Cloud Computing – Page 7

Groundwire for Android & iOS: The Best $10 You’ll Ever Spend

Monday, October 26, 2020 / 1 Comment on Groundwire for Android & iOS: The Best $10 You’ll Ever Spend

Now we’re getting down to the tough choices in VoIP softphones. You certainly need the Linphone app to make free SIP URI calls worldwide from your mobile phone. But you also need a softphone that connects to Incredible PBX® so that you never miss a traditional call to your home or office PBX. If you want tight integration with Asterisk® and FreePBX®, Clearly Anywhere is the hands-down winner, and you’ve still got until October 31 to take advantage of the $9.99 per year introductory offer. But some folks just bristle at the thought of annual fees for software licenses. If you fall into that category but still want the incoming call reliability offered by Clearly Anywhere, then the one-time $9.99 payment for Acrobits Groundwire on either the Android or iOS platform is your baby.¹ Here’s the setup process.

Configuring Incredible PBX 2020 PUBLIC for Acrobits

Before you deploy Acrobits Groundwire on your smartphones, you first must set things up on the PBX side. Here’s how the process works. First, you will need to deploy the PUBLIC version of Incredible PBX 2020 on the Internet. It requires assignment of a fully-qualified domain name (FQDN) for server access. We’ll also need to reconfigure the IPtables firewall and Fail2Ban to support Acrobits Groundwire.

Begin by choosing a cloud provider to host your public server. Here are some of our favorites starting at only a couple bucks a month with an annual subscription. CrownCloud is highly recommended. You’ll need a KVM CentOS 7 platform with at least 1GB of RAM and 10GB of storage. Once your server is operational, log in as root and follow our Incredible PBX 2020 tutorial to begin. When the install finishes, reboot your server and log back in to get the latest updates with the Automatic Update Utility.

Now you’re ready to convert your PBX into a PUBLIC-facing server. You’ll need a fully-qualified domain name for the server. If you don’t have your own domain, you can always obtain a free FQDN from a service such as NoIP.com. With your FQDN in hand, switch over to the Incredible PBX 2020 PUBLIC tutorial to complete the setup. It only takes a few minutes.

Next, we need to whitelist the Acrobits server IP addresses in the IPtables firewall. Begin by editing iptables-custom in the /usr/local/sbin folder. Just above the "# End of Trusted Provider Section" marker, add the following block of code. Then save the file and restart IPtables with the following command: iptables-restart

/usr/sbin/iptables -A INPUT -s 159.65.167.207 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.65.186.176 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.65.251.173 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.65.252.186 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.65.253.49 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.65.252.186 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.65.253.49 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.89.179.103 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 162.243.226.164 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.65.164 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.115.186 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.182.9 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.184.188 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.190.186 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.210.221 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.223.68 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 167.99.48.91 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 167.99.119.203 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 167.99.119.244 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -s 165.227.103.7 -p tcp -m tcp --dport 443 -j ACCEPT
/usr/sbin/iptables -I INPUT -p tcp -m tcp --dport 7343 -j ACCEPT
/usr/sbin/iptables -I INPUT -p tcp -m tcp --dport 4998 -j ACCEPT
/usr/sbin/iptables -I INPUT -p tcp -m tcp --dport 24998 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp --dport 4998 -j ACCEPT

Finally, we need to whitelist the Acrobits server IP addresses in Fail2Ban to be sure your server doesn’t block any of the Acrobits servers attempting to contact your PBX. Ask us how we know.² Edit /etc/fail2ban/jail.conf and scroll down to line 34 which begins with ignoreip. Add a space at the end of the existing line and then add the following string of IP addresses without a line break. Save the file and restart Fail2Ban: service fail2ban restart

159.65.167.207 159.65.186.176 159.65.251.173 159.65.252.186 159.65.253.49 159.89.179.103 162.243.226.164 165.227.65.164 165.227.115.186 165.227.182.9 165.227.184.188 165.227.190.186 165.227.210.221 165.227.223.68 167.99.48.91 167.99.119.203 167.99.119.244

Configuring FreePBX for Acrobits Groundwire Access

For every Acrobits Groundwire user, you’ll need to have a PJsip extension to which to register their softphone(s). Keep in mind that multiple softphones of a single user or multiple users can register to the same extension. The first phone to answer an incoming call gets connected to the calling party. Using a web browser, access the FreePBX GUI using your admin credentials. Navigate to Applications -> Extensions -> Add PJsip Extension. Choose a Name and Number for the Extension. Click on the Advanced tab, increase the Max Contacts entry by two for each Acrobits Groundwire softphone that will be connecting to this PJsip extension. Save your PJsip extension setup by clicking Submit. Then Reload the Dialplan.

Acrobits Groundwire Setup on Smartphones

The setup process on the smartphone side is simple. Begin by purchasing the Acrobits Groundwire app from either the App Store or Google Play for your device. Once the client softphone is installed, run the app. You will be prompted for the Username (extension), Password, and Domain:Port of your PBX. If your server is sitting behind a router and you’ve elected to use OpenVPN for your connection, then you’ll need to add the actual Public IP address of your server as the Proxy in Advanced Settings. Don’t forget the :5061 suffix if you’re using a PJsip extension on your PBX.

Once connected, click on the Settings icon in Acrobits Groundwire. Verify that Push Notifications are enabled for Incoming Calls. In setting up your connection to a PJsip extension on your PUBLIC server, don’t forget to tack :5061 onto the server’s FQDN. In the Preferences tab, set a Ring Tone for incoming calls and decide whether to record calls. The Network preference by default will choose WiFi and then Cellular with automatic roll-over when needed. If you want 911 calls routed through your cell carrier instead of through your PBX, choose Number Rewriting. Then click +. For Conditions, choose Equals 911. For Actions, choose Override Dial Action and select gsmCall. Click Done.

We hope you enjoy using the Acrobits Groundwire softphone with Incredible PBX 2020.

Originally published: Monday, October 26, 2020 Updated: February 6, 2023

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

Special Thanks to Our Generous Sponsors

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

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

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

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

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

As with Vitelity, Acrobits is an Inteliquent company which provides financial support for Nerd Vittles and our Incredible PBX open source project. [↩]
TRUE STORY: We couldn’t get Groundwire to answer incoming calls and finally opened a ticket withe Acrobits. Within 15 minutes, we got a response from Gabriel Baker, who requested a log which was easily generated from the app itself. About 15 minutes after we sent Gabriel the log, he responded noting that our server was blocking the Acrobits IP address which he provided. Issuing the iptables -nL command on our end quickly revealed that Fail2Ban had blacklisted the address. Whitelisting the Acrobits IP addresses in Fail2Ban instantly solved the problem. One would have to conclude that Acrobits has some of the best tech support in the industry based upon our first-hand experience. [↩]

Oldie But Goodie: VoIP.ms, The Most Versatile VoIP Provider

Monday, October 12, 2020 / 1 Comment on Oldie But Goodie: VoIP.ms, The Most Versatile VoIP Provider

Photo credit: Mike van Schoonderwalt/pexels

We all are fortunate to have an extraordinary selection of options when it comes to VoIP Providers. For redundancy and reliability, nobody quite matches Skyetel. For FreePBX® and SIP phone integration, ClearlyIP is the hands-down winner. And, if you’re searching for the Most Versatile VoIP Provider, look no further than VoIP.ms, now with a $10 signup credit with your first deposit to kick the tires. We are thrilled that all three of these providers are Platinum Sponsors of Nerd Vittles and our open source projects. Here’s our VoIP.ms signup link.

