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The Most Versatile VoIP Provider: FREE PORTING

VoIP Messaging and The Golden Rule with Incredible PBX



If you want to continue to use SMS and MMS messaging on your VoIP platform, then today’s navigation guide is worth a careful read. Suffice it to say, this is what happens when the feds shirk their responsibilities and leave it to the foxes to guard the chicken coop.

The Golden Rule with all oligopolies is that he who has the gold makes the rules. And, make no mistake, there are stringent new rules for VoIP messaging. Not surprisingly, the FCC has jumped on the cellphone provider bandwagon. You can read all about the new FCC rules here. And the cellphone oligopoly has implemented additional requirements of its own that are enforced through a new organization called The Campaign Registry (TCR).

Any business that sends text messages to U.S. or Canadian mobile phone numbers is now required to register with TCR and obtain a 10-digit long code (10DLC) number. This number is used to identify the sender of each text message and to help the mobile carriers filter out spam (according to the carriers). To register with TCR, businesses must provide information about their company, including their legal name, EIN, and contact information. They must also submit a sample text message and identify the purpose for which they will be using SMS messaging.

What are TCR’s messaging guidelines?

  • Obtaining permission from recipients before sending them text messages
  • Clearly identifying the sender in each text message
  • Providing a way for recipients to opt out of receiving future text messages
  • Avoiding sending spam or unsolicited text messages

Carriers have imposed additional restrictions for certain types of messages so-called SHAFT content: sex, hate, alcohol, firearms, and tobacco (CBD is included). And, unlike email messages, SMS traffic cannot be encrypted so the providers can and do scan the contents of every message that hits their networks. If a business fails to comply with TCR’s requirements, the sender may face penalties including fines and suspension from sending text messages through the cellphone carriers.

You might wonder how these new rules came about. The short answer is that politicians flooded the cell providers’ networks with text messages during the last election cycle. And, of course, the politicians conveniently exempted themselves from all the spam rules including SMS messaging. So the new rules, while appearing admirable to the public, have little if anything to do with the root cause of the problem, the politicians.

CAUTION: What follows is NOT legal advice. It is simply our reading of available literature pertaining to TCR and VoIP.ms rules and regulations. Do NOT rely upon this interpretation of the rules in making decisions regarding SMS deployments. Do your own research. Better yet, consult an attorney.

Keep in mind that the current exception to TCR verification will probably disappear within the next several months. A word to the wise: Go ahead and get registered and verified unless you plan to use your cellphone exclusively for messaging or your usage is clearly non-business. The upfront costs are minimal. Here is an excellent summary of the various 10DLC registration categories.

Assuming your VoIP messages don’t include SHAFT content and otherwise comply with the guidelines above, there remains an exception for messaging without TCR verification… at least for now. The current limits on 10DLC SMS traffic without verification are as follows:

  • Daily limit: 500 message segments
  • Monthly limit: 5,000 message segments
  • Per-recipient limit: 10 messages per day

A message segment is equal to 158 characters. So, a single text message can be composed of one or more segments, depending on its length.

There’s one additional gotcha. For traditional 10-digit numbers, only one SMS segment per second can be sent, and it cannot be increased. So be brief. For toll-free numbers, three SMS segments per second can be sent, and the restriction can be relaxed under certain circumstances. For short code messaging (initial cost is usually $1,500 or more per month to obtain a short code), 100 SMS message segments per second are permitted, and this limit can also be increased.

Now let’s return to our Navigation Guide for those that simply want to use VoIP messaging in the traditional ways that used to work, i.e. for a coach to schedule a little league practice or for you to tell your kid you’re going to be late picking them up from school.

Rule #1: If you have enabled SMS messaging on all of your VoIP phone numbers, do not use numbers on which you depend for critical input for outbound SMS traffic. The risk you run is that breaking one of the rules or limits above may get your number blacklisted from ALL future SMS message traffic.

Rule #2: Don’t break the daily, monthly, and per-recipient messaging limits EVER.

Rule #3: Don’t send SHAFT content over SMS even if you’re joking. Big Brother does not have a sense of humor.

Rule #4: Keep messages under 158 characters in length unless you’re using a toll-free number (158×3 message size limit).

Rule #5: Don’t send more than one message per second. For example, if you’re using a script to send a team notice of a little league practice, be sure to insert a one or two-second pause between each outbound message.

Rule #6: Only use a throw-away number to send outbound SMS messages. If the number gets blacklisted, discard the number.

The Safest VoIP Messaging Platform


As you might expect, the safest way to send and receive SMS messages is through a cellphone or something that looks like a cellphone to the carrier networks. Our review of the Cudy Router spotlights a device that fits the bill perfectly if you have an extra SIM card lying around. Using the web interface on this device, you can send and receive SMS messages using the SMS link on the System Status page because the SMS messages appear to originate from a device on the cell provider’s own network where there are limited restrictions.

Using VoIP.ms for SMS Messaging

Assuming you can comply with all of the restrictions above, here’s our recommendation for a VoIP provider that lets you continue sending messages at minimal cost. That provider is one of our old favorites, VoIP.ms. Using our signup link helps keep the Nerd Vittles lights on so thank you in advance.

So long as you have an SMS-enabled DID with VoIP.ms, SMS messaging costs $0.0075 per message with no additional fees. Below we’ll walk you through getting everything set up with Incredible PBX to take advantage of VoIP.ms SMS services.

Configuring VoIP.ms for SMS Messaging

As noted, you’ll need to order a DID from VoIP.ms that supports SMS. Then enable SMS messaging in the DID setup and specify either an email address or cellphone number for delivery of incoming SMS messages addressed to that DID. If you happen to have a Yealink T46G (not T48G) or a Grandstream GXV phone that is also registered to that extension, the messages will also pop up on your desktop phone with an alert tone if you also enable "Link the SMS received to this DID to a SIP Account" and register the phone to a PJsip extension with the additions which follow. On Grandstream GXV Android phones, we recommend dragging the SMS app to the main screen so that the incoming message count appears beside the SMS icon when new messages are received. If you’re a clever programmer, you also can retrieve incoming messages from the Asterisk log by searching for "Inbound SMS dialplan invoked." The message will be in the following From and Body lines. Or tail /var/log/asterisk/full will look something like this:


To support sending SMS messages, enable the SOAP and REST/JSON API in the VoIP.ms Main Menu, set a very secure API password, and whitelist the IP addresses of each server from which you wish to send SMS messages.

Configuring Incredible PBX to Send SMS Messages

1. Login to your Incredible PBX 2027 server as root and issue the following commands:

cd /root/sms-voip.ms
rm -f /root/sms-voip.ms/*
pip install python-dotenv
wget http://incrediblepbx.com/sendsms-voipms.tar.gz
tar zxvf sendsms-voipms.tar.gz
rm sendsms-voipms.tar.gz
nano -w sendsms

2. When the editor opens, scroll down and replace 8431234567 with your SMS-enabled DID

3. Replace yourname@gmail.com with your VoIP.ms login email address

4. Replace your-API-key with your VoIP.ms API password

5. Save the file: Ctrl-X, Y, then ENTER

6. Send an SMS test message to your cell phone using the following syntax:

/root/sms-voip.ms/sendsms 10-digit-SMS-recipient "Your SMS message"

Configuring Incredible PBX to Receive SMS Messages

To receive SMS messages through FreePBX® using a compatible SIP phone or through the Asterisk CLI, you first must use a PJsip trunk to connect to VoIP.ms. Sample General Settings for the trunk are shown below. In the Advanced tab, set Message Context to sms-in.


You also must create a PJsip extension or use the preconfigured 701 PJsip extension. In the Advanced tab, set Message Context to sms-out.

