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Cyber Monday 2020: TV Deals You Can’t & Shouldn’t Refuse


We try to leave Black Friday and Cyber Monday deals to the big boys but, hey, it’s 2020 and some of these deals may never come along again. Most of these are offerings we’ve tried for ourselves which is different than regurgitating the flyers from Amazon, WalMart, and BestBuy.

Televisions. Who would have ever thought you’d be able to purchase a 75″ 4K Android TV with built-in Chromecast for under $500? TCL has you covered this year at Best Buy.

If you have a back porch that really could use a cord-cutting TV, here’s our favorite 43″ model which is on sale for $209.99. All you need is WiFi and a little electricity to bring this TV to life.

And, speaking of cord-cutting, we assume you already have your Netflix subscription in hand. But now’s the time to add Hulu which is available for $1.99 a month if you hurry. Better yet, if you still have a kid in college, sign them up for a Spotify Student account at $5 a month and both a Hulu and ShowTime subscription are thrown in at no extra cost. Don’t worry about stepping on your kid’s use of the services. They sleep at completely different hours than you do. With the new Android TVs, adding these services is as easy as adding an Android app on your smartphone. Add a Sling TV subscription for $30 a month, and you’ll have more stations than you could ever watch.

DVR Delight. One of the complaints we often hear from the cord-cutters is the absence of a decent digital video recorder for these streaming services. And, of course, none of them let you skip the endless ads in 4+ minute clusters sprinkled throughout your shows and movies. Well, now there’s a solution. If you haven’t tried PlayOn Cloud, you’ve missed the greatest advancement since the Betamax. Here’s how it works. You pay PlayOn a fee to make a cloud-based, ad-skipping recording of your movie or TV show using your credentials for as little as 10¢ for a full-length movie or TV show. Then you get a week to download the MP4 recording at no additional cost. Or you also can sign up for a PlayOn cloud-based storage plan, and they keep your recordings in the cloud for as long as you need them and they remain in business. PlayOn recording, playback, and content management software is free.

For Black Friday through Cyber Monday, you can purchase credits to record 1,000 shows in the cloud for just $100. That’s 10¢ per show or movie, regardless of length. The recordings then can be downloaded to a PC, Mac, or Smartphone at no additional cost for one week from the date of the recording. If you purchase long-term cloud storage, it’s $4.99 a month for 100GB of storage with 20 recording credits every month. $19.99 a month gets you a terabyte of cloud storage plus 40 recording credits each month. You can record shows or movies from any of the channels shown below assuming you have the necessary accounts in place. Is it legal? While this isn’t legal advice, we would label this a close case and would be comfortable representing either side. You can read the history of the Betamax case for yourself and decide whether the time-shifting argument trumps the license agreements of all the streaming services. For the time being, you’re probably okay, but nothing lasts forever… except your downloaded MP4’s and a media player or Roku device. 😉


 

Originally published: Sunday, November 29, 2020



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



SPAM Blocker & CNAM Cornucopia for Incredible PBX 2020




If you enjoy calls from politicians and car warranty offers as much as we do, then today’s your lucky day. Blocking spam phone calls has been a challenge to put it charitably. Thanks to some earlier work by Stewart Nelson on the DSLR forum as well as Stewart’s considerable hand-holding in the development of our previous tutorials, we want to introduce a refined call screening approach to block these calls. The way it works is first time callers that are not on your WhiteList will be prompted to "press 5 to connect." Since most spam calls sit in a queue for several seconds before a live person chimes in, that person won’t hear the prompt. After 10 seconds or an invalid response, the call is sent to voicemail. In the alternative, you can play a SIT tone and disconnect the call or you can send the call to your favorite uncle, Lenny. When a successful caller calls again, the caller will be connected without a prompt.1

While today’s approach won’t block every robocaller, our testing suggests that it will catch more than 95% of these annoying calls. Using CallerID Superfecta coupled with the Asterisk® Phanebook will provide an extremely low-cost solution both for blocking spammers AND for displaying accurate CNAM data for incoming calls. The silver lining is you’ll only pay for CNAM lookups from legitimate callers once, and you have a choice of using OpenCNAM or BulkCNAM with the scripts we’ll provide today. Last, but not least, you’ll also get CNAM data for outgoing calls in your CDR logs.

Here’s the actual dialplan addition that will monitor your incoming calls:

[sub-log-caller]
exten => s,1,NoOp(*** begin sub-log-caller ***)
exten => s,n,GotoIf(${DB_EXISTS(cidname/${CALLERID(num)})}?CNAMOK)
exten => s,n,Playback(silence/1)
exten => s,n,Playback(to-call-num-press)
exten => s,n,Playback(digits/5)
exten => s,n,Read(MYCODE,beep,1,n,1,10)
exten => s,n,GotoIf($["${MYCODE}" = "5"]?ANONTEST)
exten => s,n(FLUNKED),NoOp(*** Caller FLUNKED screening ***)
;exten => s,n,Dial(local/*701@from-internal) ; uncomment to send to 701 VM
exten => s,n,Dial(local/53669@from-internal) ; uncomment to send to Lenny
exten => s,n,Zapateller()
exten => s,n,Hangup
exten => s,n,Return()
exten => s,n(CNAMOK),Set(CALLERID(name)=${DB(cidname/${CALLERID(number)})})
exten => s,n,Goto(WHITELISTED)
exten => s,n(ANONTEST),GotoIf($[${CALLERID(num)} > 0]?WHITELISTNOW:CONNECTNOW) 
exten => s,n(WHITELISTNOW),Set(DB(cidname/${CALLERID(number)})=${CALLERID(name)})
exten => s,n,Set(CALLERID(all)="${CALLERID(name)} <${CALLERID(number)}>")
exten => s,n,Goto(SENDEMAIL)
exten => s,n(WHITELISTED),Set(CALLERID(all)="${CALLERID(name)} <${CALLERID(number)}>")
exten => s,n,Goto(CONNECTNOW)
exten => s,n(SENDEMAIL),NoOp(WhiteListed: ${CALLERID(all)})
exten => s,n,Set(email="root")
exten => s,n,GotoIf($[${email} = "root"]?CONNECTNOW)
exten => s,n,system(echo "In Asterisk Phone Book, verify new CNAM entry of ${CALLERID(name)} for ${CALLERID(number)}." | /usr/bin/mail -s "Incredible PBX CNAM Reminder" ${email})
exten => s,n(CONNECTNOW),NoOp(*** end of sub-log-caller ***)
exten => s,n,Return()
;-------------------------------------------------------------------------------

 
The beauty of today’s design is that it won’t interfere with your existing call processing rules. In other words, FreePBX® Inbound Routes sent to IVRs, Ring Groups, Conferences, and even incoming Faxes still will be processed exactly as they have been in the past once the CallerID number makes it onto your WhiteList. In order to preserve all of this existing functionality, we’ve tweaked the FreePBX Core module slightly, but it will be transparent going forward. The only caveat is that whenever you upgrade the Core module, you’ll always need to run the /root/sig-fix script to reimplement this new call screening process and to disable Module Signature Checking which has always been an integral part of Incredible PBX.

We first introduced some of these concepts in our previous article in 2018. That article also documented the procedure for adding inbound call processing logic into FreePBX. If you already have implemented the steps outlined in that article, then today’s installer will back up your prior version and overwrite it with this new, improved release.

Here are the basic steps to get this working:

  1. Configuring CallerID Superfecta in FreePBX
  2. Enabling CallerID Superfecta on Inbound Routes
  3. Setting the Proper Context for Your Trunks
  4. Downloading & Installing Call Screener
  5. Importing Previous Callers into WhiteList

1. Configuring CallerID Superfecta in FreePBX

CallerID Superfecta is an integral component in today’s new call screening design. It will be used both to populate the Asterisk Phonebook’s WhiteList and to provide CallerID Name (CNAM) data about your callers while assuring that you only pay for one CNAM query even though grandma may call you a dozen times a day. We use the Asterisk Phonebook as the whitelist of authorized callers. The way CallerID Superfecta works is it checks multiple sources for a match on the incoming CallerID Number. As soon as a match is found, the checking ends and the CallerID Number and Name are passed to our Call Screening script.