As we have often stressed, the beauty of VoIP is not having to put all your eggs in one basket when it comes to communications. Most of the offerings we write about are free when not in use. So, unlike in the MaBell days, you lose nothing by signing up with multiple providers and enjoying the best of all worlds. Today we want to highlight what makes VoIP.ms extra special.

VoIP.ms Points of Presence

When it comes to Points of Presence (POPs), VoIP.ms covers all the bases. This matters because the closer your VoIP provider is to the physical location of your PBX, the better your calls will be. In the case of VoIP.ms, your choice of POPs is impressive. In the United States, there are multiple POPs in Atlanta, Chicago, Dallas, Denver, Houston, Los Angeles, New York, San Jose, Seattle, Tampa, and Washington, D.C. In Canada, you can choose between multiple POPs in Montreal, Toronto, and Vancouver. For our international friends, there are POPs in Amsterdam, London, Paris, and Sydney.

VoIP.ms DID Options

In addition to free number porting, VoIP.ms has an impressive array of DIDs from which to choose. They offer DIDs in virtually every state, province, and country in the world as well as toll-free and fax numbers in many locations with per minute and unlimited calling options.

Obtaining VoIP.ms SIP URIs

There are now more than 2,000 VoIP networks that support SIP URI access. Using a SIP URI dialing prefix, you can call any of the referenced networks @sipbbroker.com. The beauty of SIP URI calling is the calls typically are free worldwide regardless of duration. There are a number of ways to obtain a SIP URI for your PBX. Perhaps the easiest is to set up the PUBLIC Incredible PBX cloud platform that we previously introduced. Then you can create as many SIP URIs as you like, and they can be used to perform any task that’s available with Asterisk®. If you’re not quite ready to make that leap, virtual SIP URIs are available from VoIP.ms for 25¢ a month. SIP URIs are treated just like DIDs with incoming calls billed at ⅒¢ per minute.

VoIP.ms Incoming Call Routing

For call routing, the options are equally impressive. In fact, you may decide you don’t need a PBX at all. VoIP.ms supports SIP and IAX2 trunk registrations using credentials or IP address, a customizable IVR, a call queue, conferencing, call forwarding, SIP URI forwarding, call hunting, ring groups, callback, DISA, custom music on hold, voicemail transcription, and impressive call failover options for each of the following conditions: busy, unreachable, and unanswered calls. You can also perform CNAM lookups on incoming calls as well as setting the ring time, customizing each DID’s voicemail setup, and choosing whether to record calls.

VoIP.ms Outbound Call Pricing

No article would be complete without some mention of pricing. VoIP.ms is not the cheapest provider on the planet. But, as the old saying goes, you get what you pay for. Calls to toll-free numbers are free. While that may seem obvious, it is the exception rather than the rule in the VoIP world. Calls to US-48 destinations are a penny a minute and are billed in six second increments. Calls to most Canadian destinations are about a half-cent per minute. Calls to Mexico are just over a penny a minute billed in one minute increments. International calls vary based upon destination and latest published rates. International calls are blocked unless you enable them, and you can choose the countries you wish to enable as well as a dollar limit.

VoIP.ms Messaging Services

One of our favorite VoIP.ms features is the variety of SMS and MMS messaging options they provide. Virtually all of their DIDs now support messaging. With incoming messages, you have the choice of routing the message to an email address, another SMS destination, the VoIP.ms Message Portal, an SMS URL callback destination, and now an SMS SIP account. Our tutorial below sets up SMS SIP messaging with Incredible PBX® 2020 or 2021. You then can send quick messages in response to incoming calls on your Clearly Anywhere softphone.

Configuring VoIP.ms for SMS SIP Messaging

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

First, 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 near your PBX location. 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 2020 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: (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 FreePBX 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 ships with a number of chan_sip extensions preconfigured. Do NOT use these. You need to create a PJsip extension. 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.

Our special thanks and much of the credit for this SMS/SIP solution for Asterisk goes to Stepan Novotill and the participants in this thread on the VoIP-Info Forum.

Signing Up for VoIP.ms Service

Please consider using the Nerd Vittles referral link should you decide to sign up for VoIP.ms services. These referral commissions help to defray the costs of maintaining Nerd Vittles and the Incredible PBX open source project. Many thanks.

Originally published: Monday, October 12, 2020 Updated: Saturday, August 28, 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.

Android Alert: Unmasking Your Hidden SIP Phone

Tuesday, September 1, 2020 / 2 Comments on Android Alert: Unmasking Your Hidden SIP Phone

Perhaps the single most important component in the VoIP toolbox for any PBX deployment that includes traveling users is the softphone whether it’s deployed on notebook computers or smartphones. We’re kicking off football season with reviews of some of our favorite SIP softphones for mobile users. We’ve previously written about Linphone and Zoiper and Telephone. We’ve also sung the praises of earlier releases of Google’s Pixel smartphone which makes a perfect VoIP companion even without a cellphone provider. Today we’re passing along an Android tip from @w1ve on the VoIP-Info.org Forum for anyone needing PBX connectivity while away from the home or office. And it won’t cost you a dime so long as you already have an Android smartphone with either a Wi-Fi connection or a data plan with any cellphone carrier.

You may be unaware that Android has been shipping with a native SIP phone since as far back as Android Marshmallow which will be five years old next month. Some phone manufacturers such as Chinese-owned Lenovo/Moto disable the SIP functionality, but many do not including Google, Samsung, and OnePlus. If you’ve deployed an Incredible PBX 2020 PUBLIC server in the Cloud with a PJsip extension, the beauty of this discovery is that you’ll always have VoIP connectivity through your PBX with only a WiFi connection. No SIM is required!

To determine if your Android smartphone includes the SIP dialer, simply open the native Phone app and tap the three vertical bar icon at the top of the dialer menu. From the pull-down menu, choose Settings, Calls, and Calling Accounts. If your Android smartphone includes support for the native SIP dialer, there will be a SIP Accounts option in the menu.

TIP: Even if the SIP Menu doesn’t appear on your smartphone, it may be that the manufacturer simply disabled the menu without actually trashing the SIP functionality as Lenovo has done. To determine whether the menu has simply been hidden, install QuickShortcutMaker from the Play Store. Run the app and scroll down and tap the SIP Settings option and click Test. It should bring up the SIP Accounts menu shown below.

Tapping SIP Accounts and then + icon will open a dialog to add a SIP account to your phone.