Finally, edit extensions_custom.conf in /etc/asterisk and add the following code to the bottom of the file:

[sms-out]
exten => _.,1,NoOp(Outbound Message dialplan invoked)
exten => _.,n,NoOp(  TO: ${MESSAGE(to)})
exten => _.,n,NoOp(FROM: ${MESSAGE(from)})
exten => _.,n,NoOp(BODY: ${MESSAGE(body)})
;
; add your VoIPms info in the next 3 lines
exten => _.,n,Set(VOIPMS_ACCOUNT="123456_subacct")
exten => _.,n,Set(VOIPMS_POP="atlanta.voip.ms")
exten => _.,n,Set(VOIPMS_TRUNK="VoIPms-PJsip") ; actual VoIP.ms trunk in FreePBX
;
exten => _.,n,Set(NUMBER_TO=${CUT(CUT(MESSAGE(to),@,1),:,2)})
exten => _.,n,Set(EXTENSION_FROM=${CUT(CUT(MESSAGE(from),@,1),:,2)})
;
; Now map your sending extensions EXTENSION_FROM to corresponding DIDs NUMBER_FROM
exten => _.,n,Set(CASE_701=6005550101) ; ext 701 msgs originate from 6005550101
exten => _.,n,Set(CASE_702=6005550102) ; ext 702 msgs originate from 6005550102
exten => _.,n,Set(CASE_703=6005550101) ; ext 703 msgs originate from 6005550101
;
exten => _.,n,Set(NUMBER_FROM=${CASE_${EXTENSION_FROM}})
exten => _.,n,Set(ACTUAL_FROM="${NUMBER_FROM}" )
exten => _.,n,Set(ACTUAL_TO=pjsip:${VOIPMS_TRUNK}/sip:${NUMBER_TO}@${VOIPMS_POP})
exten => _.,n,MessageSend(${ACTUAL_TO},${ACTUAL_FROM})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,Hangup()
;-------------------------------------------------------------------------

[sms-in]
exten => _.,1,NoOp(Inbound SMS dialplan invoked)
exten => _.,n,NoOp(  TO: ${MESSAGE(to)})
exten => _.,n,NoOp(FROM: ${MESSAGE(from)})
exten => _.,n,NoOp(BODY: ${MESSAGE(body)})
;
; enter your default incoming SMS extension below
; if you want SMS messages delivered to multiple extensions,
; clone additional MessageSend lines below with extension numbers
exten => _.,n,Set(EXTENSION=701)
;
exten => _.,n,Set(ACTUAL_FROM=${MESSAGE(from)})
exten => _.,n,Set(HOST_TO=${CUT(MESSAGE(to),@,2)})
exten => _.,n,Set(NUMBER_TO=${MESSAGE_DATA(X-SMS-To)})
exten => _.,n,MessageSend(pjsip:${EXTENSION}@${HOST_TO},${ACTUAL_FROM})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,Hangup()
;-------------------------------------------------------------------------


In the pasted [sms-out] context, insert your actual VOIPMS_ACCOUNT, VOIPMS_POP, and VOIPMS_TRUNK name in the lines provided. Then map each extension from which you wish to send SMS messages to a VoIP.ms DID on your PBX in the lines provided. In the pasted [sms-in] context, enter the EXTENSION number which should receive incoming messages from the PJsip trunk in which you designated [sms-in] as the Message Context. There is no magic to the [sms-in] context name. If you have more than one PJsip trunk, simply create additional incoming contexts (such as [sms-in-2]) for each additional trunk and clone the [sms-in] code designating the desired extension to receive incoming messages from each DID. For the [sms-out] context, it can be used as the Message Context for multiple extensions that should be enabled to send outbound SMS messages.

Save the file, and reload the Asterisk dialplan: asterisk -rx "dialplan reload"

Introducing the FreePBX SMS Connector Module

Bill Simon recently released another messaging alternative with his SMS Connector Module for FreePBX. The beauty of his new approach is it lets you use Sangoma’s User Control Panel (UCP) to send and receive messages with Incredible PBX 2027. It also supports messaging on both Sangoma’s and ClearlyIP’s SIP phones including the Incredible PBX SIP phones. Here’s the setup process with Incredible PBX 2027 for non-business messaging using VoIP.ms.

At VoIP.ms…
1. Create a Subaccount and DID/Trunk
2. Enable SMS on the trunk and Link SMS Messages received on this Trunk to your SubAccount
3. Enable VoIP.ms API, create an API Password, and Whitelist the public IP address of your server
4. Copy your VoIP.ms email address and API Password for use on your server’s SMS setup

On Your Incredible PBX server…
1. Login to the FreePBX GUI as admin
2. Create a PJsip Trunk for VoIP.ms
3. In Advanced Settings, set Message Context to voipms-sms-in
4. In Admin -> User Management, create a password for extension 701
5. Add the following context to the end of /etc/asterisk/extensions_custom.conf:

[voipms-sms-in]
exten => _.,1,NoOp(Inbound Voip.ms SMS dialplan invoked)
same => n,Set(TO=${MESSAGE_DATA(X-SMS-To)})
same => n,Set(FROM=${CUT(MESSAGE(from),\",2)})
same => n,Set(ENV(QUERY_STRING)=provider=voipms\;to=${TO}\;from=${FROM}\;message=${URIENCODE(${MESSAGE(body)})})
same => n,Set(ENV(REQUEST_METHOD)=GET)
same => n,System(php /var/www/html/smsconn/provider.php)
same => n,Set(ENV(QUERY_STRING)=)
same => n,Hangup()
;-------------------------------------------------------------------------

6. Reload your dialplan: rm /tmp/* ; fwconsole reload

Install and Configure SMS Connector Module…
1. Login to your server as root and issue the following commands:

fwconsole ma downloadinstall https://filedn.com/lBgbGypMOdDm8PWOoOiBR7j/SMSconnector/smsconnector-16.0.11.tar.gz
fwconsole reload

2. In the FreePBX GUI, navigate to Connectivity -> SMS Connector
3. Click Provider Settings and enter your email address for Username and API Secret for VoIP.ms. Click Submit.
4. In SMS Connector menu, click Add Number and enter your DID and PJsip extension 701 to associate with it.
5. Enter VoIP.ms as Provider and click Save Changes.

Using User Control Panel (UCP)…
1. If you have not already done so, apply these UCP patches for Incredible PBX:

mysql -u root -ppassw0rd asterisk -e "update freepbx_settings set value = 'Latest-16' where keyword = 'MIRROR_BRAND_VERSION'; "
mysql -u root -ppassw0rd asterisk -e "update admin set value = 'true' where variable = 'need_reload'; "
rm -f /tmp/*
fwconsole reload
fwconsole ma downloadinstall ucp
rm -f /tmp/*
fwconsole reload

2. Open UCP from FreePBX GUI
3. Login as 701 with your new password
4. Click + in Upper Left of display and add SMS Module for 701.
5. When SMS Module appears on UCP console, click Start Conversation
6. Send a test message to your cellphone
7. Reply to the SMS message from your cellphone
8. Reply should appear in UCP within 20-30 seconds

Let’s close today with a final cautionary note. The Bell Sisters define non-business usage as conversational messaging much like what most already do using their cellphones. If you push the envelope, you risk $100 fines for every message sent. Unless you are a lawyer or have deep pockets to hire one and fight The Oligopoly, you are well advised to obtain a 10DLC number and avoid any potential issues going forward.

Originally published: Monday, November 6, 2023



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.
 



Incredible ChatGPT: Artificial Intelligence For Your Phone



Unless you’ve been sleeping under a rock, you already know that Artificial Intelligence (AI) has the potential to transform every aspect of our lives. The reasons are fairly obvious. AI can process and analyze massive amounts of data in seconds that humans could spend months and years collecting. AI is being used to develop new drugs and treatments, diagnose diseases, and provide personalized care to patients. It’s being used to develop self-driving cars and trucks, optimize traffic flows, and improve public transportation. It can be used in manufacturing to automate tasks, improve quality control, and reduce costs. And AI can be used in the financial world to detect fraud, assess risk, and make investment decisions. Because of AI’s encyclopedic prowess, it can also write a mean term paper with human-like prose. That’s the good news. The bad news is that not everything AI regurgitates is accurate so be extremely careful relying upon AI exclusively to make decisions. See if you can spot the problem in this ChatGPT response:
 


ChatGPT is a large language model chatbot developed by OpenAI, a company backed by Microsoft®. Within two months after launch, ChatGPT had over 100 million subscribers. It is trained on a massive dataset of text and code and is able to generate human-like text, translate languages, write different kinds of creative content, and answer your questions in an informative way. The knowledge cutoff date for the gpt-3.5-turbo version of ChatGPT is September 2021. For users of Incredible PBX, today we’re pleased to bring that ChatGPT model to a telephone near you.

