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The Most Versatile VoIP Provider: FREE PORTING

Introducing OSS Endpoint Manager for FreePBX 16 & Incredible PBX 2027


If you’ve been searching for a free, open source provisioning system for your SIP telephones and you’re using Incredible PBX 2027 or other FreePBX® 16 platforms, your prayers have been answered thanks to the hard work of Bill Simon. What this buys you (for free) is a quick way to configure SIP devices from most of the major players shown above including Aastra, AudioCodes, Cisco, Linksys, CloudTC, Digium, Grandstream, Intelbras, Mitel, Norphonic, Patton, Polycom, Snom, Thomson, Unidata, Xorcom, and Yealink.

To get started, log into FreePBX on your PBX platform and click on the Settings tab. If you see OSS Endpoint Manager as you will on most Incredible PBX 2027 installations, you can skip installing OSS Endpoint Manager and move on to the next step. Otherwise, here are the steps to install OSS EPM 16. Using SSH, log into your PBX as root and issue the following commands:

cd /var/www/html/admin/modules
wget https://filedn.com/lBgbGypMOdDm8PWOoOiBR7j/OSS-EPM/ossepm16.tgz
tar zxvf ossepm16.tgz
fwconsole ma install endpointman
rm -f /tmp/*
fwconsole reload 

Once you have OSS Endpoint Manager installed, return to the FreePBX GUI and open the application under the Settings tab. Read the reminder that you’re not using Sangoma’s commercial endpoint management product. Then click on the pull-down box in the right margin. Now let’s get the templates for the various SIP devices loaded onto your server.

  1. On the right pull-down, choose EndPoint Manager Settings.
  2. Adjust the entries as desired to set up your server as the host.
  3. For Package Server, enter: https://ossepm.incrediblepbx.com/
  4. Click in any other field to save your entries.
  5. On the right pull-down, choose Template Manager.
  6. Click the displayed link to download the latest templates from GitHub.
  7. On the right pull-down, choose Package Manager.
  8. Click Check for Updates.
  9. Click Install button beside each desired Brand.


The rest is a walk in the park. Plug in your SIP devices and perform the manufacturer’s steps to place the device in configuration mode. Insert the credentials to access OSS EPM on your PBX. Then sit back and enjoy the show.

Originally published: Thursday, February 1, 2024



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



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.
 



Interconnecting Incredible PBX Servers with PJsip


Much has been made of the lack of multi-tenant functionality with Asterisk® and FreePBX®. Today we’ll show you how to interconnect as many Asterisk servers as you like and make transparent calls between all of them at zero cost. Setup time is under 5 minutes per server. When you couple PBXs with distributed infrastructure and inexpensive cloud servers costing $25 or less a year and on-premise virtual machine platforms including Proxmox, VMware ESXi, VirtualBox, Windows 11 WSL, and Mac UTM, it makes little sense in terms of reliability and redundancy to put all your eggs in one basket with a MultiTenant telephony solution.

PJsip Interconnection Design

Our basic design goes like this. Create as many decentralized Incredible PBX platforms as you need to cover the various locations of your business. Whether that’s two cities or eighty-nine, the design is the same. Next, use the same extension design at every site. For example, in each location choose 3-digit extensions in the 100 range for management offices and the receptionist, in the 200 range for human resources, in the 300 range for IT services, and so on. Next, number your locations. For nine or fewer cities, use single digits. For up to 89, use two digits beginning with 10. We actually use the last octet of our server’s OpenVPN IP address for the dialing prefix. For our example, we’ll use 1 for the Atlanta office, 2 for Dallas, and 3 for Boston. When someone in Atlanta wishes to call the main human resources number in Dallas, the user would dial 2200. For Boston, they’d dial 3200. And, when someone in Dallas or Boston wishes to reach the main HR number in Atlanta, they’d dial 1200. Assuming the receptionist in each location has extension 100, callers in Dallas would dial 1100 for Atlanta and 3100 for Boston. Hopefully, you get the idea now. Sketch these out on a piece a paper until it all makes sense to you.