The CallerID Superfecta lookup sequence needs to be set as follows in the United States: AsteriDex, Asterisk Phonebook, and then one of the following commercial CNAM lookup services: OpenCNAM or BulkCNAM. In other countries, there still may be free CNAM services, but they’ve all disappeared in the U.S. market. We’ve documented the other available sources in a previous Nerd Vittles article.

Low-volume OpenCNAM Value pricing provides global lookups for $0.0028 each. BulkCNAM provides CNAM queries with RoboCall identification for $0.002 per query. If you sign up with OpenCNAM, you will need your Account SID and Auth Token to configure CallerID Superfecta and to populate our Call Screening script. If you sign up with BulkVS, you will need your API Key from the API Credentials tab in your BulkVS Dashboard.

With your credentials in hand, login into FreePBX as admin and navigate to Admin -> CID Superfecta -> Default. Arrange and enable the lookup sources in the following order: AsteriDex, Asterisk Phonebook, and then either OpenCNAM or BulkCNAM (in the U.S. market) or your country’s best CNAM lookup source. Be sure to enter your credentials for the CNAM provider by clicking on the wrench icon beside the provider. If your incoming trunks already provide CNAM lookups (such as BulkVS and Incredible PBX Trunking), then you can substitute Trunk Provided as your CNAM lookup service. With Incredible PBX Trunking, in addition to free CNAM lookups, you also get SPAM detection at no additional cost. For details on the service, follow this link. Then we typically set Telco Data as the last lookup source which will at least give you the city and state of the caller.

2. Enabling CallerID Superfecta on Inbound Routes

By default, CallerID Superfecta is not enabled for incoming calls to your PBX. You must enable it on every Inbound Route by navigating to Connectivity -> Inbound Routes and then editing each of your routes. Then click on the Other tab and set Enable Superfecta Lookup to YES and set the Superfecta Scheme to DEFAULT. Click SUBMIT to save your route settings and then reload the dialplan when prompted.

3. Setting the Proper Context for Your Trunks

It’s equally important to make certain that the CallerID Numbers for all of your incoming calls arrive in the same format. Computers are stupid. 8005551212 and 18005551212 and +18005551212 are completely different callers as far as your PBX is concerned. If different trunks deliver calls with CallerID Numbers formatted differently, then you would need to whitelist ALL of the various permutations for every caller in the Asterisk Phonebook. For those in the U.S. and Canada that primarily receive calls from the U.S. and Canada, we recommend setting the context entry in every trunk to from-pstn-e164-us. This will handle the translation of all 3 number formats above into 10-digit numbers. Calls from other countries will not be affected.

4. Downloading & Installing Call Screener

Now let’s put all the Call Screener components in place and configure the screening setup to meet your own requirements. To get started, log into Incredible PBX as root and issue the following commands:

mkdir /tmp/CALL-SCREENER
cd /tmp/CALL-SCREENER
wget http://incrediblepbx.com/CallScreener.tar.gz
tar zxvf CallScreener.tar.gz
rm -f CallScreener.tar.gz
./install

Once the install is begun, the editor will open to the dialplan code. In the [sub-log-caller] context, you have a few options. First, you need to choose how to handle incoming calls where the caller does not enter the "press 5″ number prompt in a timely manner. The default setup (line 9) sends these callers to voicemail for extension 701. You can change the voicemail extension, or you can elect to treat the calls differently. We’ve provided two additional options. Line 10 will send the calls to Lenny at extension 53669. Line 11 will send the calls to Zapateller which is the universal tone for numbers that are not in service. You should enable only one of these three options and comment out the other two by placing a semicolon (;) at the start of the other two lines. If you have fax detection enabled on your PBX, you probably would not want to send failed calls to either Lenny or Zapateller since you may never know the incoming faxes failed. Similarly, if you get calls from people with rotary dial phones such as Grandma, you probably don’t want her talking to Lenny or listening to Zapateller tones.

The next option is which number to prompt callers to press. The default is 5. But you can change it by modifying the existing 5 entry on BOTH lines 5 and 7.

The final option in the [sub-log-caller] context is to activate email notifications for new callers that pass the screening test. This is especially important if you receive lots of calls from cellphone users. Most of those calls will arrive with a CNAM entry of nothing more than the caller’s City and State. Activating an email reminder will notify you to update the Asterisk Phonebook entry for such callers to replace the City, State entry with the caller’s actual name so that your CDR listings and future calls provide accurate CNAM information for the caller. To activate email reminders, replace root in Set(email="root") line with your actual email.

The [macro-dialout-trunk-predial-hook] context handles populating the Asterisk Phonebook WhiteList for outbound calls you make to people that are not yet in your Asterisk Phonebook. These numbers will automatically be added to your whitelist, but you also have the option of adding CNAM entries for these outbound calls using either OpenCNAM or BulkCNAM for outbound calls to numbers that are not yet in your Asterisk Phonebook. To activate CNAM lookups, simply uncomment either line 4 or 5 in the context. For the service you have activated, remember to also enter your Account SID and Auth Token in the case of OpenCNAM or your API Key in the case of BulkCNAM. If you leave both lines commented out which is the default, the callee’s phone number will be entered as both the CNAM and CNUM entry in the Asterisk Phonebook.

Once you have made all the changes desired, save the template by pressing Ctl-X, then Y, then ENTER. The installer then will complete installation of the Call Screener components.

5. Importing Previous Callers into WhiteList

We appreciate that you may not want to aggravate callers that have been calling you for years by making them jump through hoops the next time they call. So here’s a quick way to populate your Asterisk Phonebook with the names and numbers of previous callers. For entries where the CNAM is merely the CallerID Number, future calls from these numbers still will be looked up with OpenCNAM or BulkCNAM to obtain an actual CNAM match. We’ve made a couple of assumptions that you are more than welcome to adjust to meet your own needs. First, we’ve limited the list to callers from the past two calendar years. Second, we’ve only captured calls that lasted more than 15 seconds. We’ll drop down to the Linux CLI to build the list of callers to import. Then we’ll use the FreePBX GUI to import the list into the Asterisk Phonebook. While building the import list, you’ll have an opportunity to prune the list and remove any undesirable entries using nano. To generate the .csv file, issue the following commands:

cd /root
./export-CDR

Now you should have a 2YR-clean.csv file in its final form for import. Copy the file to your desktop PC and open FreePBX in your browser. Navigate to Admin -> Asterisk Phonebook. Click Import Phonebook and then Browse. Select the 2YR-clean.csv file from your desktop. Then click Upload. Take a final look at the new entries in your Asterisk Phonebook to make sure nothing came unglued, and you’re all set.
 

Originally published: Monday, September 14, 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.
 




 

  1. Once installed, you can change the voice prompt to a number other than 5 by modifying lines 10 and 12 of the context sub-log-caller which you will find in extensions_custom.conf in the /etc/asterisk directory at the completion of this install. []

Frozen in Time: Sangoma at a Crossroads or the Cliff

We are fast approaching the 18 month anniversary since the mass exodus of talent from Sangoma and Digium. During that time, Asterisk® and FreePBX® have languished in limbo with FreePBX commercial modules mired in discontinued encryption methods and the Asterisk and FreePBX feature set frozen. Sangoma’s acquisition of Digium gave the company a well-respected business phone system in Switchvox® while relegating FreePBX deployments with commercial modules to second-class citizen status.