To get started, you’ll need three components. First, you’ll need an Incredible PBX 2027 platform with Debian 11 or Ubuntu 22.04 running on Windows, a Mac, or Linux. Turnkey versions are available for dozens of virtual machine and cloud-based platforms. If you’re using Incredible PBX 2027 on the Rocky 8 platform, you will also need to install the gTTS text-to-speech engine from here. Second, you’ll need to obtain a free OpenAI_KEY here using your Google, Apple, or Microsoft email account. And, third, you’ll need to obtain a free Speech-to-Text API_KEY and API_URL from IBM. Once you have the three pieces in hand, you’re ready to proceed with the installation for your Incredible PBX platform. After installation, you can make ChatGPT queries using any telephone connected to your PBX. Simply dial 2428 (C-H-A-T) and speak your query.

Installing the ChatGPT Telephone Interface

Not every ChatGPT response is suitable for use with a telephone. You wouldn’t want ChatGPT reading you a term paper or spouting out some Asterisk® dialplan code. Nor can most telephones display photos. So our deployment for Incredible PBX today provides two ChatGPT solutions: (1) a command-line interface that is accessible from a terminal or via SSH: chatgpt -p "your query". (2) The telephone interface is accessible by dialing 2428. For the telephone interface, be careful what you ask. You don’t want a 10,000-word response. For example, a good query might be "What are the five best Atlantic coast beaches in the United States." A not-so-good query would be "What are the best restaurants in the world."

To get started after installing Incredible PBX using one of the numerous tutorials available here, log into your server as root and issue the following commands:

cd /
wget https://filedn.com/lBgbGypMOdDm8PWOoOiBR7j/ChatGPT/incredible-chat.tar.gz
tar zxvf incredible-chat.tar.gz
cd /root

Once the components have been downloaded and installed, navigate to the /root folder.
Enter your ChatGPT and IBM STT credentials in the following files:

  • Edit chat and insert your OPENAI_KEY in line 6
  • Edit chatgpt and insert your OPENAI_KEY in line 15
  • Edit chatgpt.sh and insert your OPENAI_KEY in line 12
  •  Also insert your IBM STT API_KEY in line 16 of chatgpt.sh
  •  Also insert your IBM STT API_URL in line 17 of chatgpt.sh

Complete the install by issuing the following commands:

cd /root
sed -i '/\[from-internal-custom\]/r chat.code' /etc/asterisk/extensions_custom.conf
chmod +x chat*
mv chat /usr/local/sbin
mv chatgpt /usr/local/bin
mv chatgpt.sh /var/lib/asterisk/agi-bin
asterisk -rx "dialplan reload"

NOTE: The chatgpt command-line tool does not work on the Rocky 8 platform because of a bug in their fold implementation. However, both our chat command-line tool and the 2428 telephony interface work fine once the gTTS text-to-speech engine is installed for Rocky 8.

Making a Test Call with ChatGPT

Now that all the pieces are in place, let’s make a test call. From a phone connected to your Incredible PBX server, dial 2428. At the prompt, enter the following query: What Are the Five Best Gulf Coast Beach Resorts in the United States? Within a minute or so, ChatGPT will provide the answer using the gTTS text-to-speech engine included in Incredible PBX. Enjoy!

A Cautionary Note About ChatGPT

We’ll close today with this cautionary note about ChatGPT… from ChatGPT:

Originally published: Friday, October 20, 2023



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.
 



Zero-Day Vulnerabilities Compromise All FreePBX Systems



If you’re a user of Asterisk® and FreePBX®, the DEFCON 31 Conference in Las Vegas did not disappoint this year. It exposed not one but three critical, unpatched vulnerabilities in affected FreePBX-based platforms that can compromise your servers in under a minute. I would hasten to add that all of these vulnerabilities were disclosed to Sangoma® months ago and remain unaddressed for months. What this meant was a hacker could easily get administrator privileges on your server with a blank check to make free calls on your nickel or further infect your server with additional hidden components.




 

How Vulnerable Are You? Here’s a quick summary of the bugs documented in the presentation above. If you expose a port on your server to configure SIP phones, you’re compromised. If your users have public IP access to the User Control Panel (UCP), you’re compromised. Any user can delete any asterisk-owned file from your server. Use a Digium® or Sangoma® VoIP phone? You’re compromised. Actually, all you need is the MAC address of one of these phones and its password login and the User-Agent header of any Digium Phone (Digium D60 2_7_0), and you’re compromised if the dphone API RestApp is running on your server. Are you running the API module in FreePBX with public IP address access to your server? You’re compromised because of a bug in the generateDocumentation function. These are classic command injection and authentication bypass issues in FreePBX that can even be triggered from the bad guys’ servers using generated access tokens.


Sangoma, Sangoma. Wherefore Art Thou? You can read all about Sangoma’s Bug Bounty Program here. It was conveniently deleted immediately after this zero-day vulnerability was reported. We’ve reproduced the page from the Wayback Machine. So what happened? According to the good pseudonym researcher, not much. Aside from an initial response indicating that the bugs had been addressed, there was never a follow-up response when the researcher advised that the patches did not work.




 

What Can You Do? Your safest bet is to switch to a security model that does not expose your server or its assets to the public Internet. Incredible PBX is an out-of-the-box platform that provides this security. It’s available for Rocky 8 (not recommended), Debian 11, Ubuntu 22.04 as well as virtualization platforms including VirtualBox, VMware, Proxmox, Windows WSLg, LXC Linux Containers, and Apple’s UTM platform. OpenVPN is also strongly recommended.

At the very minimum, put your server behind a hardware-based firewall with no public Internet exposure until these bugs are properly resolved. You’ve been warned!

Follow updated comments on this issue on the FreePBX Forum and the VoIP-info.org Forum.

Originally published: Sunday, September 17, 2023    Updated: October 13, 2023



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 2027 for LXC Linux Containers



We introduced Zorin OS in our recent review of the latest Acer Aspire 5 notebook PC. If you’ve never heard of Zorin, it’s probably the best desktop operating system available. While it is Linux-based, it can look like any desktop you’re already familiar with including Windows 11 and even a Mac if you spring for the $39 Pro version. It can be installed on as many PCs as you personally own, and Zorin Pro provides an incredible assortment of apps including the full LibreOffice suite, VirtualBox, music apps including Clementine, video streaming with VLC Media Player, Gnome’s Gmail, Firefox, and a number of free photo editing tools. In short, any app that will run under Ubuntu will run under Zorin. If you’re a diehard OpenVPN fan like us, it works swimmingly under Zorin and is a breeze to install.


After walking you through a Zorin install, we want to introduce you to a new Linux virtualization solution called LXC Linux Containers. It offers many of the same features you’d find in the OpenVZ cloud platform including shared kernel access and almost instantaneous startups. On the Zorin platform, it’s a one-minute setup procedure. We’ve even built Incredible PBX 2027 LXC images for both Debian 11 and Ubuntu 22.04 servers so you can bring up one or more full-featured unified communications platforms in a matter of minutes.

Preparing Zorin 16 Installation Media

Prepare a USB stick after downloading the Zorin 16 ISO of your choice. The easiest method is to download and install balenaEtcher on your desktop machine. It’s available for Windows, Mac, and Linux. Insert your USB stick into your desktop computer and then run the balenaEtcher app. Choose your Zorin ISO after clicking Flash from File. Choose your USB stick from Select Target. Be sure you don’t accidentally choose your desktop’s main drive! Click the Flash button to begin. Once the ISO is transferred, gracefully eject your USB stick from your desktop machine.

Configuring Windows 11 for Dual-Boot

We won’t bore you with a tutorial on setting up Windows 11. It works much the same way on almost any modern computer. Once you get Windows 11 installed, the only change we need to make is to shrink the main Windows partition so that there is some room to install and use Zorin. For Lenovo and HP machines, don’t waste your time trying to get Zorin installed. It won’t. For other machines, from the Windows desktop, tap the Windows key and Search for “Create and format hard disk partitions”. Highlight your main Windows drive, usually called “OS (C:)”. Control-tap on the Touchpad or Right-Click your mouse in that partition and select “Shrink Volume”. Choose the amount of space to shrink the Windows partition and allocate to Zorin. We chose 250GB, but that’s your call. Reboot your machine for the changes to take effect. When the boot logo appears, press F2 repeatedly to enter the BIOS setup. Click on the Security tab and set a Supervisor Password. No other changes are necessary. If you can disable UEFI do so, but it’s not required. In the boot options, specify your USB drive as the primary boot source so we can install Zorin from a USB stick. Move to the Exit tab and choose Exit and Save Changes to initiate a reboot.