With an Incredible PBX 2020 PUBLIC server, the entries should look like the following:

Username: Your PJsip extension
Password: Your PJsip password
Server: FQDN of Your PBX
Optional Settings -> Port: 5061

Keep in mind that an FQDN for the Server address is required with an Incredible PBX 2020 PUBLIC server. If your server is using a dynamic IP address, you also would want to configure the FQDN using a Dynamic DNS service and refresh that FQDN periodically on your PBX using a cron job. Once you’ve entered your credentials, tap SAVE to activate the SIP account on your smartphone. It should then appear in the SIP Accounts window as shown above.

Next, you have some choices to make as to how the SIP account is actually used. As you can see from our setup (shown above), we allow outbound calls using either the SIM card or the SIP phone, and the phone will prompt for a choice whenever you make a call. We also have activated Inbound SIP calling which, as the dialog explains, uses some battery life. Finally, if you elect to use a chan_SIP extension on your PBX, make certain that you have enabled NAT Mode in the Advanced tab, or you will experience one-way or no audio on calls. This is not required with PJsip extensions. One of the other beauties of PJsip extensions is that you can assign this extension to multiple SIP devices including softphones and desktop phones so long as you increase the Max Contacts entry in the Advanced tab for the PJsip extension. In this way, you can answer incoming calls on your desktop phone when you’re at home or in the office and answer the same calls on your smartphone when you’re out and about. Enjoy!

Originally published: Tuesday, September 1, 2020

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

Special Thanks to Our Generous Sponsors

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

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

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

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

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

The Asterisk Superfecta: Incredible PBX 2020 Finishes First

Monday, August 24, 2020

At the track, picking the first four winning horses pays big bucks, and you’ll never forget winning the Superfecta. When it comes to unified communications, it’s equally advantageous to have cloud, on-premise, VirtualBox, and VMware offerings from which to choose. With the latest rollouts of Incredible PBX® 2020, you literally have the Asterisk® Superfecta. With the exception of the cloud offering, you don’t even have to buy a ticket. The CrownCloud offerings for $25/year in Los Angeles, Atlanta, Germany, and the Netherlands have no equal.¹ And soon we’ll roll out IncredibleBackup2020 so you can move transparently between all the platforms.

The current Incredible PBX 2020 offerings include the following:

Incredible PBX 2020 for CentOS 7 and Raspbian 10
Incredible PBX 2020 for VirtualBox®
Incredible PBX 2020 for VMware®
Incredible PBX 2020 for CrownCloud

Snapshot images are available for Raspbian 10, VirtualBox, VMware, and CrownCloud which means all of these builds can be installed in just a few minutes. The Incredible PBX 2020 offering for CentOS 7 is built from source code on the fly which typically takes about 30 minutes once you have your CentOS 7 Minimal platform in place. And, of course, Incredible PBX 2020 for CentOS 7 can be installed on any platform offering a CentOS 7 base image. Unlike most of the other Asterisk aggregations, all Incredible PBX 2020 offerings include both Asterisk and FreePBX® source code which makes upgrades incredibly flexible whether upgrading Asterisk 16 to the latest, trying out Asterisk 17, or updating FreePBX 15 modules.

Upgrading to the Latest Asterisk 16 Release. All Incredible PBX 2020 platforms include an Asterisk 16 upgrade script in the /root folder which will build you the latest release of Asterisk 16 from source code. It also provides the flexibility to choose which modules to include in the build as well as whether to enhance performance on a particular platform by turning on the BUILD_NATIVE Compiler Flag, a feature that is not available with packaged RPMs.

Upgrading to the Latest Asterisk 17 Release. Appreciating that some of you like to live on the bleeding edge, we now have released an Asterisk 17 upgrade script for the CentOS 7, VirtualBox, VMware, and CrownCloud platforms. Just issue the following commands, and you’ll be off to the races.

cd /root
wget http://incrediblepbx.com/upgrade-asterisk17.tar.gz
tar zxvf upgrade-asterisk17.tar.gz
rm -f upgrade-asterisk17.tar.gz
./upgrade-asterisk17

Updating Your FreePBX 15 Modules. While the Module Admin utility in FreePBX gives you the flexibility to update any or all of your FreePBX modules, a simpler method is available to do a bulk upgrade from the Linux CLI. After logging into your PBX as root, issue these commands:

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

Regardless of the Incredible PBX 2020 platform you choose, always remember our Automatic Update Utility will keep your PBX current, running reliably, and as bug-free as we possibly can make it. No one else in the VoIP community has anything close. Enjoy the free ride!

Originally published: Monday, August 24, 2020

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

Special Thanks to Our Generous Sponsors

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

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

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

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

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

We don’t make a dime on the CrownCloud platforms, and the on-premise, VirtualBox, and VMware offerings all are free. [↩]

Harnessing the Cloud to Start An Incredible PBX Business

Monday, July 27, 2020

If you’ve ever wanted to start your own VoIP business and earn some big bucks through consulting and hosting cloud-based PBXs, now’s your chance. One of the requests we often receive from those that deploy Incredible PBX 2020® for a living is a quicker way to produce new Incredible PBX servers on cloud platforms such as Vultr and Digital Ocean while also preserving Incredible PBX’s unique ability to upgrade source components for Asterisk® and FreePBX®. For small businesses, these cloud providers offer a perfect $5 a month platform for Incredible PBX. You can mark it up to $10 or $15 a month and make a handsome 100% to 200% profit without lifting a finger as a VoIP consultant. And Vultr and Digital Ocean will spot you a $100 credit to get the ball rolling.

Today’s solution was especially designed for those that would like to host virtual machines for customers in your own cloud account. It would work equally well for anyone wanting a quick way to create multiple Incredible PBX platforms in 5 minutes for friends and neighbors.

To begin, you’ll need to create a master image of Incredible PBX 2020 on the cloud platform of your choice using the recommended $5/month platform with CentOS 7. The July 1, 2020 or later tarball of Incredible PBX 2020 is required. Here are the Five Easy Steps:

1. Create the base Incredible PBX 2020 platform in the traditional way:

# create a secure root password to hand out to future customers
passwd
yum -y update
yum -y install net-tools nano wget tar
cd /root
wget http://incrediblepbx.com/incrediblepbx2020.1.tar.gz
tar zxvf incrediblepbx2020.1.tar.gz
rm -f incrediblepbx2020.1.tar.gz
# to add swap file on non-OpenVZ cloud platforms with no swap file
./create-swapfile-DO
# kick off Phase I install
./IncrediblePBX2020.sh
# after reboot, kick off Phase II install
./IncrediblePBX2020.sh
# set desired timezone
./timezone-setup
# optionally install Incredible Fax 2020
./incrediblefax2020.sh

2. Once you complete the Phase I and Phase II installs and optionally install Incredible Fax, log out of your server and log back in so that the Automatic Update Utility can do its thing.

