access to the Incredible PBX GUI.

    apache-pw-change lets you update the admin password for Apache applications such as AsteriDex and Reminders.

    avantfax-pw-change lets you update the root password for AvantFax access (coming soon!).

    add-fqdn is used to whitelist a fully-qualified domain name in the firewall. Because Incredible PBX 2022 blocks all traffic from IP addresses that are not whitelisted, this is what you use to authorize an external user for your PBX. The advantage of an FQDN is that you can use a dynamic DNS service to automatically update the IP address associated with an FQDN so that you never lose connectivity.

    add-ip is used to whitelist a public IP address in the firewall. See the add-fqdn explanation as to why this matters.

    del-acct is used to remove an IP address or FQDN from the firewall’s whitelist.

    configure-exim-email lets you reconfigure the email server if you need to use an SMTP relay such as Google to get outbound email flowing. Tutorial here.

    iptables-restart is the ONLY command you should ever use to restart the IPtables firewall and Fail2Ban.

    knock.FAQ contains your PortKnocker credentials for emergency access to your server if the firewall locks you out. Tutorial here.

    proximity (once configured) will automatically forward calls to your cellphone when you are out of BlueTooth range from your RasPi. Also must enable running of script in /etc/crontab.

    reset-conference-pins is a script that automatically and randomly resets the user and admin pins for access to the preconfigured conferencing application. Dial C-O-N-F from any registered SIP phone to connect to the conference.

    reset-extension-passwords is a script that automatically and randomly resets ALL of the SIP passwords for extensions 701-705. Be careful using this one, or you may disable existing registered phones and cause Fail2Ban to blacklist the IP addresses of those users. HINT: You can place a call to the Ring Group associated with all five extensions by dialing 777.

    reset-reminders-pin is a script that automatically and randomly resets the pin required to access the Telephone Reminders application by dialing 123. It’s important to protect this application because a nefarious user could set up a reminder to call a number anywhere in the world assuming your SIP provider’s account was configured to allow such calls.



    rpi-clone is a utility that makes it easy to make a bootable image of the microSD card used to start your Raspberry Pi. You’ll need a USB-to-microSD adapter to begin. Insert a backup microSD card large enough to hold all of the data on the primary microSD card (df -h). Insert the USB stick with the card. Identify the backup microSD card, usually sda (fdisk -l). Format the backup microSD card: mkfs.vfat /dev/sda1 && mkfs.ext4 /dev/sda2. Then issue the following command to clone the primary microSD card: rpi-clone -f sda. Tutorial here.

    show-feature-codes is a cheat sheet for all of the feature codes which can be dialed from any registered SIP phone. It documents how powerful a platform Incredible PBX 2022 actually is. A similar listing is available in the GUI at Admin -> Feature Codes.

    show-passwords is a script that displays ALL of the passwords associated with Incredible PBX 2022. This includes SIP extension passwords, voicemail pins, conference pins, telephone reminders pin, and your Anveo Direct outbound calling pin (if configured). Note that voicemail pins are configured by the user of a SIP extension the first time the user accesses the voicemail system by dialing *97.

    timezone-setup lets you reconfigure the correct time zone for your server.

    purge-cdr-cel-records cleans out all existing entries in both the CDR and CEL tables of the Asterisk CDR database.

    log-cleanup removes all entries from most of the logs in /var/log.

    sig-fix disables module signature checking in FreePBX. It is automatically disabled upon installation.

    readme-RonR.txt documents the scripts provided from RonR build. We do NOT recommend using the FCC Blacklist because of its current size.

    update-asterisk16 is a utility that updates Asterisk 16 to the latest release. This should only be necessary when a security issue or bug is identified that affects the operation of your PBX.

    update-IncrediblePBX is the Automatic Update Utility which checks for server updates from incrediblepbx.com every time you log into your server as root using SSH or Putty. Do NOT disable it as it is used to load important fixes and security updates when necessary. We recommend logging into your server at least once a week.

    pbxstatus (shown above) displays status of all major components of Incredible PBX 2022.

    Forwarding Calls to Your Cellphone. Keep in mind that inbound calls to your DIDs automatically ring all five SIP extensions, 701-705. The easiest way to also ring your cellphone is to set one of these five extensions to forward incoming calls to your cellphone. After logging into your PBX as root, issue the following command to forward calls from extension 705 to your cellphone: asterisk -rx "database put CF 705 6781234567"

    To remove call forwarding: asterisk -rx "database del CF 705"

    Keeping FreePBX 15 Modules Current

    We strongly recommend that you periodically update all of your FreePBX modules to eliminate bugs and to reduce security vulnerabilities. Make a backup image with rpi-clone first! From the Linux CLI, log into your server as root and issue the following commands:

    rm -f /tmp/*
    fwconsole ma upgradeall
    fwconsole reload
    /root/sig-fix
    systemctl restart apache2
    /root/sig-fix
    

    Upgrading Asterisk 16 to Asterisk 18

    For those that enjoy living on the bleeding edge, we’ve create a script which makes it easy to upgrade Incredible PBX 2022 to Asterisk 18. The tutorial is available on the new Incredible PBX Wiki along with dozens of other tutorials.

    Resolving an Expired Certificate Alert

    1. Navigate to Admin -> Certificate Management in the FreePBX GUI
    2. Click the Trashcan to delete the Self-Signed Certificate
    3. Click New Certificate -> Generate Self-Signed Certificate
    4. In the Description field, type: Default
    5. Click Generate Certificate button

    Continue Reading: Icing on the Cake for Incredible PBX and Raspberry Pi

    Now Available: Amazon’s Polly TTS for Incredible PBX. Works great on the RasPi platform!

    Originally published: Tuesday, March 24, 2022  Updated: Monday, February 22, 2021



    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.
     



    Systems Integration and Public Participation with FreePBX



    It’s been an interesting few days in the FreePBX® VoIP community with a thread on the VoIP-Info.org and another on the FreePBX Forum. It’s prompted us to revisit what open source development is all about and what all of this means to those of you that rely upon Asterisk® and FreePBX.

    After the departure of the Schmooze folks from Sangoma, virtually all development has been moved behind closed doors with the first opportunity for public participation occurring after new features appear in modules pushed to the so-called Edge repository. This is where folks can shake the kinks out of modules that presumably are almost ready for prime time. For example, take a look at the Changelog for the Framework Module. Prior to the departure of the Schmooze team, changes were managed by openly-accessible tickets. But now you will notice tickets bear a FREEI designation indicating restricted Sangoma-internal access only.


    Beta previously was an appropriate moniker for these modules. If you read the FreePBX forum post above, you’ll note that now modules are being pushed to the Edge repository not only before public comment but apparently before much of any internal testing is performed by the Sangoma folks. The net result is you’d be crazy to ever use Edge modules in any production environment where, in the past, Edge modules more typically were used to fix something that was already broken in the traditional repository. Because the Bootstrap module is the lynchpin of virtually all other FreePBX modules, a recent glitch that had not been tested broke almost everything in FreePBX 16 if you happened to be using Edge modules.

    I’m reminded of the old adage about marine aquariums. You never want to put a new fish in your main tank unless you’re willing to risk killing all of your other fish. This latest fiasco prompts our cautionary note about further use of the FreePBX Edge repository. Don’t load new modules on your production servers without testing them first in a VirtualBox sandbox unless you have a snapshot or backup of your server that can be deployed in minutes when something cataclysmic occurs.