Installing Zorin 16 on a Desktop PC

Now we’re ready to install Zorin on your desktop machine. Insert the USB stick and turn on or reboot the computer. If the machine doesn’t boot Zorin from the USB stick, you may need to make the USB port your primary boot device in the BIOS settings. Choose the LIVE Session option to make sure Zorin will start on your machine. Also make sure you can configure WiFi in the Settings -> Network tab. If all goes well, click the Install Zorin link on the desktop to install Zorin onto your empty partition when prompted. Upon reboot, if UEFI is enabled, be sure to register Zorin in the UEFI whitelist when prompted. In some BIOS configurations, you may need to set NVME:ubuntu as Primary in the Boot setup’s Drive Priorities after installing Zorin. This was the case on our favorite MiniPC desktop, the MINISFORUM Venus Series NAB6 Mini PC (shown below). Now reboot and choose Zorin from the dual-boot menu.


One final tip. If you’ve activated OpenVPN on your Zorin 16 machine, all of the LXC Containers can connect to other OpenVPN servers without the need to install an OpenVPN client on each of the LXC Containers.

Setting Up LXC Containers on a Linux Desktop

1. After loading Zorin, drop down to the Terminal window from the desktop. If you haven’t previously configured the root user account, issue the following commands:

#Set up a very secure root password
sudo passwd root
# Login as root with your root password
su root

2. Install the LXC Linux Container components:

su root
apt install lxc lxc-templates lxctl
lxc-checkconfig

3. Download, install, and configure an IncrediblePBX2027-U LXC container (3GB):

su root
cd /
wget https://filedn.com/lBgbGypMOdDm8PWOoOiBR7j/IncrediblePBX2027-LinuxContainers/iPBX2027U.tar.gz
tar zxvf iPBX2027U.tar.gz
lxc-start -n IncrediblePBX2027-U
lxc-attach -n IncrediblePBX2027-U
/etc/profile.d/helloworld.sh
reboot
su root
cd ~
pbxstatus
./update-IncrediblePBX
passwd root
./admin-pw-change
./apache-pw-change
./timezone-setup

4. Download, install, and configure an IncrediblePBX2027-D LXC container (3GB):

su root
cd /
wget https://filedn.com/lBgbGypMOdDm8PWOoOiBR7j/IncrediblePBX2027-LinuxContainers/iPBX2027D.tar.gz
tar zxvf iPBX2027D.tar.gz
lxc-start -n IncrediblePBX2027-D
lxc-attach -n IncrediblePBX2027-D
/etc/profile.d/helloworld.sh
reboot
su root
cd ~
pbxstatus
./update-IncrediblePBX
passwd root
./admin-pw-change
./apache-pw-change
./timezone-setup

5. Every time you start or restart the container, be sure to issue these commands:

iptables-restart
fwconsole restart
/root/update-IncrediblePBX
pbxstatus

Mastering LXC Container Commands

Here’s a quick list of the main LXC Container commands together with examples:

Switch to root user to begin or add sudo prefix to every command!

1. List LXC Containers: lxc-ls
2. Start LXC Container: lxc-start -n IncrediblePBX2027-D
3. Information about LXC Container: lxc-info -n IncrediblePBX2027-D
4. Activate Console for LXC Container: lxc-attach -n IncrediblePBX2027-D
5. Duplicate/clone an LXC Container: lxc-copy OrigContainer -N NewContainer
6. Stop an LXC Container: halt | lxc-stop -n IncrediblePBX2027-U
7. Create Debian 11 image: lxc-create -t download -n DEBIAN11 — -d debian -r bullseye -a amd64
8. Delete/Destroy an LXC Container: lxc-destroy -n IncrediblePBX2027-D

Originally published: Monday, September 11, 2023



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.
 



VirtualBox Wonder: It’s Incredible PBX 2027-D 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 can purchase a $1 a month phone number in your choice of area codes from CallCentric. Setup instructions here. If you decide VoIP is not for you, you don’t have to buy anything else 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. Here’s a great alternative for you.

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 2027-D image for Debian 11, 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® 20 and FreePBX® 16. There are no hidden fees or crippleware to hinder your use of Incredible PBX for as long as you like. 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 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 6.1 installers. 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 6.1 platform is required. Adjust screen size in View -> Virtual Screen.

Installing the Incredible PBX for Debian 11 Image

To begin, download the Incredible PBX 2027-D image with Debian 11 (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 2027-D 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 2027-D 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 2027-D in VirtualBox

Once you’ve imported and configured the Incredible PBX Virtual Machine, you’re ready to go. Highlight the Incredible PBX 2-27-D 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 may 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 2027-D 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.

Setting Your Incredible PBX Passwords

When the base install finishes and the VM restarts, log back in as root. Change your root password by issuing the command: passwd. 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.

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 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 User/Password Authentication. Create a username and password for your subaccount. For Transport, choose UDP. For Device Type, choose Asterisk. 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 Desktop 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 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 a 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.


The easiest way to get started is to download a free softphone onto your desktop. We recommend Zoiper 5 for personal use. You can download and install it from here. Once installed, you’ll first need to decipher your extension password for an extension you wish to use to connect to Zoiper. Log into your Linux CLI and switch to the root user as documented above. Then run: /root/show-passwords. Specify 701 and 705 as the starting and ending extensions. Make note of the 701 extension password. Run pbxstatus and make note of your LAN IP address, e.g. 179.xx.yy.zz. Next, start up Zoiper from your desktop and choose the free version. For the desired account enter: 701@179.xx.yy.zz:5061 where 179.xx.yy.zz is the local IP address of your virtual machine. For the password, enter the 701 extension password you deciphered above. Press ENTER twice to complete the connection. When the dialer appears, try out some of the free Incredible PBX applications below.

NOTE: You must use the Keypad option shown in the right window of Zoiper after your call is connected for any app that prompts for keyboard input.

Here are some numbers to try:

123 - Reminders
222 - Timeclock for Employees (try 12345)
223 - AsteriDex Lookup & Dialer (try 335 for Delta Airlines)
947 - Weather by ZIP Code (requires keyboard entry of 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.

Configuring a Softphone Extension on a Smartphone

Adding an Incredible PBX extension to your smartphone gets a little trickier. Whether you’re an iPhone or Android lover, all smartphones use batteries, and you don’t want to drain your battery by running a softphone as a foreground app all the time. Fortunately, you now have some choices in softphones engineered to work without draining your battery. While they all cost money, it’s not much money. We’ve written about all the choices, and you’ll find the links in our Softphone Provider Recommendations on the new Incredible PBX Wiki.

With PJsip extensions, you’re not limited to a single phone connection at a time, and we’ve preconfigured extension 701 to support ten simultaneous connections. The setup on the softphone side is simple. For the server, enter the actual IP address of your PBX in the following format: 22.33.44.55:5061. Then enter 701 for the username and enter the password assigned to the 701 extension on your PBX. When an incoming call arrives, all the phones registered to extension 701 will ring simultaneously. Simply answer the call on the phone that is most convenient. For extension 702, you can change the number of simultaneous connections by clicking the Advanced tab and setting the number in Max Contacts.

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 16 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 4, 2023



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. []

Adding Incredible PBX Goodies & More to VitalPBX 4



As continued use of FreePBX® becomes more and more precarious because of deprecated components and looming incompatibility with Asterisk® 21, the appeal of 3CX and VitalPBX as a VoIP platform becomes increasingly compelling. Whether you’re a home user, a small business, or a call center, VitalPBX provides a solution to meet your requirements. To make the transition a bit less painful, today we introduce a number of popular Incredible PBX applications for VitalPBX 4. And, as always, all of the Incredible PBX additions are free, open source, and GPL code.


If you’re unfamiliar with the VitalPBX VoIP platform, here are some features that may be of interest. First, it runs on the latest Debian 11 platform and is Asterisk-based freeware with optional commercial components. Most GPL applications designed for FreePBX will run equally well under VitalPBX without modification. Second, VitalPBX provides multi-tenant functionality with the purchase of a commercial module. Third, VitalPBX supports Asterisk High Availability (HA) failover at no cost using an open source script provided by the VitalPBX developers. Complete tutorial here. Compare this to the FreePBX HA offering which retails for $1,500. Commercial modules offer Microsoft Teams integration as well as the full complement of Sonata Suite Call Center offerings: Billing, Switchboard, Stats, Dialer, and Recordings. Faxing, Paging, Queues Callback, and Phone Provisioning modules are also available at modest cost. Keep reading if any of these are of interest to you.


