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The Most Versatile VoIP Provider: FREE PORTING

Meet Linphone: Free Worldwide Calling to Anybody with SIP

Earlier this year we demonstrated how to set up a publicly-accessible Asterisk® server to enable free worldwide calling using SIP URIs which are email-like addresses for VoIP and video calls. But not everyone has an Asterisk server so today’s tutorial extends free calling to everyone with a Windows or Linux PC, a Mac, or any smartphone or tablet. All you need is a desktop computer with wired or wireless Internet access or, on a smartphone or tablet, a cell data plan or WiFi connection will suffice. When friends sign up, their calls also will be free.

The secret sauce on all of these platforms is the Linphone app (shown above) which can be downloaded and used at no cost. Source code is available for those that want it. Use it as often and for as long as you like. Here are the Linphone download links for each of the platforms:

The only other piece you’ll need to get started is a free Linphone SIP account. Sign up here. Once you’ve signed up, simply respond to the confirmation email to activate your account. Your registration gets you credentials to plug into your Linphone app that you downloaded above. In addition, it gets you a free Linphone SIP URI which looks something like this: yourname@sip.linphone.org. This is the SIP URI address that anyone in the world can use to contact you. Here are the pieces you’ll need to plug into your desktop or smartphone app:

  • Account Name
  • Account Password
  • Domain: sip.linphone.org

Be very careful not to lose your password. You can’t retrieve it, and you can’t change it without knowing the original password. All you can do is delete your account and start over.

The Linphone feature set is downright impressive. Here’s what you and your friends will be using at zero cost:

IMPORTANT TIP: Missing audio or one-way audio is a common problem on SIP calls. For best results, configure your account in the Linphone app to use UDP for the Transport, disable the Outbound Proxy, configure stun.linphone.org as the Stun Server, and enable ICE. In Network settings, turn off IPv6 and Media Encryption. In Audio settings, enable Opus, G.722, PCMU, and PCMA only. In Video settings, enable both VP8 and H.264. Then close the app and reopen it.

Once you have your Linphone credentials, another option in addition to using one of the SIP clients above is to acquire a stand-alone SIP telephone which can easily be connected to your Linphone SIP account. While there are literally hundreds of SIP telephones from which to choose, here’s a $35 offering from Grandstream that we use. It’s available from Amazon.1

Unlike other proprietary communications apps, the beauty of using Linphone with its native SIP URI support is you can call any SIP phone in the world for free whether the recipient uses Linphone or not. For example, to annoy your friends and spammers, you can transfer their calls to Lenny: 2233435945@sip2sip.info or 883510001198938@sip.inum.net. And here are some other SIP URI calls you might want to try. Store them in your Linphone Phonebook.

Yahoo News Headlines - news@demo.nerdvittles.com
Yahoo News Headlines - 951@demo.nerdvittles.com
Weather by Zip Code  - weather@demo.nerdvittles.com
Weather by Zip Code  - 947@demo.nerdvittles.com
Directory Assistance - information@demo.nerdvittles.com
Directory Assistance - 411@demo.nerdvittles.com
Lenny for Spammers   - 53669@demo.nerdvittles.com
Technical Support    - 0@sip.incrediblepbx.com
Call Any TollFree #  - **1800XXXXXXX@tollfree.future-nine.com

There are now more than 2,000 VoIP networks worldwide that support SIP URI access. Any person or organization with an account on any of these networks can be reached at no cost via SIP URI or via several hundred PSTN numbers. Using a SIP URI dialing prefix, you can call any referenced network@sipbroker.com. For example, *656news@sipbroker.com would reach the Nerd Vittles News Headlines from Yahoo. Or choose a local access number from the SipBroker worldwide directory, e.g. 702-789-0530 and then dial *656951 at the prompt.

Of course, every 3CX platform provides dedicated SIP URIs for every extension on the PBX. Our recent article covers adding SIP URI access to any Asterisk PBX.

If you want to associate a phone number with your Linphone SIP URI, you can do it in a couple of ways. First, using a smartphone, you can link your cell number to Linphone within the Linphone app itself. If you have a free DID from IPComms, you can point it to your Linphone SIP URI. If you have a $1/month CallCentric DID, it can also be pointed to your Linphone SIP URI. A 25¢/month iNum DID from LocalPhone.com also can be pointed to your SIP URI. LocalPhone supports Nerd Vittles through referral revenue from your 25¢ investment. 🙂

Speaking of iNUMs, you can reach anyone with an iNUM DID by dialing the iNUM number in SIP URI format: 8835100xxxxxxxx@sip.inum.net. One of the real beauties of signing up for an iNUM number as well is that it can be reached in most places around the globe by dialing a local number from any telephone. As part of the iNum initiative, local access numbers have been established in more than 50 countries around the globe. By placing a local call from any telephone to one of these local access numbers, any individual with an iNum phone number anywhere in the world can be reached without further cost. Here is a current list of the local access numbers. If the link is down (frequently), try here or here or the iNUM listing here. Once your call is answered, simply enter the 15-digit iNum phone number you wish to reach, and you will be connected. It’s worth pointing out that iNUMs aren’t as unwieldy as they may appear. The numbers always begin with 8835100 followed by 8 digits starting with a zero.