3. Next, we need to configure your master image so that it can be replicated using a simple image snapshot. A snapshot is free on the Vultr platform and will cost you about $5 a month with Digital Ocean. While still logged into your server as root, issue the following commands and then shut down your server gracefully:

cd /etc/sysconfig
cp -p rules.v4.tm3 iptables
sed -i 's|#-A|-A|' iptables
touch /etc/update_hostconfig
touch /etc/update_serverconfig
halt

4. Once your server has halted, create a snapshot image of the server from Vultr or Digital Ocean dashboard. You do NOT need to preserve your Master VM once the snapshot is created.

5. Create a new virtual machine but, instead of choosing CentOS 7 as the base platform, choose the snapshot image built in the previous step. Once the 5-minute install completes, it’s ready for handover to a new customer by providing the root password from the Master Image together with the IP address of the new virtual machine.

When the new customer logs in via SSH using the root password from the Master Image, the Incredible PBX reconfiguration script will complete the setup of the new platform in a couple minutes prompting the user to change all of the passwords, resetting the ports for PortKnocker, and reconfiguring the firewall by whitelisting the customer’s IP addresses. If the build includes Incredible Fax, the customer should be instructed to change the AvantFax password. Run: /root/avantfax-pw-change. If the customer is in a different time zone, the customer should run /root/timezone-setup. Whooda thunk making money could be this easy.

The real beauty of this design is that you keep control of all the virtual machines you create. If a customer fails to pay, it’s easy to either shut down their VM or even delete it. You also can schedule automatic backups for the customer while recovering the extra $1 per month charge from the provider. In addition, if the customer ever needs hands-on support, you can use the Console link in the Dashboard without the need to whitelist your IP address. The customer still retains full control over the root password which would have to be provided.

Originally published: Monday, July 27, 2020

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

Special Thanks to Our Generous Sponsors

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

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

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

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

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

Introducing OpenVPN for Incredible PBX

Monday, July 20, 2020 / 1 Comment on Introducing OpenVPN for Incredible PBX

Star VPN Topology. Photo courtesy of ankara-gtug.org.

We’ve been wrestling with virtual private networks for more than 22 years now. Here’s a quick walk down memory lane. Our adventure began with the Altiga 3000 series VPN concentrators which we introduced in the federal courts in 1999. It was a near perfect plug-and-play hardware solution for secure communications between remote sites using less than secure Windows PCs. Cisco quickly saw the potential, gobbled up the company, and promptly doubled the price of the rebranded concentrators. Over a decade ago, we introduced Hamachi® VPNs to interconnect Asterisk® and PBX in a Flash servers. At the time, Hamachi was free, but that was short-lived when they were subsequently acquired by LogMeIn®. What followed was a short stint with PPTP VPNs which worked great with Macs, Windows PCs, and many phones but suffered from an endless stream of security vulnerabilities. Finally, in April 2012, we introduced the free NeoRouter® VPN. Version 2 still is an integral component in every Incredible PBX® platform today, and PPTP still is available as well. While easy to set up and integrate into multi-site Asterisk deployments, the Achilles’ Heel of NeoRouter remains its inability to directly interconnect many smartphones and stand-alone SIP phones, many of which now support the OpenVPN platform.

The main reason we avoided OpenVPN® over the years was its complexity to configure and deploy.¹ In addition, it was difficult to use with clients whose IP addresses were frequently changing. Thanks to the terrific work of Nyr, Stanislas Angristan, and more than a dozen contributors, OpenVPN now has been tamed. And the new server-based, star topology design makes it easy to deploy for those with changing or dynamic IP addresses. Today we’ll walk you through building an OpenVPN server as well as the one-minute client setup for almost any Asterisk deployment and most PCs, routers, smartphones, and VPN-compatible soft phones and SIP phones including Yealink, Grandstream, Snom, and many more. And the really great news is that OpenVPN clients can coexist with your current NeoRouter VPN.

Finally, a word about the OpenVPN Client installations below. We’ve tested all of these with current versions of Incredible PBX 13-13 and 16-15 as well as Incredible PBX 2020 and Incredible PBX 2021. They should work equally well with other server platforms which have been properly configured. However, missing dependencies on other platforms are, of course, your responsibility.

Building an OpenVPN Server Platform

There are many ways to create an OpenVPN server platform. The major prerequisites are a supported operating system, a static IP address for your server, and a platform that is extremely reliable and always available. If the server is off line, all client connections will also fail. While we obviously have not tested all the permutations and combinations, we have identified a platform that just works™. It’s the CentOS 7, 64-bit cloud offering from Vultr. If you use our referral link at Vultr, you not only will be supporting Nerd Vittles through referral revenue, but you also will be able to take advantage of their $100 free credit for new customers. For home and small business deployments, we have found the $5/month platform more than adequate, and you can add automatic backups for an additional $1 a month. Cheap insurance!

A more recent and less costly hosting alternative is the $25/year Crown Cloud offering that we introduced several weeks ago. It includes a free snapshot backup in the $25 annual price.

To get started, create your CentOS 7 instance and login as root using SSH or Putty. Immediately change your password and update and install the necessary CentOS 7 packages. Be sure to turn off SELinux if it is installed by default.

passwd
setenforce 0
# edit /etc/selinux/config
# insert: SELINUX=disabled
# save the SELinux config file
yum -y update
yum -y install net-tools nano wget tar iptables-services
systemctl stop firewalld
systemctl disable firewalld
systemctl enable iptables

We recommend keeping your OpenVPN server platform as barebones as possible to reduce the vulnerability risk. By default, this installer routes all client traffic through the VPN server which wastes considerable bandwidth. The sed commands below modify this design to only route client VPN traffic through the OpenVPN server.


#!/bin/bash
##filename # openvpn-install-mod
echo "      Fix script /root/openvpn-install.sh to ensure internet traffic doesn't use vpn-tunnel."
echo " "
read -p "     Press 'Enter' to continue at your own risk,  or Ctrl+c to abort."
##trap user non root
if [ "$(id -u)" -ne 0 ]; then
echo ""
echo "Must be run as root user: sudo $0" echo ""
exit 1
fi
# cd /root
echo "     Fetching latest copy of install script  /root/openvpn-install.sh from github.com/Angristan"
curl -O https://raw.githubusercontent.com/Angristan/openvpn-install/master/openvpn-install.sh
chmod +x openvpn-install.sh
echo "        running  3 sed commands to ensure only local traffic uses vpn-tunnel :-"
echo '        1st commenting-out line 857'
#### fails to complete with \\"redirect-gateway ## sed -i "s|\\techo 'push \\"redirect-gateway|#\\techo 'push \\"redirect-gateway|" openvpn-install.sh
sed -e '/redirect-gateway d/s/^/#/' -i openvpn-install.sh

echo '2nd commenting-out line 865'
###sed -i "s|push \\"redirect-gateway|#push \\"redirect-gateway|" openvpn-install.sh
sed -e '/redirect-gateway ipv6/s/^/#/' -i openvpn-install.sh

echo '3rd after line 1042 ;  newline 1043   pull-filter ignore redirect-gateway'
###sed -i 's|tls-client|tls-client\\npull-filter ignore "redirect-gateway"|' openvpn-install.sh
sed -i 's|tls-client|tls-client\npull-filter ignore "redirect-gateway"|' openvpn-install.sh

Here are the recommended entries in running the OpenVPN installer:

Server IP Address: using FQDN strongly recommended to ease migration issues
Enabled IPv6 (no): accept default
Port (1194): accept default
Protocol (UDP): accept default
DNS (3): change to 9 (Google)
Compression (no): accept default
Custom encrypt(no): accept default
Generate Server
Client name: firstclient
Passwordless (1): accept default

NOTE: On CentOS 7 platforms, edit /usr/lib/systemd/system/openvpn@.service. Scroll down to the ExecStart= line and change %i.conf to %I.conf. Then save the file. Special thanks to @mattburris for catching the error.