Getting Started with VitalPBX

Before you can install VitalPBX applications, you’ll obviously need a VitalPBX server. You can build the platform with on-premise hardware, or in the cloud using one of our recommended providers, or on a Raspberry Pi. We recommend at least 4GB of RAM and at least a 30GB disk. Two gigs of RAM will suffice with a 2GB swap file. VitalPBX can be deployed using the VitalPBX ISO, or you can start with a fresh Debian 11 platform and then run the VitalPBX install script:

wget https://repo.vitalpbx.com/vitalpbx/v4/apt/debian_vpbx_installer.sh
chmod +x debian_vpbx_installer.sh
apt install sudo
./debian_vpbx_installer.sh

For Raspberry Pi deployments, here are the steps using a 32GB microSD card:

Begin by downloading Raspberry Pi Imager for PC, MAC, or Ubuntu desktop. Run the Imager from your desktop computer with the following settings after inserting your 32GB microSD card into your desktop machine (see the sidebar for an inexpensive microSD/USB device):

OS: Raspberry Pi OS (other) -> Raspberry Pi OS Lite (64-bit)
Storage: Select your microSD card (32GB Type 10 recommended)
Click WRITE

Remove the microSD card from your desktop computer. Insert it into your Raspberry Pi and power on the device. The initial Raspberry Pi OS setup for the United States follows. For users elsewhere, follow your nose.

Choose keyboard layout: (Other, English (US) for USA users)
Keyboard Layout: English (US)
username: nerd
password: make it secure, type it twice
login: nerd with new password
sudo passwd root
create new secure root password
logout: exit
login: root with new root password
userdel nerd
nano -w /etc/ssh/sshd_config
  edit and uncomment: PermitRootLogin yes
  uncomment PasswordAuthentication yes
  save: Ctrl-X, Y, then ENTER key
run: raspi-config
  Settings Apply to: pi
  Localization: WLAN Country: US
  System Options: Wireless LAN: Enter your SSID and SSID passphrase
  System Options: Hostname: debian
  System Options: Power LED: YES
  Interface Options: SSH: YES
  Localization: Locale: Disable en_GB.UTF-8 and Enable en_US.UTF-8
  Localization: TimeZone: America, NewYork
  FINISH and Reboot

Once your Raspberry Pi has restarted, login as root with your root password and run the debian_vpbx_installer.sh script from above.

Adding a Whitelist & Hardening Your Firewall

We’ve built firewall whitelist rules for some of our favorite providers: Skyetel, BulkVS, VoIP.ms, Acrobits, SignalWire, Nexmo, Callcentric, and Anveo Direct. Also included are all private LAN, non-routable IP addresses and the default OpenVPN addresses. Issuing the following commands will install this whitelist and overwrite your existing firewall whitelist, if any. WARNING: The existing VitalPBX Firewall exposes all of your SIP ports as well as SSH, HTTP, and HTTPS so deploy VitalPBX behind a hardware-based firewall unless you significantly harden the VitalPBX Firewall ports. If you’re sure you’ve whitelisted the IP addresses of all your remote PCs, extensions, and trunk providers in Admin -> Firewall -> Access Control, then you can harden your firewall and protect your server by deleting the following entries in Admin -> Firewall -> Rules: HTTP, HTTPS, SSH, PJSIP, SIP, and IAX2. Then test all your connections to make certain they still are accessible. For future additions, we strongly recommend using OpenVPN addresses which require no new Firewall additions.

cd /root
wget https://filedn.com/lBgbGypMOdDm8PWOoOiBR7j/VitalPBX-4/whitelist.sql
mysql -u root ombutel < whitelist.sql
vitalpbx apply-firewall
iptables -nL

gTTS Text-to-Speech Engine for VitalPBX

We've tested and implemented at least a half dozen text-to-speech engines to support Asterisk applications including Festival, FLITE, Amazon's Polly, IBM's Bluemix TTS, Pico TTS, and more. None are better than Google's free gTTS engine. Here's how to deploy it with VitalPBX to support all of your applications requiring TTS support. Login to your server as root and issue the following commands:

apt-get update
apt-get -y install jq libsox-fmt-all
apt-get -y install python3-pip
pip install --upgrade pip
pip3 install --upgrade pip
ln -s /usr/bin/pip3 /usr/bin/pip
pip install gTTS

Adding Custom Contexts Support to VitalPBX

In addition to the commercial modules, there are a number of free VitalPBX add-ons, one of which is Custom Contexts. We would recommend adding all of the free ones to get started. After logging into the web interface as admin, navigate to Admin -> Add-ons -> Add-ons. Click the Check Online button to load the latest available add-ons. Then click the Install icon for the following add-ons: System API, Authentication Codes, Bulk Extensions, Custom Contexts, Phone Books, and Task Manager. Once these add-ons are installed, you can install the following components.


Adding Incredible PBX Starter Kit to VitalPBX

We've put together a collection of some of our favorite Incredible PBX applications to enhance the VitalPBX platform. These include telephone apps like Yahoo News Headlines (dial 951), NWS Weather Reports by ZIP Code (947), Today in History (86329), and Telephone Reminders (123). In addition, we've reworked the pbxstatus utility (above) which will display whenever you log into your server as root from the Linux command line.

Many of these applications rely upon the gTTS text-to-speech engine so be sure you install it before proceeding.

To install the Incredible PBX collection, log into your server as root and issue the following commands:

cd /etc/asterisk/vitalpbx
cp extensions__80-IncrediblePBX.conf /root/extensions__80-IncrediblePBX.conf.bak
cd /
wget https://filedn.com/lBgbGypMOdDm8PWOoOiBR7j/VitalPBX-4/incrediblepbx.tar.gz
tar zxvf incrediblepbx.tar.gz
rm -f incrediblepbx.tar.gz
asterisk -rx "dialplan reload"
echo "0 0 * * * root /var/lib/asterisk/agi-bin/run_recurring >/dev/null 2>&1" >> /etc/crontab
echo "3 0 * * * root /var/lib/asterisk/agi-bin/run_reminders >/dev/null 2>&1" >> /etc/crontab

Using Telephone Reminders with VitalPBX

Nerd Vittles Telephone Reminder System has been reworked for VitalPBX 4 and PHP 8.1. It lets you schedule reminders for future events (at least 4 minutes in the future) by telephone by dialing 123. When the appointed date and time arrives, Asterisk swings into action and places a call to the number you designate to deliver a customized reminder message. Recurring reminders also are supported. You can set up reminders that place calls daily or on weekdays as well as weekly, monthly, and annually. This means it can be used to wake you up in the morning, or to remind Granny to take her medicine every day, or to remind your Little League team of practice times and locations, or to remind you and your customers of scheduled and recurring events. External reminder calls are supported using your default outbound route's dial string, e.g. NXX-NXX-XXXX.

The complete tutorial for Telephone Reminders 4 is available here. The web interface is not yet supported on the VitalPBX platform; however, this Telephone Reminders app adds features that are not available in the *38 offering included in the VitalPBX Feature Code listing. Among these are optional recurring reminders as well as the ability to revise your reminder message before actually scheduling it.

Headline News & Weather Forecasts & Today in History

These three applications are self-explanatory. The best way to learn about them is to dial the three extensions from any phone registered on your VitalPBX server: Headline News (dial 951), Weather Forecasts by ZIP Code (dial 947), and Today in History (dial T-O-D-A-Y)

Adding OpenVPN to VitalPBX

The most secure method for accessing VitalPBX is to place your server behind a hardware-based firewall and use OpenVPN from the client PCs and phones to access the server. VitalPBX includes an OpenVPN add-on that includes both a server and a free 2-client license. For unlimited clients, you can purchase the commercial module for $120. In the alternative, you can deploy your own OpenVPN server and clients using this Nerd Vittles tutorial for Debian.