And another iNUM listing from DSL Reports:

Country             City                     Access Number
------------------- ------------------------ ---------------
Argentina           Buenos Aires             +54 1159839500
Australia           Sydney                   +61 280148200
Austria                                      +43 720880500
Bahrain                                      +973 16199200
Belgium             Brussels                 +32 28081771
Brazil              Brasilia                 +556135500791
Brazil              Florianopolis            +554840420809
Brazil              Rio De Janeiro           +552135006959
Brazil              Sao Paulo                +551146803621
Bulgaria            Sofia                    +359 24917555
Canada              Calgary                  (403) 775-1446
Canada              Edmonton                 (780) 669-9257
Canada              Halifax                  (902) 982-6937
Canada              London                   (519) 488-9336
Canada              Montreal                 (514) 907-7500
Canada              Ottawa                   (613) 686-4519
Canada              Quebec City              (418) 800-0384
Canada              St. Johns, Newfoundland  (709) 757-0060
Canada              Regina                   (306) 988-1600
Canada              Toronto                  (416) 800-4303
Canada              Toronto                  (647) 724-8777
Canada              Vancouver                (778) 786-3497
Canada              Winnipeg                 (204) 272-8182
Chile               Santiago                 +56 25813444
Croatia             Zagreb                   +385 17776363
Cyprus              Nicosia                  +357 22030500
Czech Republic      Prague                   +420 246019777
Denmark                                      +45 69918686
Dominican Republic  Santiago                 (829) 947-9610
El Salvador                                  +503 21131899
Estonia                                      +372 6681881
Finland             Helsinki                 +358 942419200
France              Paris                    +33 170619800
Germany             Frankfurt                +4969257385876
Germany             Frankfurt                +4969257380439
Greece              Athens                   +30 2111768444
Hungary             Budapest                 +36 14088951
Ireland             Dublin                   +353 15262600
Israel              Tel Aviv                 +972 37219555
Italy               Rome                     +39 0662207777
Japan               Tokyo                    +81 345209777
Latvia              Vilnius                  +370 52059090
Lithuania                                    +371 67652500
Luxembourg                                   +352 20880108
Malta                                        +35627780107
Mexico              Guadalajara              +52 3346242977
Mexico              Mexico City              +52 5511678222
Mexico              Monterrey                +52 8141703540
Netherlands         Amsterdam                +31 208080808
New Zealand         Auckland                 +64 99250499
Norway              Oslo                     +47 21031306
Panama                                       +507 8322488
Peru                Lima                     +51 17085500
Poland              Warsaw                   +48 223982688
Portugal            Lisbon                   +351 308803219
Puerto Rico         Bayamon Norte            (787) 395-7140
Romania                                      +40 318103500
Singapore                                    +65 31581212
Slovakia            Bratislava               +421 233002555
Slovenia            Ljubljana                +386 16001422
South Africa        Johannesburg             +27105002854
South Africa        Pretoria                 +27120042701
Spain               Barcelona                +34 931815653
Spain               Madrid                   +34 911883777
Sweden              Stockholm                +46 852500111
Switzerland         Zurich                   +41 435006262
United Kingdom      London                   +44 2033556363
United States       Albuquerque              (505) 225-8243
United States       Charlotte                (980) 202-0283
United States       Charlotte                (980) 236-0398
United States       Kansas City              (913) 951-0932
United States       Chicago                  (312) 253-4880
United States       Houston                  (713) 474-2323
United States       Los Angeles              (213) 221-3799
United States       New York                 (646) 843-6969
United States       Phoenix                  (602) 354-9444
United States       San Diego                (619) 330-9640
United States       San Francisco            (650) 360-0999
United States       Santa Barbara            (805) 308-9649
United States       Seattle                  (206) 420-5904
United States       Spokane, WA              (509) 931-0459
United States       Tacoma, WA               (253) 343-1529

More iNUM details are available here. If sip.inum.net is down, try 81.201.82.50.