In the following steps, we will use IPtables to block all server access except via SSH or the VPN tunnel. Then we’ll start your OpenVPN server:

cd /etc/sysconfig
wget http://incrediblepbx.com/iptables-openvpn.tar.gz
tar zxvf iptables-openvpn.tar.gz
rm -f iptables-openvpn.tar.gz
echo "net.ipv4.ip_forward = 1" >> /etc/sysctl.conf
sysctl -p
systemctl -f enable openvpn@server.service
systemctl start openvpn@server.service
systemctl status openvpn@server.service
systemctl enable openvpn@server.service
systemctl restart iptables

Once OpenVPN is enabled, the server can be reached through the VPN at 10.8.0.1. OpenVPN clients will be assigned by DHCP in the range of 10.8.0.2 through 10.8.0.254. You can list your VPN clients like this: cat /etc/openvpn/ipp.txt. You can list active VPN clients like this: cat /var/log/openvpn/status.log | grep 10.8. And you can add new clients or delete old ones by rerunning /root/openvpn-install.sh.

For better security, change the SSH access port replacing 1234 with desired port number:

PORT=1234
sed -i "s|#Port 22|Port $PORT|" /etc/ssh/sshd_config
systemctl restart sshd
sed -i "s|dport 22|dport $PORT|" /etc/sysconfig/iptables
systemctl restart iptables

We’ve made changes in the Angristan script to adjust client routing. By default, all packets from every client flowed through the OpenVPN server which wasted considerable bandwidth. Our preference is to route client packets destined for the Internet directly to their destination rather than through the OpenVPN server. The sed commands added to the base install above do this; however, if you’ve already installed and run the original Angristan script, your existing clients will be configured differently. Our recommendation is to remove the existing clients, make the change below, and then recreate the clients again by rerunning the script. In the alternative, you can execute the command below to correct future client creations and then run it again on each existing client platform substituting the name of the /root/.ovpn client file for client-template.txt and then restart each OpenVPN client.


cd /etc/openvpn
sed -i 's|tls-client|tls-client\\npull-filter ignore "redirect-gateway"|' client-template.txt

Creating OpenVPN Client Templates

In order to assign different private IP addresses to each of your OpenVPN client machines, you’ll need to create a separate client template for each computer. You do this by running /root/openvpn-install.sh again on the OpenVPN server. Choose option 1 to create a new .ovpn template. Give each client machine template a unique name and do NOT require a password for the template. Unless the client machine is running Windows, edit the new .ovpn template and comment out the setenv line: #setenv. Save the file and copy it to the /root folder of the client machine. Follow the instructions below to set up OpenVPN on the client machine and before starting up OpenVPN replace firstclient.ovpn in the command line with the name of .ovpn you created for the individual machine.

Renewing OpenVPN Server’s Expired Certificate

The server certificate will expire after 1080 days, and clients will no longer be able to connect. Here’s what to do next:

systemctl stop openvpn@server.service
cd /etc/openvpn/easy-rsa
./easyrsa gen-crl
cp /etc/openvpn/easy-rsa/pki/crl.pem /etc/openvpn/crl.pem
systemctl start openvpn@server.service

Installing an OpenVPN Client on CentOS/RHEL

cd /root
yum -y install epel-release
yum --enablerepo=epel install openvpn -y
# copy /root/firstclient.ovpn from server to client /root
# and then start up the VPN client
openvpn --config /root/firstclient.ovpn --daemon
# adjust Incredible PBX firewall below
iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT
cd /usr/local/sbin
echo "iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT" >> iptables-custom

Running ifconfig should now show the VPN client in the list of network ports:

tun0 Link encap:UNSPEC  HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00  
     inet addr:10.8.0.2  P-t-P:10.8.0.2  Mask:255.255.255.0
     UP POINTOPOINT RUNNING NOARP MULTICAST  MTU:1500  Metric:1
     RX packets:9 errors:0 dropped:0 overruns:0 frame:0
     TX packets:39 errors:0 dropped:0 overruns:0 carrier:0
     collisions:0 txqueuelen:100 
     RX bytes:855 (855.0 b)  TX bytes:17254 (16.8 KiB)

And you should be able to login to the VPN server using its VPN IP address:

# enter actual SSH port replacing 1234
PORT=1234
ssh -p $PORT root@10.8.0.1

Installing an OpenVPN Client on Debian and Ubuntu

cd /root
apt-get update
apt-get install openvpn unzip
dpkg-reconfigure tzdata
# copy /root/firstclient.ovpn from server to client /root
# and then start up the VPN client
openvpn --config /root/firstclient.ovpn --daemon
# adjust Incredible PBX firewall below
iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT
cd /usr/local/sbin
echo "iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT" >> iptables-custom

Running ifconfig should now show the VPN client in the list of network ports:

tun0 Link encap:UNSPEC  HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00  
     inet addr:10.8.0.2  P-t-P:10.8.0.2  Mask:255.255.255.0
     UP POINTOPOINT RUNNING NOARP MULTICAST  MTU:1500  Metric:1
     RX packets:9 errors:0 dropped:0 overruns:0 frame:0
     TX packets:39 errors:0 dropped:0 overruns:0 carrier:0
     collisions:0 txqueuelen:100 
     RX bytes:855 (855.0 b)  TX bytes:17254 (16.8 KiB)

And you should be able to login to the VPN server using its VPN IP address:

# enter actual SSH port replacing 1234
PORT=1234
ssh -p $PORT root@10.8.0.1

Installing an OpenVPN Client on Raspbian

The OpenVPN client now is easy to install on the latest Incredible PBX builds for the Raspberry Pi. Log into your server as root and issue the following commands to set your time zone and install the OpenVPN client. pbxstatus should then show the 10.8.0.x VPN address in the Private IP listing.

dpkg-reconfigure tzdata
apt-get install openvpn unzip
# copy your .ovpn template into /root
# edit template and comment out setenv line
# start up the client using actual .ovpn filename
openvpn --config /root/raspi.ovpn --daemon
# adjust Incredible PBX firewall
iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT
cd /usr/local/sbin
echo "iptables -A INPUT -s 10.8.0.0/24 -j ACCEPT" >> iptables-custom
iptables-restart
pbxstatus

Installing an OpenVPN Client on a Mac

While there are numerous OpenVPN clients for Mac OS X, none hold a candle to Tunnelblick in terms of ease of installation and use. First, create a new client config on your server and copy it (/root/*.ovpn) to a folder on your Mac where you can find it. Download Tunnelblick and install it. Run Tunnelblick and then open Finder. Click and drag your client config file to the Tunnelblick icon in the top toolbar. Choose Connect when prompted. Done.