Now Sangoma has a difficult and expensive decision to make. In order to continue to deploy FreePBX commercial modules on a current operating system and PHP platform, they are facing a multi-million dollar investment to retool all of the commercial modules using an entirely different encryption platform which will generate additional headaches with their module repositories which now must manage and maintain two different sets of modules. Think of it as migrating from an English-based operation to one in which the customer base speaks either English or French, but not both. In the meantime, Asterisk 17 is fast approaching end-of-life while Asterisk 16 is getting long in the tooth with only minor fixes of late. And there’s FreePBX 15 which hasn’t seen significant upgrades since the departure of the former Schmoozecom staff in April of 2019.

While we have never agreed with the Sangoma CEO about much of anything, let’s take over the management reins for a moment and look at the landscape. The Digium acquisition coupled with some other purchases has left Sangoma with a mountain of debt in the middle of a global pandemic. Coronavirus also has forced many businesses to reassess their long-term office strategy. With nearly everyone on the planet now carrying a smartphone and most current employees having high-speed Internet connectivity at home neither of which the company has to pay for, does a fancy office for every employee still make good business sense? When you have a well-positioned business phone offering in Switchvox, does it make good business sense to continue to pump boatloads of money into a competing FreePBX commercial product simply to preserve its functionality? At the same time, Sangoma’s open source Asterisk and FreePBX products need a serious shot in the arm to remain viable. For example, FreePBX 15 with its User Control Panel won’t run on CentOS 8, and CentOS 7 reaches EOL status in December.

So here’s our first act as Sangoma CEO. Convert all of the existing FreePBX commercial modules into open source, GPL code while offering existing FreePBX commercial module customers a 50% discount to migrate to Switchvox within the next six months. We would stagger conversion of the commercial modules to GPL code by releasing three on the first day of every month. This would keep Sangoma in the news for many months to come. The move would not only restore Asterisk and FreePBX to their well-deserved position at the forefront of the open source community, but it would also give Switchvox a leg up in the commercial phone system marketplace. By keeping the identity of the monthly releases a secret, it would build much needed momentum for Sangoma’s open source offerings while reinforcing Sangoma’s commitment to the open source community. Give it some thought, Sangoma!
 

Originally published: Tuesday, September 8, 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.
 



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.
 



Harnessing the Cloud to Start An Incredible PBX Business




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.
 



An Incredible PBX Cloud Platform That’s Second to None




With mixed results, we’ve invested a lot of time scouring the Internet for reasonably priced cloud platform solutions for Incredible PBX®. The fatal flaw with many of the low-cost providers was bankruptcy. But today we have a winning solution that we’ve used for many years with excellent results. Vultr’s latest offer of a 30-day $100 credit means you can test out the performance levels of various Vultr instances to determine which best meets your needs. You can also try out their automatic backup service. In addition to providing a terrific platform for your VoIP server, this latest Vultr deal also provides financial assistance to Nerd Vittles to assist with future development of the Incredible PBX project. When the dust settles for most users, we think you’ll find their $5/month plan is more than adequate to host Incredible PBX and another $1/month buys you automatic backups. Setup takes 30 minutes.

To get started, follow our Vultr referral link and sign up for a new account with your email address. Then deploy a new Cloud Compute instance in your choice of the locations shown above. Next choose 64-bit CentOS 7 as your Server Type. Then select the $5/month 25GB Server Size with 1GB RAM and 1TB monthly bandwidth. Choose a Server Hostname and click Deploy Now. CAUTION: Make sure you select CentOS 7 and not the default CentOS 8 platform!



Penny for Your Thoughts. One of the real beauties of Vultr is you pay by the hour for your instances. It only takes a minute or so to delete an instance and create a new one. If you’re new to Incredible PBX and VoIP servers, this makes it easy to experiment with the setup process until you are happy with the results. And the setup cost for a functioning Incredible PBX platform is about a penny. You’ll be hard-pressed to use up your $100 credit.

Installing Incredible PBX. Once your instance has been created, open an SSH or Putty session on your desktop PC and login to the IP address of your instance as root. You can copy-and-paste your default password from the Vultr dashboard. When you’ve successfully logged in, immediately change your default root password and follow the steps below to install Incredible PBX. It takes about 30 minutes to complete the process.

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
# add a swap file for your instance
./create-swapfile-DO
# kick off Phase I install
./IncrediblePBX2020.sh
# after reboot, kick off Phase II install
./IncrediblePBX2020.sh
# set desired timezone
./timezone-setup
# set FreePBX admin password
./admin-pw-change
# set Apache admin password for AsteriDex and Reminders
./apache-pw-change
 # display your passwords
./show-passwords
# optionally install Incredible Fax 2020
./incrediblefax2020.sh
# set up NeoRouter VPN client, if desired
nrclientcmd
# check network speed and smile
wget -O speedtest-cli https://raw.githubusercontent.com/sivel/speedtest-cli/master/speedtest.py
chmod +x speedtest-cli
./speedtest-cli

Getting Started with Incredible PBX. Before you can actually make and receive calls using Incredible PBX, you’ll need to sign up with a SIP provider and configure a trunk, define how your PBX should route incoming and outgoing calls, and finally install and configure at least one softphone or SIP phone. All of this is accomplished using the FreePBX® GUI and your favorite web browser. Begin by pointing your browser to the IP address of your server. Login as admin using the password you configured with admin-pw-change above.

The way a PBX works is straight-forward. For incoming calls, the PBX identifies the Trunk delivering the incoming call, looks up the Inbound Route for that Trunk, and delivers the call to a destination specified in that route. This could be a phone registered to an Extension on your PBX, or it could be a group of phones known as a Ring Group, or it could be some other destination such as an IVR or AutoAttendant which answers the incoming call and provides the caller with a choice of options that define how the incoming call should be routed.

For outgoing calls, someone with a phone registered to the PBX picks up the phone and dials a number. The PBX looks through its defined Outbound Routes to identify how calls matching that Dial String should be handled. For example, 3-digit and 4-digit calls might be routed to an internal extension, 7-digit calls might have an area code automatically added as a prefix before sending the call to a specified Trunk provider, 10-digit calls might be routed to a different Trunk provider, and calls beginning with 01144 might be routed to yet another Trunk provider that offers inexpensive calling rates to the United Kingdom. Unlike traditional Ma Bell phone service, with VoIP calling, there is rarely a penalty for deploying multiple Trunk providers since most of the services offered are on a pay-as-you-go basis. If you don’t make any calls on a particular trunk in a given month, you don’t get a bill even though most providers require at least a minimal deposit to cover the costs of any calls you actually make.

Choosing a SIP Provider. Incredible PBX 2020 comes preconfigured with support for five SIP extensions and five of the major SIP providers: Skyetel, VoIP.ms, V1VoIP, Anveo Direct, and ClearlyIP. Both Skyetel and ClearlyIP financially support Nerd Vittles and our open source projects, and both offer clearly superior platforms with crystal-clear communications and redundancy so you never miss a call. Both also set themselves apart from the other providers in the support department. They actually respond to issues, and there’s never a charge. As the old saying goes, they may not be the cheapest, but you get what you pay for. Even without taking advantage of Nerd Vittles half-price offer on up to $500 of Skyetel services, both are still dirt cheap compared to the Bell Sisters and cable companies. Skyetel is so sure you’ll love their service that they give you a $10 credit to kick the tires before you ever spend a dime. Traditional DIDs are $1 per month. Outbound conversational calls are $0.012 per minute. Incoming conversational calls are a penny a minute, and CallerID lookups are $0.004. You only pay for minutes you use. Once you’re satisfied with the service and fund your account, you can port in your existing DIDs at no cost for 60 days after signup with Skyetel.

Configuring Skyetel for Incredible PBX 2020

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 use SIP registrations to make connections to your PBX. Instead, Skyetel utilizes Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. Here’s a typical setup for Incredible PBX 2020:

  • 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.

Configuring ClearlyIP for Incredible PBX 2020

ClealyIP Trunking is perhaps the easiest to configure because the entire setup including registration is managed in the FreePBX GUI. Choose Connectivity -> Clearly Trunking -> Sign Up. Next click Login/Signup then Home then Start Your Free Trial to get started.