If you already have an OpenVPN server in operation, create an OpenVPN client for VitalPBX and name it incrediblepbx.ovpn. Copy it into the /etc directory of your VitalPBX server. Then issue the following commands and reboot to activate OpenVPN on your VitalPBX server:

apt-get update
apt-get -y install openvpn unzip
cd /
wget https://filedn.com/lBgbGypMOdDm8PWOoOiBR7j/VitalPBX-4/openvpn-vitalpbx.tar.gz
tar zxvf openvpn-vitalpbx.tar.gz
rm -f openvpn-vitalpbx.tar.gz
shutdown -r now

Getting Started with Faxing

If your deployment is for a home or home office, then VitalPBX offers a free faxing component for a single trunk. We've tested this with VoIP.ms, and it works flawlessly. Begin by enabling the Virtual Faxes module. For your Trunk, enable FAX Detection and T.38, if desired. For your Fax Device, provide a Description, Destination Email, and CallerID Name and Number. For your Inbound Route, enable Fax Detection and Fax Destination of Fax Devices selecting the Destination Description you assigned to your Fax Device. Now place a test call to your DID from FaxZero.com. The Fax Sending module worked equally well.

Adding CallerID Names for Incoming Calls

Legal Disclaimer: Most CNAM providers have restrictions regarding caching of CNAM data. The courts consistently have ruled that phonebook data is not copyrightable. And every PBX caches CNAM data. After all, that's what CDR logs are all about. Consult with your own attorney if you have concerns, or simply stop reading here. 🙂

Some providers of DIDs also offer CallerID Name (CNAM) service for incoming calls. With VoIP.ms, it's optional and costs $0.008 per call. With BulkVS, it's mandatory and costs $0.003 per call. With many DID providers, you will only receive the CallerID Number on incoming calls. Thus was born our CallerID Trifecta and later Superfecta add-ons many years ago. Most of the free sources from yesteryear have disappeared, and we've only found two commercial sources that are reasonably priced at $0.003 per call: BulkCNAM (from the BulkVS folks) and EZCNAM at same price with a 25¢ credit to let you try out their service. Both work well.

Once you have installed Custom Context module for VitalPBX as well as the Incredible PBX Starter Kit from above, here are the steps to implement CNAM lookups on your incoming calls. First, sign up for an account with one or both of the providers and obtain a SOAP API Key from BulkCNAM or a traditional API key from EZcnam. Then login to your server as root and create an executable install script using the following template for BulkCNAM:

cd /root
rm -f superfecta-bulkcnam
wget https://filedn.com/lBgbGypMOdDm8PWOoOiBR7j/VitalPBX-4/superfecta-bulkcnam
sed -i 's|SOAP-API-KEY|actual-key|' superfecta-bulkcnam
sed -i '\:// BEGIN CallerID Superfecta:,\:// END CallerID Superfecta:d' /etc/asterisk/vitalpbx/extensions__80-IncrediblePBX.conf
cat superfecta-bulkcnam >> /etc/asterisk/vitalpbx/extensions__80-IncrediblePBX.conf
asterisk -rx "dialplan reload"


Or create an executable install script using the following template for EZCNAM:

cd /root
rm -f superfecta-ezcnam
wget https://filedn.com/lBgbGypMOdDm8PWOoOiBR7j/VitalPBX-4/superfecta-ezcnam
sed -i 's|=API-KEY|=actual-key|' superfecta-ezcnam
sed -i '\:// BEGIN CallerID Superfecta:,\:// END CallerID Superfecta:d' /etc/asterisk/vitalpbx/extensions__80-IncrediblePBX.conf
cat superfecta-ezcnam >> /etc/asterisk/vitalpbx/extensions__80-IncrediblePBX.conf
asterisk -rx "dialplan reload"


In your install script of choice, replace actual-key with the SOAP API key or API key you obtained from the provider. Make the script executable (chmod +x) and then run it to install the new script in your dialplan. Then reload dialplan: asterisk -rx "dialplan reload"

As deployed, the [superfecta] context assumes you want incoming calls routed to extension 501. You can modify this in /etc/asterisk/vitalpbx/extensions__80-IncrediblePBX.conf and reload your dialplan.

In the VitalPBX GUI, login as admin and navigate to PBX -> Applications -> Custom Contexts and create a new Custom Context and reload the dialplan:

Description: CallerID Superfecta
Context: superfecta
Extension: s
Priority: 1

Destination:

Custom Contexts -> Incredible PBX

In PBX -> Calls Routing -> Inbound Routes, edit your existing Inbound Route for your incoming DID and set the Inbound Destination to: Custom Contexts -> CallerID Superfecta. Then reload your dialplan.

How It Works: When an incoming call from a new caller is detected, the Superfecta script will greet the caller and ask the caller to press 7. Once the caller presses 7, the Superfecta script will look up the CNAM entry matching the CallerID Number and then route the call to extension 501. Successful callers are whitelisted and logged in the Asterisk database: database show cidname. When the same caller calls again, the call will be routed to extension 501 without prompting to press 7. Additional routing options are available by editing the [superfecta] context.

Configuring Gmail as SMTP Relay Host

The VitalPBX Portal includes the option to configure either a self-hosted email server (which may or may not work depending upon your upstream provider) as well as an SMTP relay host such as Gmail. You'll find it under Admin -> System Settings. In the alternative, you may prefer to do it yourself. Here's how.

1. Log into your server as root and issue the following command:

dpkg-reconfigure postfix

Click OK on the first dialog. Choose Internet Site as your Type of Mail Configuration. Accept the defaults for the System Mail Name, Root and Postmaster Recipient, and Other Destinations. Choose Yes for Forced Synchronous updates. Accept the defaults for the Local Networks, Default Mailbox Size, and Local Address Extension Character. Choose IPv4 for the Internet Protocol.

2. Once Postfix is reconfigured, edit /etc/postfix/main.cf. In the second section of code beginning with relayhost =, replace the relayhost= line with the following block of commands:

relayhost = [smtp.gmail.com]:587
smtp_use_tls = yes
smtp_sasl_auth_enable = yes
smtp_sasl_security_options = noanonymous
smtp_sasl_password_maps = hash:/etc/postfix/sasl_passwd
smtp_tls_CAfile = /etc/ssl/certs/ca-certificates.crt
smtp_fallback_relay =

3. Create the following new file using your Gmail account name and password.

nano -w /etc/postfix/sasl_passwd:

[smtp.gmail.com]:587 yourname@gmail.com:yourpassword

5. Change the permissions on the sasl_passwd file:

chmod 600 /etc/postfix/sasl_passwd

6. Use postmap to compile and hash the sasl_passwd file:

postmap /etc/postfix/sasl_passwd

7. Restart Postfix: systemctl restart postfix

8. apt -y install mailutils

9. Send yourself a test email: echo "test" | mail -s "Test Mail" somebody@gmail.com

Free Voicemail Transcription of Messages

For many years, Incredible PBX has included documentation to deploy IBM's Speech-to-Text (STT) engine to transcribe voicemail messages and deliver them by email for missed calls. Today we are pleased to bring that same functionality to VitalPBX 4. To get started, make certain that you have outbound email functioning on your server using the steps in the previous section. Then open an account with IBM and sign up for their LITE Speech-to-Text service. This provides you with 500 minutes a month of free STT transcription; however, you must use it at least once every 30 days or risk having your STT account terminated. So you may wish to setup up a recurring weekly reminder at a time when your extension will not otherwise be answered. Set up a short message to assure that voicemail transcription will be triggered. This will keep your LITE plan active without using many of your allocated minutes.

Once you have signed up for the STT-LITE service, navigate to Resources:AI/Machine Learning:STT in the LITE Tier and obtain or create an API Key and URL. Copy both the API Key and URL to your desktop. You'll need them as part of the VitalPBX component install below.


Next, login to your VitalPBX server as root and issue the following commands:

cd /root
apt -y install dos2unix lame
wget https://filedn.com/lBgbGypMOdDm8PWOoOiBR7j/VitalPBX-4/sendmailibm.tar.gz
tar zxvf sendmailibm.tar.gz
rm -f sendmailibm.tar.gz
nano -w sendmailibm
# insert your API Key and URL and Save file: Ctrl-X, Y, ENTER
cp -p sendmailibm /usr/local/sbin/.
cp -p voicemail__60-1-transcript.conf /etc/asterisk/vitalpbx/.
asterisk -rx "dialplan reload"

When the nano editor opens in step 6 above, insert your API Key and URL in the spaces provided. Then save the file: Ctrl-X, Y, then ENTER. Continue with the remaining steps above to complete the install.