Let’s tie all the pieces together now. Linphone gives you and your friends a free SIP URI as well as a SIP client for any platform to make and receive SIP voice and video calls. You can associate this SIP URI with your cellphone number as well as a free or almost free phone number (DID) that’s available from IPComms, CallCentric, and other providers. If you sign up for a LocalPhone iNUM number, you also can associate it with your Linphone SIP URI. So you can be reached on your Linphone client by SIP URI, by iNUM, and by regular phone numbers. You can place unlimited calls to any SIP URI or iNUM worldwide at no cost. What’s not to like?

Deploying Linphone as an Asterisk Trunk

If you don’t have an Asterisk PBX, you can stop reading here. The good news is you can also use a Linphone SIP account as a SIP trunk on your Asterisk PBX. Once configured, you can add an Incoming Route and send the incoming Linphone SIP URI calls to any destination desired: an extension, a ring group, an IVR, or even a Conference Room. Using the FreePBX® or Incredible PBX® GUI, create a chan_SIP Trunk and name it linphone. In the PEER DETAILS, enter the following using your actual Linphone username and password:

type=friend
qualify=yes
insecure=port,invite
host=sip.linphone.org
disallow=all
context=from-trunk
dtmfmode=rfc2833
allow=g722&ulaw
fromuser=your-username
defaultuser=your-username
secret=your-password

For the Registration String: your-username:your-password@sip.linphone.org/99999

Next, create an Inbound Route using 99999 as the DID entry. Route the call to your desired destination, SAVE your settings, and you’re in business.

There’s one more nice surprise. Linphone accounts work much like the old key telephones and Google Voice setup that we all knew and loved. What that means is you can register the same Linphone account in multiple places, e.g. as an Asterisk trunk and elsewhere using one of the Linphone softphone apps. When incoming calls to your SIP URI arrive, they will ring on both your Asterisk PBX and your Linphone softphone as long as you haven’t routed the Linphone trunk to a destination that automatically answers the calls such as an IVR.

HINT: If you’re using dual registrations and routing the Linphone trunk to an extension, we recommend disabling voicemail on that extension so that Asterisk doesn’t automatically answer the call and send it to voicemail when the extension is not registered or answered.

To make outbound calls from extensions on your PBX using the Linphone trunk, the easiest way is to create custom extensions in the [from-internal-custom] context in /etc/asterisk/ extensions_custom.conf. Make up an unused extension number (90210 in this example), enter the Linphone account name you wish to call (acctname in this example), save the file, and reload your dialplan: exten => 90210,1,Dial(SIP/acctname@linphone).

Another way to create a Custom Extension is using the FreePBX or Incredible PBX GUI. Under Applications -> Extensions -> Add Custom Extension, assign an extension number for the extension. Click on the Advanced tab and enter SIP/acctname@linphone in the Dial field. Click Submit button and reload the dialplan at the prompt. Enjoy your worldwide free calling.

Originally published: Monday, April 29, 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.

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

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



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

Keep On Trunkin’: Free International VoIP Calling Returns



Today we’re taking a fresh look at the international calling marketplace by updating the best VoIP deals available. FreeVoipDeal once again takes the prize with the best selection of "free" international calling destinations at the lowest prices. Below we’ll provide a quick tutorial to transform your Incredible PBX server into an international calling platform at minimal cost.

Here’s How It Works. For every 10 euros ($10.72) you deposit into your account, you’ll get 300 minutes a week of free calls to a specific list of countries for 120 days. After you exhaust your free minutes, calls to the "free" countries revert to their standard VoIP rates. You can also call anywhere else in the world at very reasonable per minute rates that compare favorably with other SIP providers around the world. The beauty of a PBX and SIP trunks is you can mix and match as many providers as you like to take advantage of favorable calling rates to multiple countries. We’ll walk you through the FreeVoipDeal trunk setup below.

Betamax 101. There are a few things you need to know about the so-called Betamax VoIP services up front. Most importantly, they change rates and free countries more frequently than college kids change partners. The calling rate to some country from some Betamax provider changes almost every day because Betamax has dozens of companies offering similar services with differing rates and freebies. Here’s an very old spreadsheet that will give you a good idea of what you’re up against. Don’t depend upon it for the current rates. You’ll need to visit the actual site(s) for their current rate tables or visit this site (not) maintained by Betamax for a country-by-country comparison by provider. That’s another way of saying DON’T BLAME US IF YOUR 3-HOUR CALL TO ANTARCTICA CHANGED FROM 20¢ PER MINUTE TO $1 PER MINUTE OVERNIGHT. IT PROBABLY WON’T, BUT IT MIGHT.