    And that brings us to our response concerning what Incredible PBX® is and is not all about. Going back to the early 80’s, we began tweaking hardware and software deployments to eliminate much of the pain associated with deployment of automated systems. Our friend on the VoIP-Info forum refers to Incredible PBX as a fork of FreePBX. Quite the contrary, it is anything but a fork. We use the FreePBX GPL modules exactly as they are published by Sangoma with an additional master key not controlled by Sangoma for your protection and for ours. This allows us to block specific module updates that prove to be dangerous for our users. Equally important, Incredible PBX offers improved functionality and stability, the same features that are typically associated with the work of a systems integrator… that you pay for. On the stability side, we migrated to new FreePBX repositories maintained by Clearly IP simply because the FreePBX repos had become extremely unreliable and proprietary. Sangoma deploys modules using key signatures that only they control. That means if you make any improvements or changes the FreePBX Dashboard displays all sorts of security alerts, something your customers and end-users would prefer not to see. Some of you may recall this was our primary objection to module signatures years ago.

    On the enhancement side, we’ve tried to add free components that our millions of readers have clamored for. You may recall that Nerd Vittles was the first to provide turnkey Google Voice support for Asterisk over a decade ago. We then tackled security after numerous compromises of FreePBX systems around the world. Since then the list has grown exponentially. Here’s the added feature set you’ve enjoyed by deploying an Incredible PBX platform instead of the FreePBX Distro. And, unlike FreePBX commercial modules, these components won’t cost you a dime and are freely distributable. So the choice is yours.

    • User-customizable installers for Rocky 8, Debian 10 & 11, Ubuntu 20.04, Raspbian
    • User-customizable images for VirtualBox, VMware, Proxmox, and Raspberry Pi
    • Preconfigured, free faxing with HylaFax and AvantFax
    • Preconfigured, secure IPtables firewall
    • Preconfigured, secure Fail2Ban
    • Preconfigured PortKnocker
    • Preconfigured NeoRouter VPN
    • Preconfigured OpenVPN
    • Preconfigured, secure WebMin
    • Preconfigured SendMail and Postfix
    • Preconfigured ODBC Integration for FreePBX
    • Sample ODBC Database Lookups for FreePBX
    • Dozens of Preconfigured Trunk Provider Setups for FreePBX
    • Preconfigured, secure PUBLIC Access Setups for FreePBX
    • Preconfigured scripts to update CentOS, Debian, Ubuntu, Raspbian, and Rocky
    • Preconfigured scripts to update or upgrade Asterisk
    • Preconfigured scripts to update or upgrade FreePBX
    • Preconfigured scripts to update or upgrade PHP
    • Preconfigured script to implement PPTP
    • Preconfigured script to implement TFTP
    • Preconfigured script to implement SAMBA
    • Preconfigured script to implement Gmail Smarthost for SendMail/Postfix
    • Preconfigured script to implement non-Gmail Smarthost for SendMail/Postfix
    • Automatic Update Utility to resolve bugs and security vulnerabilities
    • Integrated TTS apps: FLITE, Festival, PicoTTS, GoogleTTS, Amazon Polly
    • Integrated TTS apps for Voicemail Transcription
    • Integrated STT apps and samples for Asterisk and FreePBX
    • Integrated Voicemail Transcription for Asterisk with Email Delivery
    • AsteriDex 4 web-based MySQL GUI with FreePBX Dialer & Lookups
    • Telephone and Web-Based Reminders for FreePBX with Email and SMS Alerts
    • TTS News Headlines for FreePBX
    • TTS Weather Forecasts for FreePBX (by ZIP Code and Airport Code)
    • CallerID Superfecta for Asterisk
    • MailCall for Asterisk via TTS
    • SMS Scripts for Skyetel and VoIP.ms Message Blasting
    • Speech-to-Text Directory Assistance for Asterisk
    • Wolfram Alpha TTS for FreePBX
    • U-Rang Screenpop Utility for Asterisk
    • xTide TTS for FreePBX
    • Facebook Messaging Integration
    • Twitter Messaging Integration
    • Skype and Gizmo Telephony Integration
    • Teleyapper Message Broadcasting System for Asterisk
    • Scripts to disable Module Signature Checking with FreePBX
    • Script to configure time zones worldwide
    • Incredible Backup and Restore Utilities for all OS platforms
    • LENNY, The Robocallers Worst Nightmare

    People hear the word "fork" and get scared away from testing alternative VoIP solutions. If you’re one of those folks, we would encourage you to spend an hour with Incredible PBX using one of our supported virtual platforms: VirtualBox, VMware, or Proxmox. We think the decision will be a no-brainer after your testing. For a list of all of our free VoIP solutions, visit the Incredible PBX Wiki. Enjoy!

    Originally published: Monday, May 23, 2022



    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.
     



    5 Minute Wonder: Incredible PBX 2022 in Cloud for $25/Yr.



    We’ve been building turnkey Incredible PBX® servers for virtual machine platforms for many years. Because the servers are built from images, typical install times have been 5 minutes or less with Proxmox, VirtualBox, and VMware ESXi. But the missing piece has been a similar image install on a true cloud-based platform. This limitation was primarily due to the fact that we don’t own or control the available cloud platforms which typically limit image installs to operating systems such as CentOS, Debian, Ubuntu, and Windows. However, CrownCloud was good enough to add our Incredible PBX 2021 Debian image and the Incredible PBX 2020 CentOS 7 image to their portfolio. And, today, we have more good news. CrownCloud has now added the Incredible PBX 2022 image for Rocky 8 as well.

    These 5-minute turnkey installs of Incredible PBX 2020 for CentOS 7, Incredible PBX 2021 for Debian 10, and now Incredible PBX 2022 for Rocky 8 are being offered at the jaw-dropping price of $25 a year. The monthly cost is cheaper than a cup of coffee at Starbucks, and you’ll have a fully-functioning, production-ready KVM platform including a free snapshot with 1GB RAM, 20GB SSD storage, and 1TB of monthly bandwidth in your choice of server locations including Los Angeles and Atlanta in the United States as well as Germany and the Netherlands in Europe. And, unlike all of the other Asterisk® aggregations, Incredible PBX still provides a source code-based platform that can be tailored to meet any special requirements your organization may need.

    We don’t make a nickel on these offerings so consider this our special thanks to all of our loyal fans.

    Here are the links to sign up for the service and take advantage of these Incredible PBX deals:

    When you sign up for the service, choose any traditional OS for the base install. Once it’s on line, go into CrownPanel and choose Reinstall. Then select Incredible PBX 2022 for Rocky from the Application Images pulldown as your new install. In less than 5 minutes, your server will be ready for you to login. Be sure to use SSH and NOT the VNC utility included in CrownPanel. This will assure that your desktop machine’s IP address gets whitelisted in the Incredible PBX firewall. Otherwise, you won’t be able to SSH into your server from your desktop. Once you login, the Incredible PBX configurator will prompt you to set passwords for root login, admin login to FreePBX, and admin login credentials for Apache to access AsteriDex and Reminders. Add a trunk provider (Skyetel is preconfigured and enabled by default) and a softphone or Incredible PBX SIP phone, and your PBX is fully operational. Check out the CrownCloud Wiki.


    Planning Ahead for That Rainy Day

    One of our favorite features of Crown Cloud is the free snapshot (a.k.a. backup) at no additional charge. We recommend you take snapshots regularly as you make major changes in your server’s configuration. In this way, if something comes unglued, you can easily restore the snapshot and never miss a beat. You’ll find the Remote Snapshot option in your CrownPanel menu.

    Configuring Skyetel for Incredible PBX

    If you’ve decided to go with Skyetel, here’s the drill. Sign up for Skyetel service and take advantage of the Nerd Vittles specials. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request the $10 credit for your account by referencing the Nerd Vittles special offer. Once you are satisfied with the service, fund your account as desired, and Skyetel will match your deposit of up to $250 simply by opening another ticket. That gets you up to $500 of half-price calling. Credit is limited to one per person/company/address/location. Effective 10/1/2023, $25/month minimum spend required.