By default, this setup assumes that incoming calls are delivered to an extension on your PBX. Assuming that is extension 501, open the VitalPBX GUI and edit your Extension's settings by adding your email address in General Settings and in the Voicemail tab specify Enable Voicemail and Attach Voicemail YES. If you wish to delete the messages from your server after sending the email, specify Delete YES. Then save your settings and reload your dialplan.


Finally, make a test call to that extension and don't answer. Leave a brief message and hang up. The transcribed voicemail together with an MP3 recording of the message should arrive within a minute or two.

You Snooze, You Lose

Sorry to say our supply of free licenses to one of our favorite add-ons, the $100 Starter Kit, has been exhausted. If we get additional ones to hand out, we'll post an update here. Here's what's included in the VitalPBX Starter Kit:


Originally published: Monday, August 7, 2023    Updated: September 13, 2023



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.
 



Summer Break: Catching Up on Nerd Vittles Happenings



Nerd Vittles has been churning out technology articles for well over a decade. And, while you’re enjoying your time off this summer, we thought you might enjoy catching up on some of our most-read articles. According to Google Analytics, these were the most recent top picks from the 1.2+ million visitors who have dropped in to Nerd Vittles these past 10 years.
 

Our Favorite All-You-Can-Eat Deals in Cyberspace is self-explanatory covering a free cloud offering, domains at cost, VPN offerings for life, unlimited music streaming services, unlimited home internet service, lifetime cloud storage services, lifetime email hosting together with a few services to avoid.

Incredible PBX 2027-U for Proxmox 7 will walk you through setting up a Proxmox server and building Incredible PBX 2027 virtual machines with Ubuntu® 22.04, Asterisk® 20, and FreePBX® 16 in minutes using a powerful little MiniPC.

The 5-Minute PBX in the Cloud Platform for $2 a Month introduces the same Incredible PBX 2027-U platform from one of our favorite cloud services, CrownCloud. It’s $25 a year and includes a free backup image with your choice of locations: Los Angeles, Atlanta, Miami, Germany, or The Netherlands.

gTTS: The Ultimate (free) Text-to-Speech Engine for Asterisk introduces our favorite text-to-speech engine for Asterisk which also happens to be free. It’s now included with all Incredible PBX 2027 platforms.

For those that prefer to build your own server platforms, we’ve introduced Incredible PBX install scripts and tutorials for the following operating systems:

$1 a Month Buys a Cloud Powerhouse for Incredible PBX, an alternative for those that find $2 a month cost prohibitive for cloud hosting. You get an equally powerful platform from RackNerd but no backup option is included. Installation of Incredible PBX is a two-step process: installing an operating system image and then running the matching Incredible PBX 2027 installer.

$0 a Month Buys an Oracle Cloud Powerhouse, an alternative that’s free if you’re lucky enough to snag an instance.

Finally, a few of our favorite SIP providers with our reasons why:

  • Skyetel – Half-price calls with your first $250 purchase. $25/mo. minimum spend.
  • Clearly Anywhere – The ultimate mobile user VoIP companion
  • Linphone – Free SIP Calling to Anybody, Anywhere
  • CallCentric – $1 a month residential DID with free incoming SIP calls
  • FreeVoipDeal – €10.00 for 300 min./week of free calls to 33 countries for 120 days

Enjoy your summer break!

Originally published: Monday, July 24, 2023



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.
 



Building a Dirt-Cheap Communications Platform with VoIP



There are literally thousands of options when you finally ditch your landline and stagger into the VoIP world. We’re often asked, "What would you recommend if price was the major criteria?" Our response goes something like this. You get what you pay for and our recommended providers continue to be ClearlyIP and Skyetel in no particular order. Having said that, if price is your primary consideration, here’s our Plan B which we use regularly.

First, a word of explanation. This is not the Ma Bell days any longer so you’re not limited by cost to a single provider. It costs little more to have several VoIP providers than to have one since most VoIP services are pay-as-you-go. So, in our least costly category, we actually recommend two providers, BulkVS for VoIP calling and VoIP.ms for VoIP messaging. Faxing also works incredibly well with both of these providers. Just follow our fax tutorial to get started. In terms of deployment, it means you will have one primary phone number for making and receiving calls and a second number for sending and receiving SMS messages. Incoming SMS messages can optionally be delivered to either your primary email account and/or a third phone number such as your cellphone.

Obtaining a phone number to make and receive phone calls through BulkVS will set you back 6¢ a month with a 25¢ initial setup fee. Incoming calls are $0.0023 per minute. Optional 911 support is 49¢/month. Outbound calls to North America are $0.004 per minute. SMS messaging at BulkVS is cost-prohibitive. A phone number (DID) at VoIP.ms to send and receive both calls and messages runs $0.85 per month. Incoming calls run $0.009 per minute. Outbound U.S. calls are a penny a minute while calls to Canada are $0.0052 per minute. SMS messages are $0.0075 per message while MMS messages are 2¢.

We’ll be using Incredible PBX 2027 and PJsip with Asterisk® 20 and FreePBX® 16 to set the trunks up today. We’ll configure the default route for outbound calling to be BulkVS with VoIP.ms as an outage failover. All incoming calls from both DIDs can be directed to a phone, ring group, or IVR of your choice. For SMS messaging, we’ll use the FreePBX GUI to set things up. Scripts also are provided in /root/sms-voip.ms to send messages. We’ll configure the VoIP.ms messaging defaults to also relay incoming messages to both an email address and a cellphone. For additional alternatives, check out our VoIP.ms tutorial.

Getting Started with BulkVS

To get started, click the sign up link on the main BulkVS page. Then fund your account with $25 using PayPal. Or you can sign up for Net 15 billing and pay by check or credit card if you’re not in a rush to get started.

BulkVS offers two ways to set up your BulkVS trunking: IP-based authentication and SIP registration. If you don’t have a firewall which means you’re not using Incredible PBX, the first method is a little safer because nobody can spoof the IP address of your Asterisk® PBX. But it’s not for everyone. For example, if you’re behind a NAT-based firewall or if your server has a dynamic IP address, then IP-based authentication really isn’t an option. Similarly, if you don’t have control of the router that your PBX is sitting behind, then IP-based authentication won’t work since you have to forward both the SIP port (UDP 5060) and the RTP ports (10000-20000) to your PBX. The beauty of SIP registrations is they work from almost anywhere including double-NAT environments. We’ll cover the SIP registration approach below which will work for everyone. See our BulkVS tutorial for additional options.

BulkVS Setup with PJsip Registration

Step 1: Go to Inbound -> DIDs – Purchase and buy one or more DIDs for your PBX.

Step 2: Go to Interconnection -> Host – Add and add your PBX’s public IP address. Leave the port as 5060 for both chan_sip and chan_pjsip setups.

Step 3: Go to Interconnection -> Trunk Group – Add and create a Trunk Group.

Step 4: Go to Interconnection -> Trunk Group – Manage and add the Primary IP Address for your new Trunk Group. Set Delivery Type to 11DIGITS.

Step 5: Go to Interconnection -> SIP Registration and write down the credentials for one of the SIP credentials you wish to use to register your new trunks.

Step 6: Go to Inbound -> DIDs – Manage and select each telephone number. Then set the Trunk Group to the SIPREG Trunk Group you chose in the previous step. Click Update button.

Step 7: Wait 15 minutes for the new IP and Trunk Group settings to propagate to SBC nodes.

FreePBX PJsip Setup with BulkVS Registration

On your Incredible PBX server, navigate to Connectivity -> Trunks after logging into the FreePBX GUI as admin. Choose Add a PJsip trunk. Name the trunk BulkVS and then click on the pjsip Settings tab. Fill out the form as shown below substituting the BulkVS registration account name you chose above. Any of the three SIP registrations offered for your account under Interconnection -> SIP Registration in the BulkVS portal will work as long as you use the matching password.

Next, click on the Advanced tab and enter the following in the Match (Permit) field.

162.249.171.198,76.8.29.198,69.12.88.198,192.9.236.42,52.206.134.245

In the Codecs tab, enable ULAW and ALAW. Then click Submit and reload your dialplan.