ClearlyIP Trunking provides E911 features that enable your organization to be Kari’s Law and Ray Baum Act compliant out of the box. You also have a choice of buckets of minutes providing expandable call paths or pay-by-the-minute service or a combination of the two. Unlike others, CNAM lookups and E911 support are included in the price of all their services.

Configuring VoIP.ms for Incredible PBX 2020

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

Configuring V1VoIP for Incredible PBX 2020

To sign up for V1VoIP service, sign up on their web site. Then login to your account and order a DID under the DIDs tab. Once the DID has been assigned, choose View DIDs and click on the Forwarding button beside your DID. For Option #1, choose Forward to IP Address/PBX. For the Forwarding Address, enter the public IP address of your server. For the T/O (timeout) value, set it to 2o seconds. Then click the Update button. Under the Termination tab, create a new Endpoint with the public IP address of your server so that you can place outbound calls through V1VoIP. In the Incredible PBX GUI, be sure to enable all of the V1VoIP trunks.

Configuring Anveo Direct for Incredible PBX 2020

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 Softphone for Incredible PBX 2020

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

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

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

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

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

Audio Issues with Incredible PBX 2020

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

Configuring SendMail with Incredible PBX 2020

In order to receive voicemails by email delivery, outbound mail functionality from your server obviously is required. If you’ve deployed your server in your home, your Internet Service Provider probably blocks downstream mail servers such as Incredible PBX from sending mail. This is done to reduce SPAM. In this case, you will need to configure SendMail using either your ISP or Gmail as an SMTP Relay Host. 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. Here are the steps using a Gmail account:

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

At Vultr, you may need to open a ticket and request that TCP port 25 be unblocked. Then you can send mail without using a SmartHost; however, your server's hostname must actually be real or downstream mail servers will reject your mail. You can set your server's hostname like this: hostname noreply.incrediblepbx.com. This is usually sufficient; however, it's a good idea to also add the hostname in /etc/hostname and in /etc/hosts as the first entry on 127.0.0.1 line:

127.0.0.1   noreply.incrediblepbx.com pbx.local localhost

Next, test outbound mail using this command with your actual email address:

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

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



Getting Started with Incredible Fax 2020

Believe it or not, there still are lots of folks that use faxes in their everyday lives. If you're one of them, Incredible PBX has your back. Begin by logging into your server as root and running incrediblefax2020.sh to install HylaFax and AvantFax on your server. You'll be prompted a dozen or more times for information. Answer no to the secure fax question. For the rest of the prompts, just press ENTER to accept the default entries. Rebooting your server is required when the install finishes. Once your server is back on line, there will be a new AvantFax tab in the GUI. Before proceeding, be sure to set an Apache web apps password by running /root/apache-pw-change. Next, login to AvantFax with your browser. You first will be prompted for your Apache credentials. Enter admin for the username and whatever password you set up in the previous step. Then you will be prompted for your AvantFax credentials. The default is admin:password. After you enter the username and password, you will be prompted to change your admin password. The old password is still password. Then enter your desired password twice and save the setting. The AvantFax dashboard then will display. If nothing has come unglued, you should see four green Idle icons:



You can Send Faxes from within AvantFax by choosing the Send Fax tab, or you can use one of many HylaFax clients. Google is your friend.

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.

HINT: An Incredible PBX server on Vultr makes an ideal platform for your OpenVPN server.

Originally published: Monday, March 2, 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.
 



Going Public with Incredible PBX 16 and VitalPBX 2.3.8



As part of our ongoing development efforts, we maintain about a dozen honeypot servers across the U.S. and Canada to monitor the latest adventures of the bad guys. Security becomes especially important for those wishing to live on the bleeding edge and deploy a cloud-based, public-facing VoIP server. Today we want to walk you through our latest suggestions to set up and secure a VitalPBX platform using just the built-in FirewallD, IPset, and Fail2Ban components. If you opt to deploy VitalPBX in the Cloud, a KVM-based VPS is absolutely essential in order to take advantage of the security mechanisms we will introduce today.

Here are 6 Key Security Features in today’s public design:

  • SIP Registration Lockdown by FQDN
  • Extension Lockdown by IP Address
  • Trunk Provider Lockdown by IP Address
  • Web Access Lockdown by WhiteList
  • Disguised Ports for SIP and SSH Access
  • 100,000+ VOIP Blacklist for FirewallD

Is it 100% safe? Nothing ever is. That’s what backups are for. 😉

FYI: The CentOS folks reintroduced a previous FirewallD bug on October 22 which (again) broke new VitalPBX installs. On October 23, the VitalPBX developers fixed the bug (again). There should be no problems with new installs. For previous installs, see this thread on the PIAF Forum for the fix.

Taking Incredible PBX with VitalPBX to the Cloud

Because Incredible PBX with VitalPBX 2.3.8 was originally distributed as an ISO, getting it installed in the cloud was a challenge. A few cloud providers let you bring your own ISO to install on their VPS platforms, but it was still a tedious process. So today we’re pleased to introduce a new install script that can be run on any CentOS 7 platform.

We have a few cloud providers that we recommend without reservation. Both Vultr and Digital Ocean provide referral credits to Nerd Vittles to support our VoIP project development efforts. We’ve used both of them for many years with no problems. Either of the platforms works well using the $5 a month option in your choice of cities. Just be sure to choose the CentOS 7 platform, not CentOS 8. For an extra buck, you can add automatic backups.

Our favorite bargain is now CrownCloud in Los Angeles. For $25 a year, they offer a KVM VPS that is ideal as a VoIP platform. And the offering includes a free snapshot image as well. As you might imagine, it’s very popular and goes Out of Stock from time to time so check back often. For our international friends, CrownCloud offers similar platforms at the same price point in both Germany and the Netherlands.

Installing Incredible PBX with VitalPBX on CentOS 7

Once your CentOS 7 platform is up and running, here’s how to install Incredible PBX for VitalPBX. Log into your server as root using SSH or Putty. Then issue these commands:

cd /root
passwd
yum -y install net-tools wget nano tar
wget http://incrediblepbx.com/incrediblepbx.sh
chmod +x incrediblepbx.sh
./incrediblepbx.sh

Incredible PBX Cloud Setup Recipe for VitalPBX

We think the easiest way to configure your new VitalPBX platform is to follow the simple steps outlined below. This will avoid your having to jump back and forth between tutorials to get all the pieces in place. When you’re finished, you’ll have a secure VitalPBX cloud platform. Don’t be intimidated by the number of steps. If you can handle slice-and-bake cookies, you can do this!

1. Point your browser to the IP address of your server. You’ll be prompted to set a password for admin access to the GUI. Fill in the blanks to proceed. Should you ever forget your admin password, here’s how as root user to force a reset on your next login from a browser:

mysql ombutel -e 'update ombu_settings set value = "yes" where name = "reset_pwd"'


2. Register your server when prompted. The VitalPBX Dashboard will appear.

3. Decipher the public IP address of your desktop machine and any other PCs that will be used to manage your server.

4. From the VitalPBX Dashboard, navigate to Admin:Security:Firewall:WhiteList. Enter each of your IP addresses from step #3 and click Save button.

5. From the VitalPBX Dashboard, navigate to Admin:Security:Intrusion Detection:WhiteList. Enter each of your IP addresses from step #3 and click Save button.