Installing an OpenVPN Client for Windows 10

The installation procedure for Windows is similar to the Mac procedure above. Download the OpenVPN Client for Windows. Double-click on the downloaded file to install it. Create a new client config on your server and copy it (/root/*.ovpn) to a folder on your PC where you can find it. Start up the OpenVPN client and click on the OpenVPN client in the activity tray. Choose Import File and select the config file you downloaded from your OpenVPN Server. Right-click on the OpenVPN icon again and choose Connect. Done.

Installing an OpenVPN Client for Android

Our favorite OpenVPN client for Android is called OpenVPN for Android and is available in the Google Play Store. Download and install it as you would any other Android app. Upload a client config file from your OpenVPN server to your Google Drive. Run the app and click + to install a new profile. Navigate to your Google Drive and select the config file you uploaded.

Installing an OpenVPN Client for iOS Devices

The OpenVPN Connect client for iOS is available in the App Store. Download and install it as you would any other iOS app. Before uploading a client config file, open the OpenVPN Connect app and click the 4-bar Settings icon in the upper left corner of the screen. Click Settings and change the VPN Protocol to UDP and IPv6 to IPV4-ONLY Tunnel. Accept remaining defaults.

To upload a client config file, the easiest way is to use Gmail to send yourself an email with the config file as an attachment. Open the message with the Gmail app on your iPhone or iPad and click on the attachment. Then choose the Upload icon in the upper right corner of the dialog. Next, choose Copy to OpenVPN in the list of apps displayed. When the import listing displays in OpenVPN Connect, click Add to import the new profile. Click ADD again when the Profile has been successfully imported. You’ll be prompted for permission to Add VPN Configurations. Click Allow. Enter your iOS passcode when prompted. To connect, tap once on the OpenVPN Profile. To disconnect, tap on the Connected slider. When you reopen the OpenVPN Connect app, the OVPN Profiles menu will display by default. Simply tap once on your profile to connect thereafter.

Installing a Web Interface to Display Available Clients

One advantage of NeoRouter is a simple way for any VPN client to display a listing of all VPN clients that are online at any given time. While that’s not possible with OpenVPN, we can do the next best thing and create a simple web page that can be accessed using a browser but only from a connected OpenVPN client pointing to http://10.8.0.1.

To set this up, log in to your OpenVPN server as root and issue the following commands:


yum --enablerepo=epel install lighttpd -y
systemctl start lighttpd.service
systemctl enable lighttpd.service
chown root:lighttpd /var/log/openvpn/status.log
chmod 640 /var/log/openvpn/status.log
cd /var/www
rm -rf lighttpd
wget http://incrediblepbx.com/lighttpd.tar.gz
tar zxvf lighttpd.tar.gz
ln -s /var/log/openvpn/status.log /var/www/lighttpd/status.log
sed -i 's|#server.bind = "localhost"|server.bind = "10.8.0.1"|' /etc/lighttpd/lighttpd.conf
systemctl restart lighttpd.service

UPDATE: On some cell phones and on Windows PCs, you may observe that you can no longer reach your favorite web sites after enabling the OpenVPN client. Luckily there’s a simple fix that allows 10.8.0.x traffic to be sent through the OpenVPN tunnel while all other traffic is routed out of your standard network connection. Here’s the fix. Make sure the .ovpn client config file includes the following lines:

pull-filter ignore redirect-gateway
route-nopull
route 10.8.0.0 255.255.255.0

Originally published: Monday, July 20, 2020 Updated: Saturday, June 25, 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.

Our discussion today is focused on the free, MIT-licensed version of OpenVPN. For details on their commercial offerings, follow this link. [↩]

A New World: Adding Cellular Extensions to Incredible PBX

Monday, July 13, 2020 / 4 Comments on A New World: Adding Cellular Extensions to Incredible PBX

Over the past few weeks, we’ve introduced a revolutionary new technology for the Asterisk® and FreePBX® community by marrying Incredible PBX® with GTI Global cellular extensions. For $20 to $35 a month, you get a new SIM card for your unlocked GSM cellphone that links all four U.S. carrier networks to a PJsip extension on your PBX. Incoming calls ring on both the cellphone and any other SIP phone connected to the same PJsip extension. Outbound calls from the cellphone get processed just like every other outbound call on your PBX. All the other bells and whistles that make Asterisk and FreePBX a one-of-a-kind platform work exactly the same way on the smartphone as they do with every other PBX extension. And your cell phone is no longer tied to a single carrier’s network. Based upon your current U.S. location, the GTI SIM finds the strongest signal by automatically selecting from the cell towers of the four major U.S. carriers: AT&T, Verizon, T-Mobile, and Sprint. And last week, we added additional icing to the cake by introducing the perfect cloud platform for Incredible PBX in either Atlanta or Los Angeles for a jaw-dropping $25 a year with a 5-minute install.

Today we want to document the setup procedure to put all the cellular pieces in place to add a GTI-enabled cellphone to your Incredible PBX platform. In a nutshell, there are preliminary configuration steps on your PBX so that GTI can verify that your PBX can be successfully integrated into their cellular network. Next, you’ll need to sign up for service with GTI Global and provide some basic information about your server configuration. Once you receive credentials from GTI Global, you’ll need to complete some additional setup steps on your PBX including creation of a GTI trunk as well as configuring inbound and outbound routes to support your cellphone calls. You’ll also need to add exceptions for each GTI DID to the [from-sip-external] context. Finally, await delivery of your SIM card and then decide how you will deploy the GTI SIM on a GSM cellphone of your choice.

GTI Global provides promotional consideration to help defray costs of our Asterisk projects.

Preliminary Configuration Steps with Incredible PBX

To connect a GTI-enabled cellphone to Incredible PBX, here are the preliminary steps:

Incredible PBX must have a static IP address on the public Internet.
Whitelist a number of GTI IP addresses in your firewall.
Create a PJsip extension for each GTI-enabled cellphone.
Create a Preliminary Chan_SIP Trunk for Registration to GTI.

These whitelist entries should be added to iptables-custom in the /usr/local/sbin directory:

/usr/sbin/iptables -I INPUT -p udp -m udp -s 99.198.122.166 --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 99.198.110.51  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 96.127.174.38  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.170  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.171  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.172  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.173  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s someid.mvnoserver.com --dport 5060:5069 -j ACCEPT

For each GTI cellphone you plan to deploy, add a PJsip extension with a very secure password:

Extension: 484xx1
Display Name: 484xx1 GTI
Secret: yourSuperSecretpassword

Advanced -> Max Contacts: 3

Also make note of the SIP port that is configured for your PJsip extensions, typically 5061. It’s shown as Port to Listen On under Settings -> SIP Settings -> PJsip Settings in FreePBX GUI.