    Skyetel does not require SIP registrations to make connections to your PBX. Instead, Skyetel can use Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. Here’s a typical setup for Incredible PBX:

    • Name: MyPBX
    • Priority: 1
    • IP Address: PBX-Public-IP-Address
    • Port: 5060
    • Protocol: UDP
    • Description: my.incrediblepbx.com

    To receive incoming PSTN calls, you’ll need at least one DID. On the Skyetel site, you acquire DIDs under the Phone Numbers tab. You have the option of Porting in Existing Numbers (free for the first 60 days after you sign up for service) or purchasing new ones under the Buy Phone Numbers menu option.

    Once you have acquired one or more DIDs, navigate to the Local Numbers or Toll Free Numbers tab and specify the desired SIP Format and Endpoint Group for each DID. Add SMS/MMS and E911 support, if desired. Call Forwarding and Failover are also supported. That completes the VoIP setup on the Skyetel side. System Status is always available here. Everything is already in place on the Incredible PBX 2022 side of the house so you can start making and receiving calls immediately.

    Configuring ClearlyIP SIP Trunking

    For the tightest integration with FreePBX, no SIP provider holds a candle to Incredible PBX SIP Trunking with ClearlyIP. The reason is fairly obvious. The ClearlyIP folks were the original developers of FreePBX. In addition to all of the traditional SIP trunking services, you also get CNAM support and state-of-the-art E911 service which can be deployed in full compliance with Kari’s Law and the Ray Baum Act. If you’re a system integrator and don’t know about your financial liability for failure to comply with the new rules, it’s time to do some reading.

    Configuring VoIP.ms for Incredible PBX

    To sign up for VoIP.ms service, may we suggest you use our signup link so that Nerd Vittles gets a referral credit for your signup. Once your account is set up, you’ll need to set up a SIP SubAccount and, for Authentication Type, choose Static IP Authentication and enter your Incredible PBX server’s public IP address. For Transport, choose UDP. For Device Type, choose Asterisk, IP PBX, Gateway or VoIP Switch. Order a DID in their web panel, and then point the DID to the SubAccount you just created. Be sure to specify atlanta1.voip.ms as the POP from which to receive incoming calls. In the Incredible PBX GUI, be sure to enable the VoIP.ms trunk.

    Configuring Anveo Direct for Incredible PBX

    To sign up for Anveo Direct service, sign up on their web site and then login. After adding funds to your account, purchase a DID under Inbound Service -> Order DID. Next, choose Configure Destination SIP Trunk. Give the Trunk a name. For the Primary SIP URI, enter $[E164]$@server-IP-address. For Call Options, select your new DID from the list. You also must whitelist your public IP address under Outbound Service -> Configure. Create a new Call Termination Trunk and name it to match your server. For Dialing Prefix, choose six alphanumeric characters beginning with a zero. In Authorized IP Addresses, enter the public IP address of your server. Set an appropriate rate cap. We like $0.01 per minute to be safe. Set a concurrent calls limit. We like 2. For the Call Routing Method, choose Least Cost unless you’re feeling extravagant. For Routes/Carriers, choose Standard Routes. Write down your Dialing Prefix and then click the Save button.

    Before you can make outbound calls through Anveo Direct from your PBX, you first must configure the Dialing Prefix that you wrote down in the previous step. Log into the GUI as admin using a web browser and edit the Anveo-Out trunk in Connectivity -> Trunks. Enable the Trunk. Then click on the custom-Settings tab and replace anveo-pin with your actual Dialing Prefix. Click Submit and Apply Config to complete the setup. In the Incredible PBX GUI, be sure to enable all of the remaining Anveo trunks.

    By default, incoming Anveo Direct calls will be processed by the Default inbound route on your PBX. If you wish to redirect incoming Anveo Direct calls using DID-specific inbound routes, then you’ve got a bit more work to do. In addition to creating the inbound route using the 11-digit Anveo Direct DID, enter the following commands after logging into your server as root using SSH/Putty:

    cd /etc/asterisk
    echo "[from-anveo]" >> extensions_custom.conf
    echo "exten => _.,1,Ringing" >> extensions_custom.conf
    echo "exten => _.,n,Goto(from-trunk,\\${SIP_HEADER(X-anveo-e164)},1)" >> extensions_custom.conf
    asterisk -rx "dialplan reload"
    

    Configuring a SIP Phone for Incredible PBX 2022

    We’re in the home stretch now. You can connect virtually any kind of telephone to your new PBX. Plain Old Phones require an analog telephone adapter (ATA) which is a standalone SIP device such as ObiHai’s OBi100 or OBi110 (if you have a phone line from Ma Bell to hook up as well). SIP phones can be connected directly so long as they have an IP address. We obviously recommend the Incredible PBX IP phones from ClearlyIP which are the most versatile.

    If you’ve been keeping up with recent Nerd Vittles developments, then you already know that we have just introduced a new Cellular Phone which connects directly to your PBX and serves as a perfect remote extension and traveling companion. You can read all about it here.

    Software devices such as the YateClient softphone are another option for desktop machines. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the VoIP-Info.org Forum when you’re ready to get serious about VoIP telephony.

    We recommend YateClient for Windows which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You can find them by running /root/show-passwords. You’ll need the IP address of your server plus your extension 701 password. In the YateClient, fill in the blanks using the IP address of your Server plus :5061 for the PJsip 701 extension, 701 for your Username, and whatever Password was assigned to the extension when you installed Incredible PBX. Click OK to save your entries.

    Once you are registered to extension 701, close the Account window. Then click on YATE’s Telephony Tab and place some test calls to the numerous apps that are preconfigured on Incredible PBX. Dial a few of these to get started:

    DEMO - Apps Demo
    123 - Reminders
    947 - Weather by ZIP Code
    951 - Yahoo News
    TODAY - Today in History
    LENNY - The Telemarketer's Worst Nightmare
    

    If you are a Mac user, another great no-frills softphone is Telephone. Just download and install it from the Mac App Store. For Android users, check out the terrific new VitalPBX Communicator. Works flawlessly with Incredible PBX.

    For smartphone solutions, visit the Incredible PBX Wiki for our softphone recommendations.

    Configuring SendMail with Incredible PBX

    In order to receive voicemails by email delivery, outbound mail functionality from your server obviously is required. We strongly recommend configuring SendMail using either your ISP or Gmail as an SMTP Relay Host. NOTE: If you are using a Gmail account with 2-step verification enabled, you MUST use a Gmail App Key instead of your Gmail account password. You also must enable Less Secure Apps access to the Gmail account.

    Configuring a Gmail account with Incredible PBX 2022 is as simple as entering your Gmail credentials. Just run this script: /root/enable-gmail-smarthost-for-sendmail.

    Here are the steps using a Gmail account with Incredible PBX 2020:

    cd /etc/mail
    yum -y install sendmail-cf
    hostname -f > genericsdomain
    touch genericstable
    cd /usr/bin
    rm -f makemap
    ln -s ../sbin/makemap.sendmail makemap
    cd /etc/mail
    makemap -r hash genericstable.db < genericstable
    mv sendmail.mc sendmail.mc.original
    wget http://incrediblepbx.com/sendmail.mc.gmail
    cp sendmail.mc.gmail sendmail.mc
    mkdir -p auth
    chmod 700 auth
    cd auth
    echo AuthInfo:smtp.gmail.com \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" > client-info
    echo AuthInfo:smtp.gmail.com:587 \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" >> client-info
    echo AuthInfo:smtp.gmail.com:465 \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" >> client-info
    # Stop here and edit client-info (nano -w client-info) in all three lines.
    # Replace  user_id with your gMail account name without @gmail.com
    # Replace password with your real gMail password OR
    #  use your Gmail App Key if 2-step verification is enabled
    # Be sure to replace the double-quotes shown above if they don't appear in the file!!!
    # Save your changes (Ctrl-X, Y, then Enter)
    chmod 600 client-info
    makemap -r hash client-info.db < client-info
    cd ..
    make
    systemctl restart sendmail
    

    Even though these servers are hosted in the cloud, we still recommend using a SmartHost to minimize email delivery problems.