6. Modify the default SSH port by logging in to your server as root and issuing the following commands using the year you were born in the first line replacing 2000:

sed -i 's|#Port 22|Port 2000|' /etc/ssh/sshd_config
systemctl restart sshd

 
7. From the VitalPBX Dashboard, navigate to Admin:Security:Firewall:Services. Change the SIP port to 5080 or some other port number not in the 5060-5065 range. Change the SSH port to a 4-digit number matching the year you were born. Click Save button. Monitor your SSH log for attempted breaches and change your port if necessary:

cat /var/log/secure | grep password

 
8. Verify that you can log back into your server with SSH using the new SSH port number you assigned in step #6: ssh -p 2000 root@server-IP-address

9. From the VitalPBX Dashboard, navigate to Admin:Security:Firewall:Rules. Delete the HTTP and HTTPS items by clicking the Trash icon beside each entry. In the GENERAL tab, set Block ICMP Requests to YES. Click Save button. This blocks web access to everyone except those you’ve whitelisted in step #4 above. If you ever lock yourself out of web access, login to your server as shown in step #8 and temporarily whitelist the public IP address desired. This gets removed automatically the next time you save your Firewall settings from within the VitalPBX GUI.

iptables -A vpbx_white_list -s 12.34.56.78 -j ACCEPT

10. Before we get too far along, let’s put another layer of security in place for your new server. We’re going to add the VoIP Blacklist which blocks about 100,000 bad guys from around the globe. We’ll also add a cron job to update the blacklist every night. Log back into your server as root and issue these commands to put the pieces in place and enable the VoIP Blacklist.

TIP: The cron job below is scheduled to run at 20 minutes after 3 a.m. Change the time to something else so we don’t all bombard the VoIP Blacklist site for downloads at exactly the same time every night.

cd /etc
wget http://incrediblepbx.com/voipbl-firewalld.tar.gz
tar zxvf voipbl-firewalld.tar.gz
rm -f voipbl-firewalld.tar.gz
echo "20 3 * * * root /etc/update-voipbl.sh >/dev/null 2>&1" >> /etc/crontab
/etc/update-voipbl.sh

11. From the VitalPBX Dashboard, navigate to Admin:Add-Ons:Add-Ons. Click Check Online button. Click Install button beside Custom Contexts. Click Install button beside Phonebooks. Click Install button beside Domotic.

12. From the VitalPBX Dashboard, navigate to Settings:Tech Settings:SIP Settings.

  a. In the GENERAL tab, set the Bind Address port to 5080 or whatever port you chose in step #7 above. This is the port number together with the FQDN of your PBX (set in the next step) that any SIP phone will need to successfully register to an extension.

  b. In the SECURITY tab, set Allow Guest to NO, set Auto-Domain to NO, set Allow External Domains to NO, and enter a fully-qualified domain name (FQDN) pointing to the IP address of your server in the Domain field. We cannot stress enough how important this FQDN is to the security of your cloud-based server. It limits SIP registrations to this FQDN only, and all SIP registration attempts by IP address are automatically blocked. Don’t skip this step!

  c. In the NETWORK tab, enter the IP address of your server in External Address. Click the ADD button in the Local Networks section and enter the private IP addresses associated with your LAN and VPN, e.g. 192.168.0.0/255.255.0.0 and 10.0.0.0/255.240.0.0. Change NAT to Force,Comedia if your server is behind a NAT-based router.

  d. In the CODECS tab, enable ULAW, ALAW, G722, and G729.

  e. In the OTHERS tab, set SRV LOOKUPS to Yes. Click SAVE button.

13. From the VitalPBX Dashboard, navigate to Settings:Tech Settings:Profiles. Click Show All Profiles bar and choose Default PJSIP Profile. In the GENERAL tab, set the following entries to YES: Force rport, Rewrite Contact, Direct Media, RTP Symmetric, and Send Diversion Header. Click UPDATE button.

14. From the VitalPBX Dashboard, navigate to PBX:Applications:Parking. Click Show All Parking Profiles bar and choose Default. Change Code from 700 to 7000 and click Update button. This changes your Parking Lot extensions to the 7000 range so that 700 range can be used for Extensions, just like other versions of Incredible PBX.

15. Log out of your Dashboard and then log back in so that the menus get refreshed with the Custom Contexts addition.

16. From the VitalPBX Dashboard, navigate to PBX:Applications:Custom Contexts. Create the new sample IVR context with the following entries. Then click Save button.

  • Description: IncrediblePBX
  • Context: incrediblepbx
  • Extension: s
  • Priority: 1
  • Destination: Terminate Call -> Hangup

17. From the VitalPBX Dashboard, navigate to PBX:Applications:Custom Applications. Create the custom application for the sample IVR and Save it.

  • Code: 3366
  • Name: DEMO
  • Enabled: YES
  • Destination: Custom Contexts -> IncrediblePBX

18. From the VitalPBX Dashboard, navigate to PBX:Applications:Conferences. Create the new sample conference application and Save it.

  • Code: 2663
  • Description: CONF
  • Music on Hold When Empty: YES
  • User Count: YES
  • Announce Join/Leave: YES
  • Announce Only User: YES
  • User PIN: 1234
  • Leader PIN: 4321
  • Drop Silence: YES

19. If you didn’t read last week’s article on Custom Contexts, now would be a good time to do so. Here are the commands to put all those pieces in place on your new cloud-based server:

cd /
yum -y install dialog wget nano tar mailx
cp -p /etc/crontab /etc/crontab.bak
wget http://incrediblepbx.com/incrediblepbx-vitalpbx.tar.gz
tar zxvf incrediblepbx-vitalpbx.tar.gz
rm -f incrediblepbx-vitalpbx.tar.gz
chown asterisk:asterisk /var/lib/asterisk
cd /etc/asterisk/ombutel
echo "[cos-all-custom](+)" >> extensions__80-custom.conf
echo "exten => 412,1,NoOp(Voice Dialer)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,1,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 951,1,NoOp(News)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,5,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 947,1,NoOp(Weather by ZIP)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,6,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 3172,1,NoOp(DISA Voice Dialer)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,9,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 4747,1,NoOp(Wolfram Alpha)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,3,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 8463,1,NoOp(Time of Day)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,*,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 53669,1,NoOp(Lenny)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,53669,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 86329,1,NoOp(Today in History)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,7,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
systemctl restart asterisk
chown asterisk:asterisk /var/lib/asterisk
chown asterisk:apache /var/lib/asterisk/agi-bin

20. Create new Extensions for your PBX by navigating to PBX:Extensions:Extensions. You only need to fill in the Extension, Name, and Email Address fields. We recommend extension numbers beginning with 701. If the extension will be used from a phone behind a NAT-based router, change the NAT entry to Force,Comedia. If the phone associated with the extension has a static IP address, enter it in the Permit field for an extra layer of security. In the VOICEMAIL tab, you will note that voicemail is enabled by default with a password matching the extension number. This forces the user to set the voicemail password the first time they access voicemail with their phone. We recommend the YES setting for Attach Voicemail, Ask Password, Say CID, Say Duration, and Envelope. Then press SAVE.

21. Once you have created your extensions, you can create Ring Groups to assign multiple extensions and external numbers to a designated number which will ring all of the extensions and external numbers in the ring group either simultaneously or serially. Navigate to PBX:Call Center:Ring Groups to set this up.

22. Trunk Setup. While we don’t recommend it, if you just want to play around with some toll-free calls using option 1 in the DEMO IVR to see how everything works, here’s a simple trunk setup to get you started. First, navigate to Settings:Telephony:Channel Groups and save a group named Default with no entries. Then navigate to PBX:External:Trunks:CUSTOM. Create TollFree trunk with this Dial String: SIP/1${EXTEN}@ovh.starcompartners.com. No other entries are required. Click SAVE and reload your dialplan. Finally, create an Outbound Route for these calls in PBX:External:Outbound Routes like this:

  • Description: TollFree
  • Trunks: TollFree
  • Dial Pattern: Pattern=NXXNXXXXXX

Save your settings and reload the dialplan. You now can skip down to step #25. NOTE: You will not be able to receive outside calls or make calls to numbers other than toll-free ones.