With PJsip registrations, you may also need to add the following lines to the end of extensions_custom.conf in /etc/asterisk using your actual DID. Then reload your dialplan: asterisk -rx "dialplan reload"

[from-sip-external]
; BulkVS
exten => 18005551212,3,Goto(from-trunk,${DID},1)

VoIP.ms Messaging Services

One of our favorite VoIP.ms features is the variety of SMS and MMS messaging options they provide AT LOW COST. Virtually all of their DIDs now support messaging. With incoming messages, you have the choice of routing the messages to an email address, another SMS destination, the VoIP.ms Message Portal, an SMS URL callback destination, and now an SMS SIP account. The steps below set up SMS SIP messaging with Incredible PBX 2027. You also can send quick messages in response to incoming calls using your Clearly Anywhere softphone.

Configuring VoIP.ms for SMS SIP Messaging

Prerequisites: DID supports messaging, SMS SIP messaging enabled on the DID

First, use our VoIP.ms signup link to create a VoIP.ms account. Next, create an Asterisk SubAccount using the SIP protocol with User/Password Authentication. In the Security section, enter the public IP address of your PBX, and Save your Settings. Next, acquire a DID in the VoIP.ms portal. Then choose the Manage DIDs option and edit your DID configuration. For Call Routing, select the SIP/IAX option and pick your SubAccount. Choose a DID POP location near your PBX. In the Message Service section, enable SMS SIP Account and pick your SubAccount. Then Apply Changes.

Configuring Incredible PBX for SIP Messaging

Prerequisites: PJsip VoIP.ms Trunk, PJsip Extension for SMS, sms-in and sms-out Contexts

Both PJsip Trunks and PJsip Extensions in FreePBX now support a Messages Context option in the Advanced tab of the setup GUI. Using the sms-in and sms-out contexts documented below, FreePBX now can process incoming and outgoing SMS messages. A typical use case in the Incredible PBX 2027 would be to quickly respond to an incoming call to the Clearly Anywhere app on your smartphone to indicate that you were in the midst of another call and would return the caller’s call. It is anything but a robust SMS messaging application for your smartphone, but it is a welcome addition for many mobile users that have to juggle both cellphone calls and office calls forwarded from a PBX to your smartphone. VoIP.ms has developed an excellent SMS Management Portal that is included in the VoIP.ms Dashboard. It allows you to read, respond, and manage SMS messages sent to your VoIP.ms DIDs.

Once you have completed the necessary setup steps on the VoIP.ms side, there are three steps to activate SMS SIP messaging with Incredible PBX 2027: (1) create and register your VoIP.ms PJsip Trunk, (2) create and configure a PJsip extension to receive incoming calls and SMS messages, (3) add the sms-in and sms-out contexts to extensions_custom.conf dialplan.

(1) Create a PJsip Trunk for VoIP.ms in the FreePBX GUI to process calls and SMS messages:


In the PJsip Settings tab, fill out the General tab. The Username will be your VoIP.ms account number followed by an underscore and then the name of the SubAccount you created above, e.g. 12345_mypbx. The Password will be the password you assigned to your VoIP.ms SubAccount. For SIP Server, enter VoIP.ms POP assigned to your DID, e.g. atlanta1.voip.ms. Accept the remaining defaults in the General tab. Click on the Advanced tab and scroll down to Message Context and enter sms-in. Click Submit and Reload your Dialplan.

(2) Next create a PJsip Extension in the FreePBX portal. This will be used to process calls and send SIP messages. NOTE: Incredible PBX 2027 ships with a number of extensions preconfigured. Only extension 701 is a PJsip extension. Do NOT use the others. If needed, create an additional PJsip extension for messaging. The General tab should look something like this:



Click on the Advanced tab and scroll down to Max Contacts and enter a number that is one more than twice the number of phones that will be connected simultaneously to this extension. For example, if you have 3 smartphones connecting to this extension, enter 7. Scroll down to Message Context and enter sms-out. Click Submit and Reload your Dialplan.

(3) Finally, cut-and-paste the following code into the bottom of extensions_custom.conf in the /etc/asterisk directory:

[sms-out]
exten => _.,1,NoOp(Outbound Message dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
;
; add your VoIPms info in the next 3 lines
exten => _.,n,Set(VOIPMS_ACCOUNT="123456_subacct")
exten => _.,n,Set(VOIPMS_POP="atlanta.voip.ms")
exten => _.,n,Set(VOIPMS_TRUNK="VoIPms-PJsip") ; actual VoIP.ms trunk in FreePBX
;
exten => _.,n,Set(NUMBER_TO=${CUT(CUT(MESSAGE(to),@,1),:,2)})
exten => _.,n,Set(EXTENSION_FROM=${CUT(CUT(MESSAGE(from),@,1),:,2)})
;
; Now map your sending extensions EXTENSION_FROM to corresponding DIDs NUMBER_FROM
exten => _.,n,Set(CASE_701=6005550101) ; ext 701 msgs originate from 6005550101
exten => _.,n,Set(CASE_702=6005550102) ; ext 702 msgs originate from 6005550102
exten => _.,n,Set(CASE_703=6005550101) ; ext 703 msgs originate from 6005550101
;
exten => _.,n,Set(NUMBER_FROM=${CASE_${EXTENSION_FROM}})
exten => _.,n,Set(ACTUAL_FROM="${NUMBER_FROM}" )
exten => _.,n,Set(ACTUAL_TO=pjsip:${VOIPMS_TRUNK}/sip:${NUMBER_TO}@${VOIPMS_POP})
exten => _.,n,MessageSend(${ACTUAL_TO},${ACTUAL_FROM})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,Hangup()
;-------------------------------------------------------------------------

[sms-in]
exten => _.,1,NoOp(Inbound SMS dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
;
; enter your default incoming SMS extension below
; if you want SMS messages delivered to multiple extensions,
; clone additional MessageSend lines below with extension numbers
exten => _.,n,Set(EXTENSION=701)
;
exten => _.,n,Set(ACTUAL_FROM=${MESSAGE(from)})
exten => _.,n,Set(HOST_TO=${CUT(MESSAGE(to),@,2)})
exten => _.,n,Set(NUMBER_TO=${MESSAGE_DATA(X-SMS-To)})
exten => _.,n,MessageSend(pjsip:${EXTENSION}@${HOST_TO},${ACTUAL_FROM})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,Hangup()
;-------------------------------------------------------------------------

In the pasted [sms-out] context, insert your actual VOIPMS_ACCOUNT, VOIPMS_POP, and VOIPMS_TRUNK name in the lines provided. Then map each extension from which you wish to send SMS messages to a VoIP.ms DID on your PBX in the lines provided. In the pasted [sms-in] context, enter the EXTENSION number which should receive incoming messages from the PJsip trunk in which you designated [sms-in] as the Message Context. There is no magic to the [sms-in] context name. If you have more than one PJsip trunk, simply create additional incoming contexts (such as [sms-in-2]) for each additional trunk and clone the [sms-in] code designating the desired extension to receive incoming messages from each DID. For the [sms-out] context, it can be used as the Message Context for multiple extensions that should be enabled to send outbound SMS messages.

Save the file, and reload the Asterisk dialplan: asterisk -rx "dialplan reload"

Once all the pieces are in place, SMS messages sent to your VoIP.ms DID will be delivered to the FreePBX trunk registered to the SMS SIP destination specified in your VoIP.ms DID setup. And here’s one more tip. If you happen to have a Yealink T46G (not T48G) or a Grandstream GXV phone that is also registered to that extension, the messages will also pop up on your desktop phone with an alert tone. On Grandstream GXV Android phones, we recommend dragging the SMS app to the main screen so that the incoming message count appears beside the SMS icon when new messages are received.

FreePBX Inbound & Outbound Route Configuration

Finally, we need to tell FreePBX how to route calls and messages into and out of your PBX. In the FreePBX GUI under Connectivty -> Inbound Routes, add a new route for BulkVS specifying the 11-digit DID you purchased from BulkVS. Choose a Destination for the incoming calls, save your settings, and reload the dialplan. Repeat this process for your VoIP.ms DID making sure to enable faxing if you’ve completed the fax tutorial.

Next, navigate to Connectivity -> Outbound Routes and modify the default Outbound Route for all outgoing calls. Assign the BulkVS trunk as the first entry in the call sequence and the VoIP.ms trunk as the second entry. In the Dial Patterns tab, you would want match patterns for 1NXXNXXXXXX and NXXNXXXXXX. For the latter entry, be sure to add a Prepend entry of 1. Then save your settings and reload the dialplan.

Originally published: Monday, July 10, 2023



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