You’ll need static IP addresses for each of your Incredible PBX servers. But, if one or more of your servers are sitting behind a corporate firewall, that’s not a deal breaker. Simply set up an OpenVPN network and assign a private OpenVPN IP address to each server. Then use these OpenVPN IP addresses in interconnecting your servers rather than static, public IP addresses. OpenVPN addresses are automatically whitelisted in the Incredible PBX firewall. For other IP addresses, you’ll need to whitelist them using the add-ip utility in the /root folder of each of your PBXs. Be sure to also add the IP addresses of the servers to the ignoreip list in /etc/fail2ban/jail.conf. Note in our design that, for security reasons, PJsip trunks must be created on BOTH the sending and receiving PBXs before interconnected calls between the servers will successfully complete. You wouldn’t want strangers making PJsip calls through your servers. In the alternative, if you only want to enable outbound calls from one PBX to another, you can avoid creating a PJsip trunk on the destination server and instead enable Outbound Authentication on the originating server. In this case, you also will need to enter extension credentials from the destination server in the PJsip Username and Secret fields. But let’s continue on assuming you want calls to flow in both directions.

Using our 3-city example, let’s begin by configuring the Atlanta PBX using the FreePBX GUI. Login to the web interface as admin and navigate to Connectivity -> Trunks. We want to create two PJsip trunks, one to connect to Dallas and another to connect to Boston. Similarly, in Dallas, we’ll create PJsip trunks for Atlanta and Boston. And, in Boston, we’ll create PJsip trunks for Atlanta and Dallas.

Configuring PJsip Trunks For Each Location

On the Atlanta PBX, click the Add Trunk button and choose Add SIP (chan_pjsip) Trunk. For the first PJsip trunk, in the General tab, let’s keep it simple and enter Trunk Name as Dallas. Leave the remaining default settings and click the pjSIP Settings tab. Leave all of the default settings as they are in both the Advanced and Codecs tabs. In the General tab, change the Authentication and Registration settings to None. For the SIP Server IP address, enter the IP address (or OpenVPN address) of the Dallas server. For the SIP Server Port, enter 5061. Finally, change the Context entry to from-internal. That’s all the settings you’ll need. So click Submit and Apply Config when prompted. Repeat the drill for Boston.


Configuring Outbound Routes For Each Location

Next, we need to create two Outbound Routes for calls to Dallas and Boston. Navigate to Connectivity -> Outbound Routes. For each city, click Add Outbound Route. Name the Outbound Route using the city name, e.g. Dallas. Change the Time Zone, if required. For the Trunk Sequence, choose the Trunk with the matching City name from the pull-down menu. Next, click the Dial Patterns tab. Here you’ll need to decide which calls you want to permit between the Headquarters office and Dallas. For the Dallas extensions, you would enter 2 in the Prefix field and NXX in the Match Pattern field. If there are 4-digit numbers on the Dallas PBX that you want to enable for callers from Atlanta, you add another Dial Pattern with the 2 Prefix and NXXX for the Match Pattern. If you want callers from Atlanta to be able to use Outbound Routes on the Dallas PBX to make PSTN calls, you would add another Dial Pattern with the 2 Prefix and NXXNXXXXXX for the Match Pattern. Similar additions could be made for international calls assuming you allow them on your Dallas PBX. Once you’ve added the desired Match Patterns, click Submit and Apply Config when prompted. Repeat these steps to add an Outbound Route for Boston, and you’re finished on the Atlanta PBX.


Repeating the Setup Process in Other Cities

Now simply repeat the steps above on each of your PBXs. For PJsip trunks, you’ll need to create one for every other city in your operation. And, for Outbound Routes, do the same. Once all of your PJsip trunks are activated, you can verify functionality in the Asterisk CLI with this command: pjsip show aors. Once the Status for each city displays as Avail, you can begin making test calls between the servers using a phone connected to each PBX.

Originally published: Tuesday, May 16, 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.
 