Our preference is that you use our Platinum Provider, Skyetel, for your default trunk and DID because they offer quadruple redundancy so you never miss a call. Sign up for Skyetel service and take advantage of the Nerd Vittles specials which include a $10 credit to kick the tires. 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. You can also port in your DIDs at no cost for 60 days after funding your account. Credit is limited to one per person/company/address/location. Effective 10/1/2023, $25/month minimum spend required.

We don’t recommend trunk registrations with a publicly exposed server because it creates a potential attack vector for intruders and any intrusion would be undetectable from the PBX since the attacker could make unauthorized calls after registering directly with your SIP provider. For this reason, Skyetel does not use SIP registrations to make connections to your PBX. Instead, Skyetel utilizes Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. Here’s a typical setup for Incredible PBX 16 for VitalPBX:

  • Name: IncrediblePBX
  • Priority: 1
  • IP Address: IncrediblePBX-Public-IP-Address
  • Port: 5062
  • 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 and fund your account) or purchasing new ones under the Buy Phone Numbers menu option.

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

If you’d like additional details on why we recommend Skyetel, see this Nerd Vittles article.

On the VitalPBX side, we need to add a new Skyetel trunk. Navigate to PBX:External:Trunks:PJSIP. The VitalPBX Trunk setup should look like the following for Skyetel. If you’d like to cut-and-paste the entries for the Match field, here you go:

52.41.52.34,52.8.201.128,52.60.138.31,50.17.48.216,35.156.192.164


[popup url="https://pbs.twimg.com/media/EGDhgsXWsAIbmw1?format=jpg&name=medium" width="1200″ height="700″][/popup]

In Admin:Security:Firewall:WHITELIST, you’ll need to individually Add the five Skyetel IP addresses used in the Match field above and then SAVE your firewall settings.

Finally in PBX:Incoming Calls:CID Modifiers, add a new entry for Skyetel with Skip/Length = 2/10 and Save your settings.

23. Before your PBX can receive calls, you’ll need at least one Inbound Route. This tells the PBX how to route calls from one or more phone numbers (DIDs) that you own to a destination on your PBX, e.g. an extension, a ring group, an IVR, or custom context. Navigate to PBX:External:Inbound Routes to get started. Let’s set up a default inbound route for all the DIDs you have acquired from Skyetel in step #22. Fill in the fields shown below. Then SAVE.

  • Routing Method: Default
  • Description: Default Skyetel
  • DID Pattern: [leave blank for ALL DIDs]
  • CallerID Modifier: Skyetel
  • Inbound Destination: Custom Contexts -> IncrediblePBX

24. Before you can make outbound calls from extensions on your PBX, you’ll need at least one Outbound Route. This tells the PBX which provider to use to complete calls dialed with a certain sequence of numbers. For example, you probably would want 10-digit numbers routed to Skyetel. And, if users dial 1 and then a 10-digit number, you’d probably want those calls routed to Skyetel as well. To create this outbound route, navigate to PBX:External:Outbound Routes. Fill in the fields shown below. Click ADD to add a second Dial Pattern. Click SAVE and Reload Dialplan when finished.

NOTE: While you can "spoof" any CallerID number here, it is only legal to assign CallerID numbers that you actually own. Most carriers do not forward CallerID names to destinations regardless of what you enter here. The CallerID name and number will be shown in your CDR logs: Reports:CDR Reports:CDR.

  • Description: Skyetel-OUT
  • Trunks: Skyetel
  • Outbound CID: [Your Name and CallerID Number]
  • Overwrite CID: YES
  • Dial Pattern: Prepend=1 Pattern=NXXNXXXXXX
  • Dial Pattern: Pattern=1NXXNXXXXXX

25. For the time being, we strongly recommend disabling IPv6 simply because we don’t have the necessary confidence that all of the security mechanisms are in place for IPv6. Here’s how on the CentOS 7 platform:

echo "net.ipv6.conf.all.disable_ipv6 = 1" >> /etc/sysctl.conf
echo "net.ipv6.conf.default.disable_ipv6 = 1" >> /etc/sysctl.conf
sysctl -p
sed -i 's|#AddressFamily any|AddressFamily inet|' /etc/ssh/sshd_config
systemctl restart sshd
sed -i 's|inet_protocols = all|inet_protocols = ipv4|' /etc/postfix/main.cf
systemctl restart postfix

 
26. Outbound email functionality is essential on your PBX. You’ll need it to be alerted to potential issues with VitalPBX, and you’ll need it for delivery of voicemail messages to users. There are a couple ways to implement it, and both are easy. If you want to use the native capabilities of Postfix to send the emails assuming your provider is not blocking outbound SMTP mail from downstream servers, then follow these steps:

  • Insert your FQDN from step #12b into /etc/hosts immediately after 127.0.0.1
  • Replace the contents of /etc/hostname with the same FQDN
  • Issue the following command using your actual FQDN: hostname FQDN
  • Sending yourself an email: echo "test" | mail -s test you@your-domain.com

If you don’t receive the test email message, then the easiest solution is to configure PostFix as an SMTP Relay using a Gmail account. You can do this easily from within the VitalPBX GUI. Navigate to Admin:System Settings:Email Settings and click the External Mail Server tab. Be sure that Gmail is selected and enter your Gmail name and password in the fields provided. Save your settings and send yourself an email using the field provided.

27. Once you get outbound email flowing, jump down to the next section and obtain IBM TTS and STT passwords. Now set up Voicemail Transcription with Email Message Delivery:

  a. After logging into your VitalPBX server as root using SSH/Putty:

cd /tmp
mkdir sendmail
cd sendmail
wget http://incrediblepbx.com/sendmailibm-vitalpbx.tar.gz
tar zxvf sendmailibm-vitalpbx.tar.gz
rm -f sendmailibm-vitalpbx.tar.gz
mv usr/sbin/sendmailibm /usr/sbin
cd /etc/asterisk/ombutel
echo "[general](+)" > voicemail__60-1-transcript.conf 
echo "; format=wav|wav49|gsm" >> voicemail__60-1-transcript.conf
echo "mailcmd=/usr/sbin/sendmailibm" >> voicemail__60-1-transcript.conf
chown apache:apache voicemail__60-1-transcript.conf
rm -rf /tmp/sendmail

 
  b. Restart Asterisk core services: asterisk -rx "core reload"

  c. Edit /usr/sbin/sendmailibm and insert your IBM Watson STT APIkey on line 23. Change the language on line 31 if you don’t want en-US. Then save the file.

  d. Log back into the VitalPBX GUI and configure the extensions desired for email delivery of voicemail. For each extension in PBX:Extensions:General, enter an Email Address for delivery of voicemails. In PBX:Extensions:Voicemail, verify the VM settings from step #20.

28. We hesitate to even mention (free) Festival TTS as a text-to-speech alternative because it is so bad compared to IBM TTS. But for those that like always free, here’s how to install it. Once installed, you can issue Festival commands in your dialplan using the keyword Festival followed by the text to be spoken in parentheses.

yum -y install festival
echo "[general]" > /etc/asterisk/festival.conf
asterisk -rx "dialplan reload"
festival_server &
systemctl restart asterisk
echo "/usr/bin/festival_server &" >> /etc/rc.d/rc.local

 

29. If you’d like to test the performance of your cloud-based server, here’s how to deploy and run SpeedTest:

cd /root
wget -O speedtest-cli https://raw.githubusercontent.com/sivel/speedtest-cli/master/speedtest.py
chmod +x speedtest-cli
/root/speedtest-cli

 
30. Associating CallerID Names (CNAM) with inbound calls for display on SIP phones and in the CDR logs is an often-requested PBX feature. There are a few ways to do it. First, for less than a penny a call, you can activate the feature with your DIDs in the Skyetel Dashboard. Or, for about half the cost, you can acquire an OpenCNAM account and activate it in VitalPBX by navigating to PBX:Incoming Calls:CID Lookup. Choose OpenCNAM as the Source and enter your credentials. Then SAVE your settings and reload the dialplan. Then, for each of your Inbound Routes, add OpenCNAM as the CID Lookup source and Update your configuration.