    Test outbound mail using this command with your actual email address:

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

    On some implementations, you may notice in the FreePBX GUI that the mail queue has failed. Here's the fix:

    chmod 777 /var/spool/mqueue
    service sendmail restart
    

    Once you are sure your emails are being delivered reliably, here's a sample GUI voicemail configuration for an extension:



    Be advised that Google has hinted that the Gmail Smarthost landscape may be changing. See our recent article for a simple SmartHost alternative.

    Incredible PBX Administration

    We've eased the pain of administering your new PBX with a collection of scripts which you will find in the /root folder after logging in with SSH or Putty. Here's a quick summary of what each of the scripts does.

    add-fqdn is used to whitelist a fully-qualified domain name in the firewall. Because Incredible PBX blocks all traffic from IP addresses that are not whitelisted, this is what you use to authorize an external user for your PBX. The advantage of an FQDN is that you can use a dynamic DNS service to automatically update the IP address associated with an FQDN so that you never lose connectivity.

    add-ip is used to whitelist a public IP address in the firewall. See the add-fqdn explanation as to why this matters.

    del-acct is used to remove an IP address or FQDN from the firewall's whitelist.

    admin-pw-change is used to set the admin password for access to the FreePBX/Incredible PBX web GUI using a browser pointed to the local IP address of your server.

    apache-pw-change is used to set the admin password for access to Apache/Incredible PBX apps including AsteriDex and Reminders. This provides a password layer of protection for access to these applications.

    reset-conference-pins is a script that automatically and randomly resets the user and admin pins for access to the preconfigured conferencing application. Dial C-O-N-F from any registered SIP phone to connect to the conference.

    reset-extension-passwords is a script that automatically and randomly resets ALL of the SIP passwords for extensions 702-705. Be careful using this one, or you may disable existing registered phones and cause Fail2Ban to blacklist the IP addresses of those users. HINT: You can place a call to the Ring Group associated with all five extensions by dialing 777.

    reset-reminders-pin is a script that automatically and randomly resets the pin required to access the Telephone Reminders application by dialing 123. It's important to protect this application because a nefarious user could set up a reminder to call a number anywhere in the world assuming your SIP provider's account was configured to allow such calls.

    show-feature-codes is a cheat sheet for all of the feature codes which can be dialed from any registered SIP phone. It documents how powerful a platform Incredible PBX actually is. A similar listing is available in the GUI at Admin -> Feature Codes.

    show-passwords is a script that displays most of the passwords associated with Incredible PBX. This includes SIP extension passwords, voicemail pins, conference pins, telephone reminders pin, and your Anveo Direct outbound calling pin (if configured). Note that voicemail pins are configured by the user of a SIP extension the first time the user accesses the voicemail system by dialing *97.

    ssh-regen.sh allows you to reset the SSH keys for your server for added security.

    update-IncrediblePBX is the Automatic Update Utility which checks for server updates from incrediblepbx.com every time you log into your server as root using SSH or Putty. Do NOT disable it as it is used to load important fixes and security updates when necessary. We recommend logging into your server at least once a week.

    pbxstatus (shown above) displays status of all major components of Incredible PBX.

    Forwarding Calls to Your Cellphone. Keep in mind that inbound calls to your DIDs automatically ring all five SIP extensions, 701-705. The easiest way to also ring your cellphone is to set one of these five extensions to forward incoming calls to your cellphone. After logging into your PBX as root, issue the following command to forward calls from extension 705 to your cellphone: asterisk -rx "database put CF 705 6781234567"

    To remove call forwarding: asterisk -rx "database del CF 705"

    Keeping FreePBX 15 Modules Current

    We strongly recommend that you periodically update all of your FreePBX modules to eliminate bugs and to reduce security vulnerabilities. From the Linux CLI, log into your server as root and issue the following commands:

    rm -f /tmp/*
    fwconsole ma upgradeall
    fwconsole reload
    /root/sig-fix
    /root/sig-fix
    

    Where To Go From Here

    Complete documentation on the ClearlyIP Devices Module is available here.

    Complete documentation on the FreePBX GPL Modules is available here.

    Complete documentation on the Incredible PBX additions is available here.

    An introduction to configuring extensions, trunks, and routes is available here.

    Free voicemail transcription with email delivery. Tutorial available here.

    Setting Up a VPN for Your PBX: OpenVPN or NeoRouter
     

    Originally published: Monday, March 28, 2022



    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.
     



    One-Minute Wonder: It’s Incredible PBX 2022 for VirtualBox




    If you’re new to the VoIP world and want to kick the tires to see what you’re missing, then today’s one minute setup is for you. You can use almost any desktop computer you already own to bring up the VirtualBox® edition of Incredible PBX® 2022.

    If you’ve followed Nerd Vittles over the years, you already know that VirtualBox from Oracle® is one of our favorite platforms. Once VirtualBox is installed on your desktop computer, adding Incredible PBX is a snap. Download the latest Incredible PBX 2022 image from SourceForge, double-click on the downloaded image, check the initialize MAC address box, and boom. In less than a minute, your PBX is ready to use with the very latest Rocky 8 platform and Asterisk® 18 build plus all of the FreePBX® 15 GPL modules. There are no hidden fees or crippleware to hinder your use of Incredible PBX for as long as you like. If you set up an account with our Platinum provider, Skyetel, you can start making calls in minutes. Of course, the Incredible PBX feature set is included as well which brings you nearly three dozen applications for Asterisk® that will revolutionize your communications platform. Speech-to-text, voice recognition, and a Siri-like telephony interface are as close as your SIP phone.

    Installing Oracle VM VirtualBox

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

    NOTE: The Incredible PBX 2022 VM requires a VirtualBox 6.x platform. Adjust screen size in View -> Virtual Screen.

    Installing the Incredible PBX 2022 Image

    To begin, download the Incredible PBX 2022 image (3.5 GB) onto your desktop.

    Next, double-click on the Incredible PBX .ova image on your desktop. Be sure to check the box to initialize the MAC address of the image if you’re using an older version of VirtualBox. Then click Import. Once the import is finished, you’ll see a new Incredible PBX 2022 virtual machine in the VM List of the VirtualBox Manager Window. Let’s make a couple of one-time adjustments to the Incredible PBX configuration to account for possible differences in sound and network cards on different host machines.

    (1) Click once on the Incredible PBX virtual machine in the VM List. Then (2) click the Settings button. In System tab, verify Hardware Clock in UTC Time is checked. In the Audio tab, check the Enable Audio option and choose your sound card. In the Network tab for Adapter 1, check the Enable Network Adapter option. From the Attached to pull-down menu, choose Bridged Adapter. Then select your network card from the Name list. Then click OK. That’s all the configuration that is necessary for Incredible PBX 2022.

    Running Incredible PBX 2022 in VirtualBox

    Once you’ve imported and configured the Incredible PBX 2022 Virtual Machine, you’re ready to go. Highlight the Incredible PBX 2022 virtual machine in the VM List on the VirtualBox Manager Window and click the Start button. The standard Linux boot procedure will begin and, within a few seconds, you’ll get the familiar Linux login prompt. During the bootstrap procedure, you’ll see a couple of dialogue boxes pop up that explain the keystrokes to move back and forth between your host operating system desktop and your virtual machine. Remember, you still have full access to your desktop computer. Incredible PBX is merely running as a task in a VM window. Always gracefully halt Incredible PBX just as you would on any computer.




     
    Here’s what you need to know. To work in the Incredible PBX virtual machine, just left-click your mouse while it is positioned inside the VM window. To return to your host operating system desktop, press the right Option key on Windows machines or the left Command key on any Mac. On Linux desktops, press the right Ctrl key. For other operating systems, read the dialogue boxes for instructions on moving around. To access the Linux CLI, login as root with the default password: password. Change your root password when you are prompted to do so. Then update your admin password for web access: ./admin-pw-change. Also update your admin password for web applications: ./apache-pw-change. You’ll need these admin passwords to access the web GUI to manage your PBX as well as to use the AsteriDex and Reminders web apps. The above password updates are automatically requested when you first activate the virtual machine. You can update all of your other passwords using the scripts provided in /root.