A Walk on the Wild Side: Meet Incredible PBX for MX Linux



This week’s adventure is just for fun. It’s a VirtualBox® Beauty featuring Incredible PBX®, Asterisk® 18, FreePBX® 15, and MX Linux 21.1. If you’re unfamiliar with MX Linux, it is one of the slickest Debian aggregations because it seamlessly integrates a desktop GUI featuring Xfce Desktop, KDE, and fluxbox into a Debian 11 platform with native support for LibreOffice, Firefox, Thunderbird, SAMBA, and some awesome games and music utilities including Clementine. If all of these apps sound like Greek, then you owe it to yourself to spend an evening or weekend exploring MX Linux while enjoying most of the integrated functionality of the Incredible PBX 2021 platform. Did we mention that printer integration is seamless with most of the major brands so you can enjoy the office suite without missing a beat by abandoning your Mac or Windows desktop.


What would we use this for? The short answer is not for production. The long answer is it’s a perfect complement for a traveling Chromebook where you want to connect back to your PBX mothership while retaining full functionality of a desktop computer back home at zero cost. VirtualBox is free. Incredible PBX is free. MX Linux including our bundled .ova image is free. And an 8GB Chromebook with 64GB of eMMC storage is dirt cheap. So what’s not to like?

Installing Oracle VM VirtualBox

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

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

Installing the Incredible PBX for MX Linux Image

To begin, visit our SourceForge page and follow the steps to assemble the 8GB+ .ova image of Incredible PBX with MX Linux.

Next, copy it to your VirtualBox desktop machine and double-click on the Incredible PBX .ova image. Then click Import. Once the import is finished, you’ll see a new Incredible PBX for MX Linux virtual machine in the VM List of the VirtualBox Manager Window. Let’s make a couple of one-time adjustments to the Incredible PBX configuration to account for possible differences in sound and network cards on different host machines.

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

Running Incredible PBX for MX Linux in VirtualBox

Once you’ve imported and configured the Incredible PBX for MX Linux Virtual Machine, you’re ready to go. Highlight the Incredible PBX virtual machine in the VM List on the VirtualBox Manager Window and click the Start button. The standard Linux boot procedure will begin and, within a few seconds, you’ll get the familiar Linux login prompt. Choose Other from the pull-down list and enter root for your username and password for your password.


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 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.

Once the Xfce Desktop displays, click on the Xfce Terminal Emulator icon in the left menu bar which will drop you into the Linux CLI.

First, create a new end-user account which we will explain in a minute. Just enter the following command: adduser nerd.

Next, update your admin password for Incredible PBX web access: ./admin-pw-change.

Then 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.

Finally, run ./timezone-setup to set the correct time zone for your virtual machine.

Make note of your virtual machine’s local IP and time by typing pbxstatus. You’ll need this IP address to log into the Incredible PBX GUI using Firefox on the Xfce Desktop.

Setting the Date and Time with VirtualBox

If pbxstatus still shows an incorrect time, manually set the date and time and then update the hardware clock. Here’s how assuming 05160709 is the month (May), day (16), and correct time (7:09 a.m.) of your server:

date 05160709
clock -w

Returning to the Xfce Desktop

Now let’s return to the Xfce Desktop and close the terminal window. Type exit and press ENTER.

MX Linux won’t let you run many of its desktop applications while logged in as root. To switch to the nerd user we created above, do the following. Click on the Application Menu icon in the bottom left column. Then click the Lock Screen icon at the top of the applications menu. When the familiar login screen reappears, log back into your virtual machine as nerd using the password you set up when you created the account. Now explore all the fabulous features that MX Linux offers. And follow our Incredible PBX tutorial to get up and running in minutes.

Originally published: Monday, May 16, 2022



Need help with Asterisk? Visit the VoIP-info Forum.


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



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

Deal of the Year: OBi2182 Color WiFi Phones now $59.99



One year and counting… If you still have some Google Voice numbers lying around or if you’d like to join the party a little late, now is your second chance. Amazon now is selling the OBi2182 phones to U.S. residents with native Google Voice as well as WiFi, Bluetooth, HD Voice, speakerphone, and a power adapter currently selling for $69.99. Hold, transfer, and conferencing support are included in the phone. Here’s our Amazon referral link to keep the lights on.