Create a preliminary Chan_SIP Trunk for registration to GTI leaving the CallerID field empty:

Trunk Name: trunk4321

PEER Details:

username=trunk4321
type=friend
secret=yourBIGpassword
host=xxxxxx.mvnoserver.com
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
port=5062

Register string: trunk4321:yourBIGpassword@xxxxxx.mvnoserver.com:5062

Registration Information to be Provided to GTI

As part of your GTI setup procedure, in addition to your name and a reachable cellphone number, you will be prompted for the following information. If any of it is missing, open a support ticket and provide it. NOTE: you must register through our portal, or your request will be denied unless you pay the standard $250 setup fee which is waived for our users. All GTI accounts through Nerd Vittles include one free SIM card, a free trunk account to process calls to and from your GTI pjSIP extension(s), and a free DID for each SIM card. Unlimited calling and unlimited SMS/MMS messaging is included on all plans. Pricing difference is for monthly data allocations: $20 (no data), $25 (2GB), and $35 (4GB). You must pay for the first month’s service BEFORE your SIM card will be shipped.

PBX Public IP Address and Trunk Listening Port (5062)
Extension IP address (PBX public IP address)
pjSIP Extension Number
pjSIP Extension Secret
pjSIP Extension Port (5061)
Service Desired: $20, $25, or $35 monthly plan
Area Code Desired for DID
Business Name and Shipping Address for Your SIM

Configuring Your PBX with GTI Credentials

Once GTI has verified that they can communicate with your PBX, you will be provided the credentials for your GTI Trunk. Simply edit the trunk you configured above inserting the Trunkname/Username, Secret, Host, and Port for your new Trunk. Modify the Register String, save your settings, and reload your dialplan. Also edit your whitelist and change the FQDN from someid.mvnoserver.com to the host name that was provided. Then restart the firewall: iptables-restart.

Verifying Successful Extension & Trunk Registrations. Once your GTI extension and trunk are configured. You can verify successful connections in the Asterisk CLI.
For the extension: pjsip show aors
For the trunk: sip show registry

Configuring an Inbound Route for GTI Global Calls. Replacing 16785551212 with your actual 11-digit DID assigned to each cellphone, create Inbound Routes that look like this:

Description: GTIglobal 16785551212
DID Number: 16785551212
Destination: Extension -> 484001 (extension number of matching SIM)

Also add the following context at the end of /etc/asterisk/extensions_custom.conf using your DID. Then reload your dialplan: asterisk -rx "dialplan reload"

[from-sip-external]
; GTI Global
exten => 16785551212,3,Goto(from-trunk,${DID},1)

Configuring an Outbound Route for GTI Global Calls. No special outbound routes are required unless you wish to use the GTI Global trunk to process U.S. calls at no cost. In this case, configure an Outbound Route pointing to your GTI Global Trunk. NOTE: Calls using this trunk can only be made from the extensions associated with GTI Global SIMs.

CallerID and Trunking Strategy. In setting up the GTI components in FreePBX, you need to consider how you wish to process outbound calls. If you want all outbound calls including those from the GTI extensions to reflect the company’s main phone number, then you can use your default outbound route making certain that CallerID override is set with the company number in the appropriate trunks. No CallerID should be set in the outbound route.

If you want calls from the GTI extensions to use the GTI trunk with the CallerID number of the individual cellphones, then the GTI trunk needs to be moved to the top of the default route sequence with no CallerID number specified in the GTI trunk or the outbound route. Instead, set the correct 11-digit CallerID numbers in each of the GTI extensions. This will not impact outbound calls from other extensions because the GTI trunk will indicate congestion forcing the calls out through the next trunk specified in the outbound route.

I’ve Got My GTI SIM. Now What?

It’s decision time on what sort of smartphone you wish to use with your GTI SIM. If you’re a heavy data consumer then it may make sense to acquire a cellphone that supports dual SIMs and use a SIM with an unlimited data plan as your data provider and use the GTI SIM for business purposes with your PBX.

Choosing a Cellphone Platform. Our favorite deployment strategy is to take advantage of the new dual SIM offerings that allow a user to have one SIM for personal use and a second SIM (GTI) for business use. This lets you acquire the least expensive cell service with unlimited data for personal use and acquire the $20 GTI SIM with no data plan through our Incredible PBX offering. You don’t really need two data plans on the same smartphone. A couple of options worth exploring include GTI’s MVNO offering of T-Mobile unlimited data service for $35 a month and Visible (a Verizon-owned MVNO) that offers unlimited plans for as little as $25 a month with a four-user signup. The users need not be related, and the users can each pay their own monthly bill. Keep in mind that MVNOs almost always have fine print allowing "deprioritization" in busy areas and during busy times of the day. And the underlying carriers always screw their MVNO customers first.

Some smartphones support two SIMs while others have one physical SIM slot plus an eSIM registration that can be acquired from the major carriers. We personally liked an unlocked $200 Motorola G8+ which has two actual SIM slots as do most of the newer Samsung, OnePlus, Huawei, and many other smartphones. Several cellphones from BLU are available for under $25. The following devices currently support a single SIM card plus an eSIM:

Apple® iPhone® SE
Apple iPhone 11
Apple iPhone 11 Pro
Apple iPhone 11 Pro Max
Apple iPhone XR
Apple iPhone XS
Apple iPhone XS Max
Google Pixel 4 / Pixel 4 XL

Configuring the API Settings for GTI. Once you insert your SIM into your smartphone and power it on, you need to modify the API settings and choose GTI Global (Mobile-X) as your provider for the GTI SIM. Here is a FAQ covering the appropriate API settings depending on whether you’re using an iPhone or Android phone. Here’s another tip. Even though your phone may show a number of available carriers other than Mobile-X, that doesn’t mean you can connect to any of them directly. Why? Because you don’t have credentials to access those networks. Your connection to those other networks is only available through Mobile-X using your GTI credentials.

Testing Out Your GTI Smartphone

Once you complete all of the steps above, it’s time to try things out. From your cellphone, try dialing *98 to connect to Asterisk voicemail. Next, try calling another extension on your PBX. Then try calling an outside number. Finally, try calling the DID of your GTI SIM from another phone and be sure it rings and you have audio in both directions.

When you complete the testing, you can add another SIP phone to the same extension as your GTI cellphone. Be sure to set the port to the proper port for your PJsip extension. When you call your GTI SIM’s DID, both the smartphone and the SIP phone should ring simultaneously.

In some environments, you may wish to configure 611 calls to ring a support ring group in your organization. If so, add a new SIP extension 611 and forward calls to that extension to the number of the desired ring group. For example, if the Ring Group number is 777, you could enter the following command at the Asterisk CLI: database put CF 611 777.