    Setting the Date and Time with VirtualBox

    On some platforms, VirtualBox has a nasty habit of mangling the date and time of your virtual machine. Verify that you have enabled the Hardware Clock in UTC Time option for your virtual machine as documented above. If pbxstatus still shows an incorrect time, manually set the date and time and then update the hardware clock. Here’s how assuming 08130709 is the month (August), day (13), and correct time (7:09 a.m.) of your server:

    date 08130709
    clock -w
    

    Configuring Skyetel for Incredible PBX 2022

    If you’d like to try out the Skyetel service at no charge, here’s the drill. Sign up for Skyetel service to take advantage of the Nerd Vittles specials. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request the $10 credit for your account by referencing the Nerd Vittles special offer. Once you are satisfied with the service, fund your account as desired, and Skyetel will match your deposit of up to $250 simply by opening another ticket. That gets you up to $500 of half-price calling. Credit is limited to one per person, company, and address. Effective 10/1/2023, $25/month minimum spend required.

    Skyetel does not use SIP registrations to make connections to your PBX. Instead, Skyetel utilizes Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. Here’s a typical setup for Incredible PBX 2022:

    • Name: MyPBX
    • Priority: 1
    • IP Address: PBX-Public-IP-Address
    • Port: 5061
    • Protocol: UDP
    • Description: my.incrediblepbx.com

    To receive incoming PSTN calls, you’ll need at least one DID. On the Skyetel site, you acquire DIDs under the Phone Numbers tab. You have the option of Porting in Existing Numbers (free for the first 60 days after you sign up for service) or purchasing new ones under the Buy Phone Numbers menu option.

    Once you have acquired one or more DIDs, navigate to the Local Numbers or Toll Free Numbers tab and specify the desired SIP Format and Endpoint Group for each DID. Add SMS/MMS and E911 support, if desired. Call Forwarding and Failover are also supported. That completes the VoIP setup on the Skyetel side. System Status is always available here.

    If VirtualBox is sitting behind a router or firewall on a private LAN, you’ll need to forward ports UDP 5060, 5061, and 10000-20000 in your router to the private LAN address of your Incredible PBX server. Also edit your extensions in the GUI and set NAT=YES in the Advanced tab of every extension. In Settings -> Asterisk SIP Settings, click the Detect Network Settings button and then Submit your changes and reload the Asterisk dialplan when prompted.

    Finally, login to the FreePBX web GUI as admin using the password you assigned when you set up the virtual machine. Navigate to Connectivity -> Trunks and edit the Skyetel-pjSIP trunk. Change the Disable Trunk setting from Yes to No. Then click Submit and reload your dialplan when prompted. That’s it.

    Configuring VoIP.ms for Incredible PBX 2022

    To sign up for VoIP.ms service, may we suggest you use our signup link so that Nerd Vittles gets a referral credit for your signup. Once your account is set up, you’ll need to set up a SIP SubAccount and, for Authentication Type, choose Static IP Authentication and enter your Incredible PBX 2020 server’s public IP address. For Transport, choose UDP. For Device Type, choose Asterisk, IP PBX, Gateway or VoIP Switch. Order a DID in their web panel, and then point the DID to the SubAccount you just created. Be sure to specify atlanta1.voip.ms as the POP from which to receive incoming calls. For more details about VoIP.ms, see this Nerd Vittles tutorial.

    Configuring SendMail with Incredible PBX 2022

    In order to receive voicemails by email delivery, outbound mail functionality from your server obviously is required. If you’ve deployed your server in your home, your Internet Service Provider probably blocks downstream mail servers such as Incredible PBX from sending mail. This is done to reduce SPAM. In this case, you will need to configure SendMail using either your ISP or Gmail as an SMTP Relay Host. We have built aninstall script to set up a SmartHost using Gmail. Simply run it and insert your Gmail username and password or App Password.

    cd /root
    wget http://incrediblepbx.com/enable-gmail-smarthost-rocky8.tar.gz
    tar zxvf enable-gmail-smarthost-rocky8.tar.gz
    rm -f enable-gmail-smarthost-rocky8.tar.gz
    ./enable-gmail-smarthost-for-sendmail
    

    Configuring a Softphone for Incredible PBX 2022

    We’re in the home stretch now. You can connect virtually any kind of telephone to your new PBX. Plain Old Phones require an analog telephone adapter (ATA) which can be a separate board in your computer from a company such as Digium. Or it can be a standalone SIP device such as the Incredible PBX SIP phone. SIP phones can be connected directly so long as they have an IP address. These could be hardware devices or software devices such as the YateClient softphone. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and in the Incredible PBX Wiki when you’re ready to get serious about VoIP telephony.

    We recommend YateClient for Windows which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You can find them by navigating to Applicaations -> Extensions -> 701 in the FreePBX GUI.

    Configuring Incredible PBX 2022 for VirtualBox

    In order to take advantage of all the Incredible PBX applications, you’ll need to obtain IBM text-to-speech (TTS) and speech-to-text (STT) credentials as well as a (free) Application ID for Wolfram Alpha.

    This Nerd Vittles tutorial will walk you through getting your IBM account set up and obtaining both your TTS and STT credentials. Be sure to write down BOTH sets of credentials which you’ll need in a minute. For home and SOHO use, IBM access and services are mostly FREE even though you must provide a credit card when signing up. The IBM signup process explains their pricing plans.

    To use Wolfram Alpha, sign up for a free Wolfram Alpha API account. Just provide your email address and set up a password. It takes less than a minute. Log into your account and click on Get An App ID. Make up a name for your application and write down (and keep secret) your APP-ID code. That’s all there is to getting set up with Wolfram Alpha. If you want to explore costs for commercial use, there are links to let you get more information.

    In addition to your Wolfram Alpha APPID, there are two sets of IBM credentials to plug into the Asterisk AGI scripts. Keep in mind that there are different usernames and passwords for the IBM Watson TTS and STT services. The TTS credentials will look like the following: $IBM_username and $IBM_password. The STT credentials look like this: $API_USERNAME and $API_PASSWORD. Don’t mix them up. 🙂

    All of the scripts requiring credentials are located in /var/lib/asterisk/agi-bin so switch to that directory after logging into your server as root. Edit each of the following files and insert your TTS credentials in the variables already provided: nv-today2.php, ibmtts.php, and ibmtts2.php. Edit each of the following files and insert your STT credentials in the variables already provided: getquery.sh, getnumber.sh, and getnumber2.sh. Finally, edit 4747 and insert your Wolfram Alpha APPID.

    Using AsteriDex with Incredible PBX

    AsteriDex is a web-based dialer and address book application for Asterisk and Incredible PBX. It lets you store and manage phone numbers of all your friends and business associates in an easy-to-use SQLite3 database. You simply call up the application with your favorite web browser: http://pbx-ip-address/asteridex4/. When you click on a contact that you wish to call, AsteriDex first calls you at extension 701, and then AsteriDex connects you to your contact through another outbound call made using your default outbound trunk that supports numbers in the 1NXXNXXXXXX format.

    Keeping FreePBX 15 Modules Current

    We strongly recommend that you periodically update all of your FreePBX modules to eliminate bugs and to reduce security vulnerabilities. From the Linux CLI, log into your server as root and issue the following commands:

    rm -f /tmp/*
    fwconsole ma upgradeall
    fwconsole reload
    /root/sig-fix
    /root/sig-fix
    

    Taking Incredible PBX 2022 for a Test Drive

    You can take Incredible PBX on a test drive by dialing D-E-M-O (3366) from any phone connected to your PBX.