We were big fans of OBiHai once upon a time. But then they sold out to Polycom. And they subsequently sold out to Plantronics. And support went from bad to worse along the way. But things appear to have stabilized so this offering may just be one of the best deals on the planet. Not only do you get unlimited free calling throughout the U.S., Canada, and Ukraine but you also get some of the best pricing ever on most international calls. While we no longer have any dealings with ObiHai, Polycom, or Plantronics, we know many of our followers were left high and dry when Google pulled the plug on Google Voice for Asterisk®. Here’s a tip.

Adding one or more of these phones makes a perfect supplement to your Asterisk platform making it easy to add free calling to your home or home office PBX. And it’s not too late to order a Google Voice number if you’re new to the game. Configuring a SIP or PJsip connection as well as Bluetooth connectivity to your cellphone is just as simple as adding Google Voice trunks using the OBiTalk web interface shown below. Trunk setups take less than a minute apiece. The phone supports 12 lines. And WiFi calls are crystal-clear thanks to HD Voice.


Grab an OBi2182 while you still can! An Admin Guide to the OBi2182 is available here.

Originally published: Monday, May 9, 2022



Need help with Asterisk? Visit the VoIP-info Forum.


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Interconnecting Asterisk Servers with PJsip and OpenVPN


It’s been several years since we discussed interconnecting Asterisk® servers so today we want to do a version refresh using PJsip Trunking. We also want to show you how easy it is to secure the communications path by setting up the trunks using OpenVPN connections. When we’re finished, you’ll have a FREE way to call between sites using FreePBX® Outbound Routes. Because Incredible PBX comes preconfigured with all the components you’ll need, we’ll use that platform to further simplify the deployment. We’ll be interconnecting two Asterisk servers today, but you can use the same methodology to connect numerous sites.

Deploying OpenVPN with Asterisk Servers

To begin, you’ll want to get all of the sites configured with a virtual private network using OpenVPN. Our tutorial will walk you through the process. Keep in mind that all current releases of Incredible PBX are preconfigured to let you drop in your OpenVPN client credentials by naming them incrediblepbx.ovpn and copying the file into the /etc directory. Rebooting your server will bring up the virtual private network with a 10.8.0.x IP address.

Configuring PJsip Trunks on Your Asterisk Servers

If you remember yesteryear’s knuckle drill configuring SIP or IAX trunks for Asterisk connectivity, you’re in for a pleasant surprise using PJsip trunking with FreePBX. Using the GUI, create a new PJsip trunk for every site to which you want to establish a connection. A similar PJsip trunk must be created on the other site as well. If you’re just interconnecting two servers, then using the City locations for the Trunk Names will suffice. But, if there are more than two servers, specify unique names for each end of every PJsip connection, e.g.

NewYork1 <-> London1
NewYork2 <-> Washington1
NewYork3 <-> Miami1
London2 <-> Washington2
London3 <-> Miami2
Washington3 <-> Miami3

For today, we’ll interconnect a server in New York and London, but we’ll plan for the future and use London1 for the Trunk Name on the New York server and NewYork1 for the Trunk Name on the London server.

Let’s begin by configuring the London1 trunk on the New York server. After entering the London1 Trunk Name in the General tab, click on the pjsip Settings tab. In the General tab, leave the defaults in the first four fields. Then change the Registration field to None. For the SIP Server field, enter the OpenVPN IP address of the London server, e.g. 10.8.0.2. Because we’re using Incredible PBX, the PJsip port is 5061 so enter that in the SIP Server Port field. If you’re using a different flavor of FreePBX, enter the appropriate port number for PJsip on your platform. Next, click on the Advanced tab and enter the London server’s OpenVPN address in the Match (Permit) field, e.g. 10.8.0.2. In the Codecs tab, make note of the enabled codecs and make certain that the entries match on all of your servers. Click the Submit button to save your settings and then reload your dialplan.