Originally published: Monday, July 13, 2020

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

Special Thanks to Our Generous Sponsors

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

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

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

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

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

Introducing Cellular Extensions for Incredible PBX & Asterisk

Monday, June 22, 2020 / 3 Comments on Introducing Cellular Extensions for Incredible PBX & Asterisk

The Holy Grail for many of us in the VoIP community has been a cellphone that functions like a traditional Asterisk® extension on your PBX. The typical use case would be a real estate agent, service technician, salesperson, or other mobile worker who interacts with a home office as part of their daily routine. The beauty of this for the mobile workforce is it allows both the home office receptionist and the mobile worker to not only exchange calls but also to transfer the calls and retrieve voicemail just as if the worker were using a phone in the office.

Some of you may recall that we introduced a service like this called vMobile from Vitelity about six years ago. vMobile had some growing pains not the least of which was total reliance upon Sprint for cellular coverage. The service has been discontinued.

Today we’re testing a new offering from Global Technologies (GTI Global) on the Incredible PBX® platform that addresses some of the shortcomings of the vMobile offering. First, it provides a SIM card that can automatically utilize all four of the major carrier networks in the United States. Second, it offers a dedicated DID, a SIP trunk with free U.S. calling, and an Asterisk extension for every cellphone. And third, it takes advantage of PJsip which permits multiple phones to be connected to and ring on the same Asterisk extension.

GTI Global provided promotional consideration to help defray costs of our Asterisk projects.

Prerequisites. System integrators will acquire SIM cards through the Incredible PBX project. Pricing for the recommended unlimited calling, unlimited messaging, and 2GB monthly data plan is $25/month. A NoData plan is available for $20/month, and other higher data plans also are available. The SIM is compatible with any iPhone® or Android phone device. Users must provide their own cellphones which need not be jailbroken. Dual-SIM smartphones allow users to dedicate one line to the GTI Global SIM and another line for personal use. GTI Global currently is exploring support for eSIMs which would facilitate iPhone use.

To use the SIM-based phone, a public-facing Asterisk platform with a dedicated IP address is required, and it must support PJsip extensions. Incredible PBX 2020, Incredible PBX 16-15, and Incredible PBX 13-13 all have been tested. Other FreePBX® 13, 14, and 15 platforms should also suffice. Accompanying each SIM is a dedicated DID assigned to each cellphone. A matching PJsip extension must be created on the Asterisk platform, and the SIP credentials must be provided to GTI Global to make the cellphone connection to the PBX. Additional phones may be connected to the same PJsip extension to support a desktop phone or receptionist. Simply adjust the Max Connects entry when creating the PJsip extension to support the number of phones desired. Each phone on an extension requires a unique port.

How Calling Works. GTI Global has positioned servers in the facilities of every carrier. When a call arrives from a carrier’s tower, the carrier processes 911 calls directly through its network. For all other incoming calls, the carrier verifies the credentials of the SIM. Once verified, the call is passed to the GTI Global server which sends incoming calls to the associated extension on your PBX. The Inbound Route associated with that DID then sends the call to the assigned destination. If the GTI Global server is unavailable, the carrier processes the incoming calls just as it would any other call on its network by sending it directly to the cellphone.

When the cellphone user places a call, it is processed just as if the call had been made from an internal extension on the PBX with the exception of 911 calls which are handled directly by the carrier. After SIM verification, GTI Global passes the outgoing call to the PBX for processing using the Outbound Routing rules of the PBX. Internal extensions and voicemail can be dialed directly. Outbound calls can utilize any trunk associated with the PBX including the free U.S. trunk provided by GTI Global. CallerID is determined by the outbound trunk processing the call.

Configuring a GTI Global Extension. After logging into the FreePBX GUI, create a PJsip extension with the following settings:

Extension: 12345
Display Name: 12345 GTIglobal
Secret: yourSIPpassword

Advanced -> Max Contacts: 4

Configuring a GTI Global Trunk. Create a chan_SIP Trunk in the FreePBX GUI using your GTIglobal-provided credentials:

Trunk Name: trunk4321

PEER Details:

username=trunk4321
type=friend
secret=yourTRUNKpassword
host=xxxxxx.mvnoserver.com
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
port=5062

Register string: trunk4321:yourTRUNKpassword@xxxxxx.mvnoserver.com:5062

IMPORTANT: When you sign up with GTI, you need to give them the IP address of your PBX, the credentials (extension and password) for your GTI PJsip extension, the outbound SIP port for your GTI trunk (UDP 5062), and the inbound PJsip port for your PJsip extension (UDP 5061).

Configuring an Inbound Route for GTI Global Calls. Replacing 16785551212 with your actual 11-digit DID assigned to each cellphone, create Inbound Routes that look like this:

Description: GTIglobal 16785551212
DID Number: 16785551212
Destination: Extension -> 12345 (your extension associated with each SIM)

You need to add the following context at the end of /etc/asterisk/extensions_custom.conf using your DID. Then reload your dialplan: asterisk -rx "dialplan reload"

[from-sip-external]
; GTI Global
exten => 16785551212,3,Goto(from-trunk,${DID},1)

Configuring an Outbound Route for GTI Global Calls. No special outbound routes are required unless you wish to use the GTI Global trunk to process U.S. calls at no cost. In this case, configure an Outbound Route pointing to your GTI Global Trunk. NOTE: Calls using this trunk can only be made from the extensions associated with GTI Global SIMs.

CallerID and Trunking Strategy. In setting up the GTI components in FreePBX, you need to consider how you wish to process outbound calls. If you want all outbound calls including those from the GTI extensions to reflect the company’s main phone number, then you can use your default outbound route making certain that CallerID override is set with the company number in the appropriate trunks. No CallerID should be set in the outbound route.

If you want calls from the GTI extensions to use the GTI trunk with the CallerID number of the individual cellphones, then the GTI trunk needs to be moved to the top of the default route sequence with no CallerID number specified in the GTI trunk or the outbound route. Instead, set the correct 11-digit CallerID numbers in each of the GTI extensions. This will not impact outbound calls from other extensions because the GTI trunk will indicate congestion forcing the calls out through the next trunk specified in the outbound route.

GTI Global Firewall WhiteList. On Incredible PBX servers and other IPtables platforms, add the following whitelist entries to your firewall:

/usr/sbin/iptables -I INPUT -p udp -m udp -s 99.198.122.166 --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 99.198.110.51  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 96.127.174.38  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.170  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.171  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.172  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.173  --dport 5060:5069 -j ACCEPT

How to Proceed. If GTI Global cellular extensions are of interest for your customers, kindly contact us at support@incrediblepbx.com, and we’ll hook you up with the folks at GTI Global. There also is an extensive KnowledgeBase for those wanting more information.

Continue Reading: Last Chance to Jump onto Incredible PBX Cellular Bandwagon
Soup-to-Nuts Tutorial: Adding Cellular Extensions to Incredible PBX

Originally published: Monday, June 22, 2020

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

Special Thanks to Our Generous Sponsors

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

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

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

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

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

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