    With Allison’s Demo IVR, you can choose from the following options:

    • 0. Chat with Operator — connects to extension 701
    • 1. AsteriDex Voice Dialer – say "Delta Airlines" or "American Airlines" to connect
    • 2. Conferencing – log in using 1234 as the conference PIN
    • 3. Wolfram Alpha Almanac – say "What planes are flying overhead"
    • 4. Lenny – The Telemarketer’s Worst Nightmare
    • 5. Today’s News Headlines — courtesy of Yahoo! News
    • 6. Weather by ZIP Code – enter any 5-digit ZIP code for today’s weather
    • 7. Today in History — courtesy of OnThisDay.com
    • 8. Chat with Nerd Uno — courtesy of SIP URI connection to 3CX iPhone Client
    • 9. DISA Voice Dialer — say any 10-digit number to be connected
    • *. Current Date and Time — courtesy of Incredible PBX

    We Missed You During February

    If you missed us last month, we missed you, too. We took a brief timeout to get some new eyeballs. Ah, the miracles of modern medicine. As the old song says, "I Can See Clearly Now." It’s been 35 years since I saw the world without the need for glasses. It’s good to be back.



    Originally published: Monday, March 7, 2022



    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.
     



    1. Many of our purchase links refer users to Amazon when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from Amazon to help cover the costs of our blog. We never recommend particular products solely to generate Amazon commissions. However, when pricing is comparable or availability is favorable, we support Amazon because Amazon supports us. []

    Introducing OpenSIPS 3 for Incredible PBX and Debian 10


    Today we’re pleased to introduce an updated OpenSIPS installer for Debian 10 featuring the latest release of OpenSIPS. Our previous tutorial with Debian 8 is now obsolete, an all-too-frequent occurrence in the open source world. Today’s open source SIP server lets you connect users to make and receive free as well as commercial calls worldwide. There’s excellent documentation making it easy to integrate into our existing Incredible PBX platform without hiring a consultant. It’s also straight-forward to secure without providing free phone service to every bad guy on the planet.

    OpenSIPS is a multi-functional, multi-purpose signaling SIP server used by carriers, telecoms or ITSPs for solutions like Class4/5 Residential Platforms, Trunking / Wholesale, Enterprise / Virtual PBX Solutions, Session Border Controllers, Application Servers, Front-End Load Balancers, IMS Platforms, Call Centers, and many others. Source: opensips.org

    We’ve often complained that the problem with many open source projects is that the developers get so focused on making money that they skimp on the documentation to encourage consulting work or participation in expensive conferences. We have found just the opposite with OpenSIPS. In fact, much of today’s implementation is based upon an excellent tutorial by the folks at PowerPBX. Down the road, if you find yourself in need of a consultant, their services would be a good place to start. What we’ve added to the PowerPBX design is security, support for clients behind NAT-based routers, and an integration scheme for Asterisk®, FreePBX®, and Incredible PBX® platforms so that you get the best of all worlds, a public facing SIP server with the UC feature set that most organizations expect. Last but not least, our turnkey GPLv2 installer will get you up and running in about 5 minutes.

    Choosing an Appropriate Platform for OpenSIPS

    Let’s begin by addressing the appropriate platform for an OpenSIPS server. The server needs to have a public IP address that is static, and the server should not be situated behind a NAT-based router. It only complicates things and is beyond the scope of what we plan to address. For those that are frequent visitors, you already know that we’ve been pushing everyone to kiss their local hardware goodbye and join the cloud revolution. When it comes to public-facing VoIP platforms like OpenSIPS, most of us don’t have a choice. You need a static IP address on the open Internet. And, for the sake of security, a KVM cloud platform is a must since older OpenVZ platforms don’t support the ipset component of IPtables which makes it easy to block hundreds of thousands of IP addresses without a performance hit on your server. Pure whitelist access simply isn’t an option if you wish to retain the functionality of a VoIP application such as OpenSIPS.

    Ten to twenty gigabytes of disk space should be more than ample for OpenSIPS. The amount of RAM in your server depends upon the volume of calls your server will be handling. If it’s a dozen simultaneous calls then 1GB of RAM will suffice. If it’s 100,000 calls, then take a look at this article for tips on sizing your server. For today’s implementation, you’ll need a Debian 10 platform so a low-cost KVM provider including Digital Ocean, Vultr, and OVH should be fine.1

    Choosing OpenSIPS Components to Deploy

    We’ve divided up today’s tutorial into bite-sized pieces so that you can pick and choose where to stop implementing and start using. You do not need to have an Asterisk server to make and receive calls with OpenSIPS. However, OpenSIPS lacks voicemail and AutoAttendant/IVR components so, if those are a requirement, then you either need a VoIP service provider that offers them, or deploy a $50 Incredible PBX for the Raspberry Pi to add the missing pieces.

    What OpenSIPS offers is a free server platform for worldwide SIP communications so that you, your friends, and business associates can call or connect from anywhere using freely available SIP softphones or any of dozens of SIP telephone instruments. We’ll stick with softphones for today, but hardware-based SIP telephones are equally simple to deploy.

    This is not a criticism because it is one of the best tutorials we’ve ever used but, if you want to see how complex a typical OpenSIPS server deployment is, take a look at the PowerPBX tutorial we used as a starting point with OpenSIPS. We’ve compressed most of those procedures into a turnkey installer that only requires you to enter a MySQL root password of passw0rd (with a zero) once you have your Debian 10/64 platform up and running.

    Deploying a Debian 10 Server Platform

    Start by choosing a cloud provider that offers the 64-bit Debian 10 minimal platform as a deployment option. Most do. As noted, we recommend a KVM platform with support for ipset making it easy to block entire countries overrun with bad guys. Choose offerings with at least 1GB RAM and a 10GB drive to get started. Configure your Debian 10 server with a fully-qualified domain name (FQDN). This is critically important with our security design because we will assign all OpenSIPS users/extensions to this FQDN and reserve your server’s IP address purely for connections from service providers and Asterisk servers. This makes it all but impossible for anyone to hack into your server since most script kiddies launch attacks on IP addresses, not FQDNs. Using an unusual FQDN adds an extra layer of security, but that’s your call. If you lack the ability to assign FQDN aliases to a domain which you own, you can obtain a free FQDN from numerous sources including ChangeIP and point it to the IP address of your OpenSIPS server.

    Installing OpenSIPS on a Debian 10 Server

    Now the fun begins. Log into your Debian 8 server as root and issue the following commands to prepare for the OpenSIPS install:

    cd /root
    wget http://incrediblepbx.com/opensips3.tar.gz
    tar zxvf opensips3.tar.gz
    rm -f opensips3.tar.gz
    

    Make sure you have logged into your Debian 10 server as root using SSH or Putty from a desktop PC that you will use to manage OpenSIPS with a browser. The reason is because this IP address automatically will be whitelisted in the OpenSIPS firewall as part of the install process. Otherwise, you will need to manually log into SSH and whitelist the IP address of your desktop PC using /root/add-ip each time you wish to access the OpenSIPS Control Panel since TCP port 80 (HTTP) is not exposed to the public Internet as a security precaution.

    To begin the install, issue this command: /root/install

    As the install progresses, you’ll first be prompted to choose the GRUB install device. Press the spacebar on the first entry. Then press TAB and ENTER. When prompted for the SSH configuration, choose "keep local version" and then press TAB and ENTER. For the MariaDB setup, press ENTER when prompted for the current password. Type N when prompted whether to switch to unix_socket authorization. Then type Y to change the root password. Be sure to use passw0rd (with a zero) as your MySQL password, or the install will fail. This is NOT a security risk unless your Debian 10 root user account is compromised. And, in that case, it won’t matter anyway since the MySQL password could easily be changed. Type Y to remove anonymous users. Type Y to disallow remote root logins. Type Y to reload the MySQL privilege tables.