Now let’s configure the NewYork1 trunk on the London server. After entering the NewYork1 Trunk Name in the General tab, click on the pjsip Settings tab. In the General tab, leave the defaults in the first four fields. Then change the Registration field to None. For the SIP Server field, enter the OpenVPN IP address of the New York server, e.g. 10.8.0.1. Because we’re using Incredible PBX, the PJsip port is 5061 so enter that in the SIP Server Port field. If you’re using a different flavor of FreePBX, enter the appropriate port number for PJsip on your platform. Next, click on the Advanced tab and enter the New York server’s OpenVPN address in the Match (Permit) field, e.g. 10.8.0.1. In the Codecs tab, make certain that the entries match those on your New York server. Click the Submit button to save your settings and then reload your dialplan. Here’s how it looks in the FreePBX GUI:


Use Outbound Routes to Interconnect Extensions

To keep things simple, let’s assume both your New York and London servers have extensions 701-705. To call an extension on the other server, we will simply dial 9 and then the 3-digit extension, e.g. dialing 9701 on the New York server will ring 701 on the London server and dialing 9701 on the London server will ring 701 on the New York server.

Create an Outbound Route on the New York server called London specifying London1 for the Trunk Sequence in the Route Settings tab. In the Dial Patterns tab, enter 9 in the Prefix field and XXX for the Match Pattern. Click Submit to save your settings and then reload dialplan.


Create an Outbound Route on the London server called New York specifying NewYork1 for the Trunk Sequence in the Route Settings tab. In the Dial Patterns tab, enter 9 in the Prefix field and XXX for the Match Pattern. Click Submit to save your settings and then reload dialplan.

If you’re interconnecting more than two sites, then you probably will want to designate a specific Prefix for every City so that users can travel between sites and use the same methodology to reach the same extensions from every location.

You can test things out using softphones by registering 701 to an extension in New York and another to the 701 extension in London. Now you can place secure and FREE calls between the sites by dialing 9701 from each softphone. Enjoy!

Originally published: Monday, May 2, 2022



Need help with Asterisk? Visit the VoIP-info Forum.


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



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



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

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

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

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

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


Planning Ahead for That Rainy Day

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

Configuring Skyetel for Incredible PBX

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

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

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

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

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

Configuring ClearlyIP SIP Trunking

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

Configuring VoIP.ms for Incredible PBX

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

Configuring Anveo Direct for Incredible PBX

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

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

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

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

Configuring a SIP Phone for Incredible PBX 2022

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

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

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

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

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

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

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

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

Configuring SendMail with Incredible PBX

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

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

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

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

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

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

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

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

chmod 777 /var/spool/mqueue
service sendmail restart

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



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

Incredible PBX Administration

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Keeping FreePBX 15 Modules Current

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

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

Where To Go From Here

Complete documentation on the ClearlyIP Devices Module is available here.

Complete documentation on the FreePBX GPL Modules is available here.

Complete documentation on the Incredible PBX additions is available here.

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

Free voicemail transcription with email delivery. Tutorial available here.

Setting Up a VPN for Your PBX: OpenVPN or NeoRouter
 

Originally published: Monday, March 28, 2022



Need help with Asterisk? Visit the VoIP-info Forum.


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Introducing OpenSIPS 3 for Incredible PBX and Debian 10


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

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

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

Choosing an Appropriate Platform for OpenSIPS

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

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

Choosing OpenSIPS Components to Deploy

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

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

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

Deploying a Debian 10 Server Platform

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

Installing OpenSIPS on a Debian 10 Server

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

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

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

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

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

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


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

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

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

Activating a SIP Server with OpenSIPS Control Panel

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

Logging into the OpenSIPS Control Panel

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



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

Creating Domains with OpenSIPS Control Panel

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



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

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


Creating Users with OpenSIPS Control Panel

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



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

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

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


Registering a Softphone to an OpenSIPS User Account

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

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

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

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

Avoiding Lockouts with NeoRouter VPN

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

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

Choosing Commercial SIP Providers

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

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

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

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

-I INPUT -s 52.41.52.34 -j ACCEPT

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

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

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

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

Introducing the VoIP Blacklist

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

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

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

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

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

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

Originally published: Monday, October 4, 2021



Need help with Asterisk? Visit the VoIP-info Forum.


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



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