31. Unless you want a full-time job monitoring the size of your logs, remove the fail2ban Asterisk log which grows every 5 seconds. Navigate to Settings:PBX Settings:Log Files and click the Trash icon beside fail2ban. It’s probably a good idea to turn OFF the Notice option for the full log while you’re at it. Then SAVE your changes.

32. Before you do anything else, navigate to Admin:Admin:Backup & Restore, configure and run a Full Backup, and then download the file and keep it in a safe place. Be advised that Backup/Restore doesn’t restore Add-Ons, /var/lib/asterisk/agi-bin, custom contexts (extensions__80*.conf) in /etc/asterisk/ombutel, custom MySQL databases (mysqldump -u root yourDB > yourDB.sql), custom and lenny sound directories in /var/lib/asterisk/sounds, phpMyAdmin, /usr/local/sbin, and /etc/crontab.

Obtaining IBM Watson TTS and STT Credentials

Incredible PBX uses IBM Watson® for TTS and STT support. This Nerd Vittles tutorial will walk you through getting your IBM account set up and obtaining both your TTS and STT credentials. Be sure to write down BOTH sets of credentials which you’ll need in a minute. For home and SOHO use, IBM access and services generally is FREE even though you must provide a credit card when signing up. Details are provided when you sign up. If you ever forget your passwords, you can retrieve them by navigating to Resource List:Services:TTS or STT:View Full Details:Show Credentials.

Obtaining Wolfram Alpha Credentials

When people ask what exactly Wolfram Alpha is, our favorite answer was provided by Ed Borasky.

It’s an almanac driven by a supercomputer.

That’s an understatement. It’s a bit like calling Google Search a topic index. Unlike Google which provides links to web sites that can provide answers to queries, Wolfram Alpha provides specific and detailed answers to almost any question. Here are a few examples (with descriptions of the functionality) to help you wrap your head around the breadth of information. For a complete list of what’s available, visit Wolfram Alpha’s Examples by Topic. Type a sample query here. Some of our favorites include:

Weather in Charleston South Carolina
Weather forecast for Washington D.C.
Next solar eclipse
Otis Redding
Define politician
Who won the 1969 Superbowl? (Broadway Joe)
What planes are flying overhead now? (flying over your server’s location)
Ham and cheese sandwich (nutritional information)
Holidays 2012 (summary of all holidays for 2012 with dates and DOW)
Medical University of South Carolina (history of MUSC)
Star Trek (show history, air dates, number of episodes, and more)
Apollo 11 (everything you ever wanted to know)
Cheapest Toaster (brand and price)
Battle of Gettysburg (sad day 🙂 )
Daylight Savings Time 2012 (date ranges and how to set your clocks)
Tablets by Motorola (pricing, models, and specs from Best Buy)
Doughnut (you don’t wanna know)
Snickers bar (ditto)
Weather (local weather at your server’s location)

Before you can actually use our TTS implementation of Wolfram Alpha, you’ll need to obtain a free Wolfram Alpha account. As you can imagine, there have to be some rules when you’re using someone else’s supercomputer for free. So here’s the deal. It’s free for non-commercial, personal use once you sign up for an account. But you’re limited to 2,000 queries a month which works out to almost 70 queries a day. Every query requires your personal application ID, and that’s how Wolfram Alpha keeps track of your queries. Considering the price, we think you’ll find the query limitation generous compared to other web resources.

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

Configuring Your Incredible PBX Credentials

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

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

If you ever want to learn how to develop applications for Asterisk, these scripts coupled with the dialplan code included in /etc/asterisk/ombutel/extensions__80-1-incrediblepbx.conf will point you in the right direction with easy to follow examples.

Managing the AsteriDex SQLite3 Database

We’ve alluded to the AsteriDex database in a couple of VitalPBX articles but never mentioned how to access it. Using a browser, point it to http://server-ip/asteridex4. You can add, edit, display, and delete entries from there. Before you can make changes in the database, issue the following command after logging into your server as root:

chown asterisk:apache /var/lib/asterisk/agi-bin

Taking Incredible PBX for a Test Drive

You can take Incredible PBX for VitalPBX on a test drive in two ways. You can call our server, and then you can try things out on your own server and compare the results. Call our IVR by dialing 1-843-606-0555. For our international friends, you can use the following SIP URI for a free call: 10159591015959@atlanta.voip.ms. For tips on setting up your own secure, hybrid SIP URI with VitalPBX, see our original tutorial. The FreePBX® setup is virtually identical except for the location of the custom SIP setting for match_auth_username=yes. On a VitalPBX server, you will enter it here: Settings:Technology Settings:SIP Settings:CUSTOM.

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

  • 0. Chat with Operator — connects to extension 701
  • 1. AsteriDex Voice Dialer (412) – say "Delta Airlines" or "American Airlines" to connect
  • 2. Conferencing (2663) – log in using 1234 as the conference PIN
  • 3. Wolfram Alpha Almanac (4747) – say "What planes are flying overhead now?"
  • 4. Lenny (53669) – The Telemarketer’s Worst Nightmare
  • 5. Today’s News Headlines (951) — courtesy of Yahoo! News
  • 6. Weather by ZIP Code (947) – enter any 5-digit ZIP code for today’s weather
  • 7. Today in History (86329) — courtesy of OnThisDay.com
  • 8. Call Extension 701 — on your local PBX
  • 9. DISA Voice Dialer (3172) — say any 10-digit number to be connected
  • *. Current TIME and Date (8463) — courtesy of VitalPBX

CAUTION: We have intentionally disabled outbound calls using Option #9 and redirected callers to Lenny. The reason is that an unscrupulous caller could easily run up your phone bill by entering a number with expensive destination charges. If you wish to enable the feature, despite the risks, you can edit extensions__80-1-incrediblepbx.conf and make the change.

You can call your own IVR in a few ways. From an internal VitalPBX phone, dial D-E-M-O (2663) to be connected. Or simply dial the number of the DID you routed to the Incredible PBX Custom Context. Local users can also dial the individual feature codes shown in parentheses above. Be sure that you heed AND test the CAUTION documented above.

Originally published: Monday, October 21, 2019





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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.
 



Finding Utopia: In Search of the Perfect VoIP Server Platform

Over the past decade, there is no subject that we have devoted more resources to than searching for the best platform on which to run a VoIP server. While our experience primarily has focused on finding the perfect fit for Incredible PBX®, much of what follows applies equally to any other Linux-based VoIP server including Wazo, Issabel, VitalPBX, and 3CX. Today we’d like to share what we’ve learned. Incredible PBX is a complex application. With close to a thousand moving parts, it requires major computing resources to support not only Asterisk® and FreePBX® but also an Apache web server, a MySQL database server, a SendMail server, a HylaFax server, and a Linux firewall with both IPtables and Fail2Ban.

Let’s begin by ticking off the platforms that Incredible PBX currently supports. These include stand-alone dedicated hardware from beefy Dell servers to the Intel NUC and Raspberry Pi. Then there are the virtual machine platforms including VirtualBox, VMware ESXi, and Proxmox. In the Cloud space we’ve covered the stratosphere from the high end with $25/month Google Cloud and Amazon EC2 instances to the dedicated $15/month VoIP platform with RentPBX to the $5/month KVM platforms including Digital Ocean and Vultr to the $2.25/month OVH KVM offering to the $1/month OpenVZ providers including HostedSimply, HostFlyte, Hosting73, HostBRZ, SnowVPS, and AlphaRacks. Have there been some train wrecks along the way? Absolutely. Just search the PIAF Forum for the threads on CloudAtCost, WootHosting, and HiFormance for the war stories and our battle scars. We would be remiss if we didn’t thank the dozens of PIAF Forum volunteers who have endured years of suffering at the hands of some of these providers to make today’s article possible.