    Next you’ll be prompted to set your timezone and TZ entries. For East Coast U.S., it’s 2,49,1,1 then America/New_York. Later you’ll be prompted twice for the MySQL root password. You must enter passw0rd (with a zero). When the OpenSIPS status screen displays, type Q to exit the display. There are a couple of steps where you will be prompted for input. Correct responses are indicated before the various prompts. Pay particular attention when you are prompted to change the SSH port from TCP 22 to a port number in the 1000-2020 range as a security precaution. We recommend using the year you were born because it will be easy for you to remember. When the install finishes and you log out of your server, the next SSH login will look like this where XXXX is the SSH port you chose and yyy.yyy.yyy.yyy is the OpenSIPS server address: ssh -p XXXX root@yyy.yyy.yyy.yyy


    Although most of the configuration of your OpenSIPS server will be handled using a web browser and the OpenSIPS Control Panel GUI, we’ve included a few scripts in /root to assist with maintenance of your server platform. Here’s a brief summary of the script functions:

    • pbxstatus – Status of your OpenSIPS server (image sample above)
    • add-ip – Temporarily WhiteList IP address until next iptables-restart
    • ban-ip – Permanently Ban an IP address
    • unban-ip – Unban a previously banned IP address
    • log-purge – Zero out all of the major Linux log files
    • opensips-check – Assures OpenSIPS and RTPproxy are running (runs automatically)
    • Fail2Ban BlackListsiptables -nL | grep -A100000 "opensips ("
    • IPset BlackList (KVM/OVZ7 platforms only) – ipset list | sort

    We secure your server in several ways: (1) by disguising the SSH port, (2) by locking down almost every port on your server with the IPtables firewall with the exception of the SIP ports, (3) by deploying Fail2Ban to scan your OpenSIPS log for errors and lock out attackers for an extended period of time, and (4) by deploying the IPset blacklist for KVM platforms. With this design, there is a symbiotic relationship between IPtables, Fail2Ban, and IPset. Therefore, it is critically important that you only restart these services using the iptables-restart command. NEVER issue other IPtables commands to restart or save your firewall settings.

    Activating a SIP Server with OpenSIPS Control Panel

    We don’t want to overload you on the first day with your new OpenSIPS 3 platform so we’ll walk you through the preliminary setup steps to create your SIP Domain. Then we’ll show you how to set up user accounts (also known as extensions). Finally we’ll walk you through setting up a trunk to make and receive calls from a commercial SIP provider. When we’re finished today, you’ll be able to make and receive calls using SIP URIs or DIDs which you have purchased from a provider. Then next week we’ll focus on integration of OpenSIPS with an Asterisk platform of your choice using Incredible PBX as an example. Once we’re finished, you’ll be able to handle user account registrations exclusively on your OpenSIPS server while leaving your Asterisk platform completely hidden from public exposure.

    Logging into the OpenSIPS Control Panel

    As deployed, the OpenSIPS Control Panel is accessible via web browser. As noted previously, HTTP Port 80 access is blocked by default unless the IP address of your desktop PC has been whitelisted either as part of the initial install or using the add-ip script in /root. Once your desktop PC’s IP address is whitelisted, point your browser to http://xxx.xxx.xxx.xxx/cp



    The default Username is admin, and the default password is opensips. Once you’re logged in, immediately click on the Users icon in the upper-right corner of the dashboard. Then click the Edit Info pencil icon for user Admin and change your password. Click Save when done.

    Creating Domains with OpenSIPS Control Panel

    In the Left column of the Dashboard, you’ll see two tabs: Users and System. Click on the System tab to expose the available choices. Then choose the Domains option.



    Domains are the essential building blocks in OpenSIPS. You can manage one or a hundred domains on a single OpenSIPS server, and each domain can have its own set of Users, Trunks/Gateways, and Dialplan rules. We’re actually going to create two domains, one for the IP Address of your OpenSIPS server and a second one for the FQDN of your OpenSIPS server. For added security, we will create all User accounts under the FQDN Domain. And we’ll reserve the IP Address Domain for DID Trunks/Gateways from registered, commercial SIP providers. This design allows attackers to attempt to register to accounts on your IP Address Domain until the cows come home, and they will never be successful because there are no existing SIP user accounts there. Keep it that way! With our OpenSIPS design, Fail2Ban will block attackers after a single failed registration attempt. And OpenSIPS itself will identify and block all SIP flood attacks using either Fail2Ban or IPset.

    Now that you understand the design, let’s set up your domains. After choosing System -> Domains, enter the IP Address of your OpenSIPS server at the SIP Domain prompt. Then click Add New Domain followed by Reload on Server. Repeat the same steps to enter the fully-qualified domain name (FQDN) of your OpenSIPS server. When finished, you should see:


    Creating Users with OpenSIPS Control Panel

    We’ve already explained the security implications and reason for creating User accounts with your FQDN Domain only. Click on Users -> User Management -> Add New to get started. You can use Numbers (what we call Extensions in Asterisk) or Names. Our preference is to use Numbers for the User accounts and then to create Alias Names (as desired) for each User account. You can’t dial names from most SIP telephones. This also keeps the design similar to what many are used to in the Asterisk environment. A completed dialog would look something like the following. Use the Domain pull-down to choose your FQDN. Obviously, the passwords must be secure and must match. Then the Register button will be enabled to save. The actual Numbers used for Usernames are completely up to you.



    Create at least a couple User accounts so that you can set up two SIP phones to call yourself and verify that everything is working. These User accounts become an integral part of the SIP URI to receive calls from any SIP phone in the world: 7701@opensips.yourdomain.com

    Before you can actually answer an incoming call to your SIP URI, you’ll need to register the User account using either a softphone or SIP phone. We’ll do that next. But, first, let’s create an Alias to 7701 User so that folks can reach you by calling joe@opensips.yourdomain.com

    Click on Users -> Alias Management -> Add New Alias to get started. Fill in the form using the example below. Make sure that you select your FQDN Domain using the pull-downs for BOTH the Domain and Alias Domain fields. Then click Add to save.


    Registering a Softphone to an OpenSIPS User Account

    There are literally dozens of free SIP soft phones from which to choose. We covered some of our favorites for every platform in previous articles. For our purposes today, we recommend you choose one of the Linphone softphones which are available for the PC, Mac, Linux, Android, and iOS platforms. We also recommend signing up for a free Linphone.org SIP account which doesn’t cost you anything. For today, we will be configuring the softphone to register to your new OpenSIPS server.

    Once you have downloaded and installed the Linphone client, go into the Preferences menu and make the following changes. Some depend upon your calling platform.

    • Audio Codecs: PCMU, G722, PCMA
    • Video Codecs: VP8, H264
    • Call Encryption: None
    • DTMF: RFC2833 only
    • Send InBand DTMF: OFF
    • Send SIP INFO DTMF: OFF
    • SIP UDP 5060: Enabled
    • SIP TCP 5060: Enabled
    • Allow IPv6: Disabled

    Then set up a new SIP Proxy account: Username (7701), Password (as defined), Domain: your FQDN not IP address, Transport: UDP, Outbound Proxy: OFF, Stun Server: stun.linphone.org, ICE: ON, AVPF: OFF, Push Notification: ON, Country Code Prefix: 1 (if required by your commercial SIP provider), Register: YES, Account Enabled: YES. HINT: You can call Alias Names via SIP URI, but you can only register to a SIP account using its actual Username.

    Avoiding Lockouts with NeoRouter VPN

    By design, Fail2Ban is unforgiving when it comes to failed registrations. A single failed registration will get an IP address banned for a full week. The reason is because the new bad guy strategy is to hit your server once to determine whether anybody is home. Then the creep bombards you later with an endless stream of registration attempts. With our design, nobody will be home when they return. The bad news is a single failed registration attempt by you or your users will also trigger a ban. There are several workarounds. The easiest is to set up the NeoRouter client on each of your machines including your OpenSIPS server and use the 10.0.0.x private network for access. These IP addresses never get banned. Our previous tutorial will walk you through setting up a free NeoRouter server and installing the free NeoRouter clients on your machines. The client software already is installed and running on your OpenSIPS server. It only requires that you log in using nrclientcmd and register to your NeoRouter server to obtain a private IP address. The other option is to install OpenVPN. Our previous tutorial will walk you through that process. The advantage of OpenVPN is that it’s supported directly on many SIP telephone instruments. The 10.8.0.x addresses are already whitelisted by our OpenSIPS installer.

    There are other options to unban an IP address which has accidentally been snagged. First, almost all of the cloud providers include a Console option in their web portals. Second, you can log into your server via SSH from any non-blacklisted IP address to remove the banned IP address. Once you’re logged in, simply run this command using the IP address you wish to unban: /root/unban-ip xxx.xxx.xxx.xxx

    Choosing Commercial SIP Providers

    Recall that you cannot register to a SIP alias on your OpenSIPS server. We’ll take advantage of this restriction in setting up incoming calls from commercial providers’ DIDs. To set up Trunks from commercial providers so that you can not only receive incoming calls but also make outbound calls over their PSTN network connections, you must use providers that support IP address authentication rather than a SIP registration. Many providers support this including our platinum sponsor, Skyetel, as well as providers such as VoIP.ms, Anveo Direct, V1VoIP, and many others. In our OpenSIPS design, you also can use DIDs from providers that support SIP URI forwarding such as CallCentric and LocalPhone; however, you are limited to receiving inbound calls only. VoIP communications really shines here because you don’t have to choose a single provider to meet all of your communications requirements.

    Skyetel is by far the easiest provider to set up with OpenSIPS. See our earlier tutorial for a special offer that will get you half-price calling for up to $500. Effective 10/1/2023, $25/month minimum spend required. Once you’re registered on the Skyetel site, add a new EndPoint Group using the IP address of your OpenSIP server and designate UDP 5060 as the access port. Sign up for a DID and map it to the OpenSIPS Endpoint Group. Done. In the OpenSIPS Control Panel, navigate to System -> Dynamic Routing and click Add Gateway. Using the template below, create 5 Proxy gateways for the following Skyetel data centers:

    • skyetel-NW 52.41.52.34
    • skyetel-SW 52.8.201.128
    • skyetel-NE 52.60.138.31
    • skyetel-SE 50.17.48.216
    • skyetel-EU 35.156.192.164

    Begin by whitelisting the IP addresses of your SIP providers in /etc/iptables/rules.v4 just below the existing 10.8.0.0/24 rule. The entries should look like this:

    -I INPUT -s 52.41.52.34 -j ACCEPT
    

    Once you’ve entered IP addresses for your providers, issue the command: iptables-restart

    Next, we need to create what Asterisk users know as an Outbound Route. This tells OpenSIPS to send dialed numbers in 11-digit format to Skyetel for termination. We’ve already created the Dial Plan rule for calling out by dialing 1 plus a 10-digit number. So, while you’re still in the Dynamic Routing section of the OpenSIPS Control Panel, click on the Rules tab at the top of the template. Then click Add Rule. Begin by clicking Add ID button and choosing Group ID 0. In the Prefix field, type 1. Now click the Add GW button 3 times after choosing the Skyetel gateways in the following order from the GW pull-down list: skyetel-nw, skyetel-sw, and skyetel-se. Those are the three currently operational Skyetel gateways. When you’re finished, your template should look like the following. Then click the Add button to save the new rule. Click Reload Server to load the new rule into OpenSIPS. Then repeat this procedure leaving the Prefix field blank so that you can make 10-digit calls as well.

    Finally, we need to create what Asterisk users know as an Inbound Route. This tells OpenSIPS where to send incoming calls from our Skyetel DID. OpenSIPS handles inbound routes by defining a User Alias for the Username to which you want to route the incoming DID calls. Click on Users -> Alias Management -> Add New Alias to get started. Fill in the form using the following template and then click Add.

    • Username: 7701 (the extension to which to route the incoming calls)
    • Domain: opensips.xyz.com (the FQDN of your OpenSIPS server)
    • Alias Username: 18435551212 (the 11-digit Skyetel DID)
    • Alias Domain: 11.12.13.14 (the IP address of your OpenSIPS server)
    • Alias Type: dbaliases

    Introducing the VoIP Blacklist

    We’ve always dreamed of an effective VoIP Blacklist, and many have tried. But the crowd-sourced VoIP Blacklist at voipbl.org is the real deal. Everybody can post entries (including the bad guys) and, magically, most of the illegitimate entries get sifted out before the next day’s list is released. The list gets populated every night while you sleep. Here are the steps to install the VoIP Blacklist with IPset:

    apt update && apt install ipset iptables netfilter-persistent ipset-persistent iptables-persistent
    cd /usr/local/sbin
    wget http://incrediblepbx.com/voipbl-update
    chmod +x voipbl-update
    sed -i 's|fail2ban restart|fail2ban restart\n/usr/local/sbin/voipbl-update|' iptables-restart
    iptables-restart
    ipset list voipbl
    ipset list voipbl | wc -l
    

    Then create a cron job in /etc/crontab to run /usr/local/sbin/voipbl-update every day to update the VoIP blacklist.

    1 4 * * * root /usr/local/sbin/voipbl-update > /dev/null 2>&1
    

    Congratulations! You now have a functioning OpenSIPS 3 server that can process incoming calls from SIP URIs as well as DIDs. And you can make SIP URI and 11-digit PSTN calls using your SIP softphone that’s registered to your OpenSIPS server. See you next week. Enjoy!

    Continue Reading: Best of Both Worlds: Safely Marrying Asterisk to OpenSIPS

    Originally published: Monday, October 4, 2021



    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.
     



    1. Nerd Vittles receives referral fees from some VoIP service providers to help cover the costs of our blog. We never recommend particular companies solely to generate commissions. We also test all services that we recommend. []

    Some Further Thoughts & Solutions Regarding DDoS Attacks



    This month’s DDoS attacks on SIP infrastructure in the VoIP community should give us all pause to reflect upon what each of us can do to lessen the impact of these attacks in our Internet-centric community. Suffice it to say, DDoS attacks can be directed toward carriers (last week it was Bandwidth.com), VoIP providers (last week it was VoIP.ms), and VoIP servers (that would be your PBX). While they may not like it, carriers and many VoIP providers have the financial resources to withstand or mitigate a DDoS attack. You, on the other hand, with your budget-basement cloud server probably do not. So what can you do?

    Almost 10 years ago, we introduced the Travelin’ Man 3 firewall for VoIP servers. The idea was novel at the time. You can’t attack what you can’t see. By placing an Incredible PBX server behind the IPtables firewall with no public exposure except for trusted sites and users, your server is essentially hidden from the Internet and all of the world’s bad guys. At the time, the design was poo-poo’d by the SIP purists who were adamant that SIP ports needed to be publicly exposed to function reliably. Wrong. Then there was the FreePBX® firewall which blocked repeated attacks from the IP address of a would-be attacker. But what if a botnet unleashed hundreds of thousands of attacks on your IP address. The FreePBX blocking mechanism obviously would fail. One of the shortcomings of Asterisk®: it isn’t a SIP proxy.

    The moral of the story is pretty simple. Unless you have an unlimited bank account to thwart DDoS attacks and unless your PBX is sitting behind a SIP proxy, you’re much safer with a fully-protected Incredible PBX platform. And, for those believing your IP address is too obscure to attract much attention, try installing a server on CloudAtCost, or Digital Ocean, or Vultr without a firewall to protect your SSH port. You’ll quickly discover how popular you are. Stay safe!

    Originally published: Monday, September 27, 2021



    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.

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    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.

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    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.