So what have we learned? Unless you’re building a VoIP platform as a tinkerer to support just your family, there is zero reason to choose dedicated hardware. And, for home use, with the availability of the $35 Raspberry Pi 3B+, buying a beefier piece of hardware to host your VoIP platform makes no sense. Not only will it be considerably more expensive both to purchase and to operate, but the performance of your VoIP server will be indistinguishable from what you’d see using a Raspberry Pi 3B+. Exhibit A is our $125 RasPi WiFi setup for traveling.

The downsides of dedicated hardware are numerous. In addition to the expense of the platform itself and the monthly cost of electricity, there also are other challenges. First, outages from most Internet service providers are frequent occurrences of unpredictable duration. Second, ISPs typically provide a dynamic IP address which is not a good fit for VoIP platforms that rely upon your IP address to reliably make and receive VoIP calls. Third, making backups using dedicated hardware is typically more expensive and less frequent than performing similar tasks with a cloud-based server. Recovery is easy with a spare SD card.

The virtual machine platforms certainly have their place in the corporate world. And, if your company already has a server farm full of VMware servers, then taking advantage of that platform to host your PBX makes perfect sense. Performance will probably never be an issue, and you’ll avoid the task of babysitting the hardware leaving that to a staff of dedicated employees. And, hopefully, someone else is making frequent backups of your VoIP server. For home users that already have a beefy desktop machine, a VirtualBox-based PBX is certainly an option worth considering although it again puts you in the driver’s seat of dealing with backups, Internet outages, and performance hiccups when your desktop machine is being used for tasks that consume substantial computing resources.

If you haven’t already guessed, our recommended VoIP platform will almost always be cloud-based. Not only does it offload most server and network management headaches, but more often than not, it’s a more dependable platform with better performance at a comparable or less expensive cost than using your own hardware. So here’s the Golden Nugget of our findings. When it comes to cloud providers, you can forget the old adage that you get what you pay for. You don’t. Our experience suggests it’s just the opposite when it comes to running a VoIP server. With cloud providers, what you typically get by paying more is an improvement in the odds that your provider will still be around when next year rolls around. Getting over that hurdle is simple. Make frequent backups. If there are a multitude of available providers offering similar services, backups are the best insurance you can have, and they cost you almost nothing. In fact, Incredible Backup handles the task with ease AND reliability. Once you get past the vendor longevity issue, the only things that really matter with a cloud platform are stability and performance. While the high-end providers certainly deliver stability, our experience suggests their performance is nothing short of abysmal unless you’re willing to pay through the nose. By way of example, our experimental Google Cloud server running as a $25/month Standard VPS instance with zero daily calls still receives regular alerts from Google recommending that the instance be upgraded to the next pricing tier which starts at $48.95/month. Performance-wise, our subjective comparison of the $25/month Google Cloud instance is virtually identical to what we are seeing on a stand-alone $35 Raspberry Pi. As a VoIP server platform, the so-called free tier with Google Cloud that provides 600K of RAM and a shared virtual CPU is laughable, and that’s being charitable.

We haven’t spent a lot of time using Amazon EC2 in the past couple years primarily because their platform was even more expensive than Google’s. But, if money is no object, it’s certainly a hosting platform worth exploring. For most VoIP applications, it doesn’t make good financial sense.

That narrows our search for the perfect VoIP platform down to two categories: the KVM and OpenVZ platforms. As a general rule of thumb, with a given provider’s offerings you can expect performance to be comparable but you typically will pay at least double for a KVM platform as opposed to an OpenVZ platform with similar RAM, storage, and bandwidth. In a nutshell, KVM servers provide your virtual machine with its own Linux kernel while OpenVZ servers share a kernel over which you have no control. If you run a VoIP application that requires kernel access, this matters. If you plan to expose your server to the public Internet, the KVM option also is desirable because it allows you to run ipset in conjunction with the Linux firewall to block entire countries from accessing your server. In the case of Incredible PBX servers which rely upon a firewall limiting access to whitelisted IP addresses, there is little reason to choose the KVM platform based solely upon performance or security.

The elephant in the room with providers below the Google and Amazon tier is reliability. In the case of Digital Ocean and Vultr, they both have been around for many years now with excellent ratings in virtually every category. Both provide financial support for our open source projects through referral revenue, but we’d use them anyway. The virtual machine pricing from the two companies is almost identical. Except for extremely busy VoIP implementations, their 1GB RAM offering has proven to be a perfect choice at $5 a month. If you don’t mind paying by the year, you can’t beat OVH’s current $2.25/month KVM offering with 2GB RAM and 20GB SSD. They, too, have been around for years. At one time or another, OVH hosted much of 3CX’s cloud infrastructure. All offer scaling options to meet even the most demanding requirements. On the D.O. and Vultr platforms, you can add automatic backups for an additional $1 a month (20% surcharge) which is dirt cheap insurance. We have run both Incredible PBX and 3CX servers on all of these platforms with no outages or other issues… and weekly backups. Both Digital Ocean and Vultr also provide excellent web tools to manage your server, and the chance of any of these providers going out of business is extremely remote. We highly recommend all of them.

FULL DISCLOSURE: We have no business relationship with OVH or any of the following VPS providers and receive no referral commissions of any kind from any of them.

For some users and especially those that just want to learn about VoIP and tinker, there is yet another tier of providers. At roughly $1/month, their VPS services are a fraction of the cost of Digital Ocean and Vultr, but backups become your responsibility and at least one previous provider that many of us used went out of business. Those without a backup lost everything.

Choosing one of these providers comes down to balancing the risks versus the financial savings. We have nearly a dozen of these $1/month servers in operation all across the United States. While the VPS providers are different, almost all of the servers are hosted by ColoCrossing in Los Angeles, New York, Chicago, Dallas, or Atlanta. These VPS providers typically rent machines directly from ColoCrossing, and the performance of their VPS offerings varies depending upon the number of users each provider authorizes on each server. Some are obviously more greedy than others. And we’ve actually done the hard work of finding the reliable ones while rejecting at least as many that proved to be pretty awful.

Server locations and special signup details for these VPS providers are documented in our previous article. Average cost is about $1/month on an annual contract with a 1Gbit port or *free 1Gbit port upgrade on request based upon LowEndBox offer. All offer money-back guarantees for at least 24 hours so you can do your own testing if you hurry. Protect yourself by paying with PayPal which gives you 6 months to dispute a charge if the provider happens to go belly up. NOTE: The sort order below reflects our subjective performance evaluation.

ProviderRAMDiskBandwidthPerformance as of 12/1/19Cost
CrownCloud KVM (LA)1GB20GB +
Snapshot
1TB/month598Mb/DN 281Mb/UP
2CPU Core
$25/year
Best Buy!
Naranjatech KVM (The Netherlands)1GB20GB1TB/monthHosting since 2005
VAT: EU res.
20€/year w/code:
SBF2019
BudgetNode KVM (LA)1GB40GB RAID101TB/monthAlso available in U.K PM @Ishaq on LET before payment$24/year
FreeRangeCloud KVM (Ashburn VA, Winnipeg, Freemont CA)1GB20GB SSD3TB/monthPick EGG loc'n
Open ticket for last 5GB SSD
$30/year w/code:
LEBEGG30

Do we recommend these providers? Absolutely, with a couple of caveats. First, there is no guarantee that one or more of them may not go out of business at some point. The odds of several of them going under at the same time are fairly slim since none are related that we’re aware of. Second, make frequent backups when you make changes to your PBX and copy the Incredible Backups to a different location. Third, bring up a second VPS platform in a different location and keep it current with your latest backup. You could bring up all six of these platforms for roughly the same monthly cost as one Digital Ocean or Vultr virtual machine that’s running with automatic backups. If you can’t afford a second $1/month VPS platform, then at least create a matching VirtualBox platform, restore your backup, and make sure it is functional before deploying your VPS in the Cloud. It’s in your hands now. Enjoy!

Originally published: Monday, April 8, 2019



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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