One other word of warning. Some Betamax sites (marked with a red asterisk in the Betamax country table) such as powervoip.com have good calling rates, but they tack on a 3.9¢ connection fee to every call. If you make lengthy calls, it’s not a big deal. If you make numerous short calls, it drives your discount calling rates through the roof. Before making a lengthy call to a remote destination, spend the two minutes it takes to look up the current rate on the actual Betamax web site and take a snapshot of the page for your records. Here’s another tip. If you make frequent calls to Antarctica, spend a little time doing your homework. Review the latest Betamax spreadsheet to track down the cheapest rates. Then double-check the actual sites for the current rates. There’s a $100+ difference in the cost of a 3-hour call at €.20/minute from some Betamax sites versus the €.70/minute rate at some other Betamax sites. THIS OFTEN CHANGES! HINT: Don’t use FreeVoipDeal for Antarctica.

Today we’ll be focusing on the company we’ve tracked for many years, FreeVoipDeal.com. Except for the domain name, the setup with other Betamax providers is similar but not identical. And, of course, you’ll have to kick in another deposit to make free calls from each site. The length of the Freebie period also may vary so read the terms carefully. FreeVoipDeal actually hasn’t changed much since our first visit about five years ago. In fact, we still had most of our ten euro credit so we could play all we wanted even though the calls were no longer free since our four month window has long since expired.

Here’s the February 23, 2019 Freebie list by country. Don’t depend upon it! Check their actual web site or the Betamax country summary for current freebies and current rates. Here’s a great trick to remember. When you visit the FreeVoipDeal Rate Table, click on the Out of Minutes tab for a quick listing of all the Free Calling Countries as well as the rates once you’ve used up your four months or 300 weekly minutes of free calls. With few exceptions, most of the "free countries" still have a rate of 1.1¢ per minute even after you run out of minutes.

 

How Free International Calling Works

Placing international calls through FreeVoipDeal can be done in a number of ways. That’s the real beauty of a PBX. First, you can either load an app to make the calls if your smartphone or PC supports it. With Incredible PBX, you can use a SIP phone to dial a FreeVoipDeal number directly through your PBX, or you can dial a DISA access number or SIP URI from anywhere to connect to your PBX and then enter your DISA password after which you will get a second dial tone to place an international call using your FreeVoipDeal trunk. The beauty of the DISA approach is you can call into your PBX from any telephone to place free or dirt cheap international calls.

Using Incredible PBX 13 and DISA for Calling

On the Incredible PBX platform, you can use the DISA application to provide secondary dialtone for processing international calls. A phone number and trunk will receive incoming calls bound for DISA from your cellphone. An inbound route will only forward incoming calls to DISA that match your cellphone number. A secondary trunk from FreeVoipDeal or other providers will be used to process outgoing international calls that are dialed using DISA. We’ll create an outbound route or rule for every country to which you want to authorize international calling. Each of these outbound routes will point to the least expensive (or free) trunk to complete the call. In the VoIP world, you actually could have dozens of outbound trunks that handle international calls based upon the country codes of each international call. This lets you take advantage of the best calling rates for each country. We will block international calls to country codes not specifically authorized.

Just to restate the obvious, a misconfigured DISA application that allows the world to make international calls on your nickel can get expensive quickly. We’ll protect today’s DISA setup for Incredible PBX with three layers of protection. First, we’ll require that the CallerID of the incoming call match your cellphone number. While this isn’t failsafe since CallerID numbers can be spoofed, it does reduce the risk considerably. Second, to make DISA calls, you’ll have to know the incoming phone number or SIP URI managing DISA on your PBX. And third, you’ll have to enter the correct DISA PIN before being prompted for an international number to dial. Without all three, nobody gets to make an international call on your nickel. Just remember, compromising DISA on your PBX is just as risky as handing out your credit card to a stranger so follow the setup steps below carefully. And then TEST, TEST, TEST to make sure strangers can’t access your DISA setup. We’ll show you how.

Here’s an overview of the DISA setup drill once you have Incredible PBX running. We’ll walk through each of the six steps below. Don’t get frustrated. There are a number of steps, but none of them are difficult. Just pretend you’re baking cookies and don’t skip any steps.

  1. Set Up Your Trunk to Process Incoming DISA Calls
  2. Set Up Your Trunk(s) to Process Outgoing International Calls
  3. Configure DISA with a Very Secure Password
  4. Configure an Inbound Route to Limit Incoming DISA Calls to Your Cellphone #
  5. Configure an Outbound Route for Each International Country Code
  6. Test, Test, Test

1. Setting Up Incoming DISA Call Trunk

Before you can make calls to your PBX, it’ll need a phone number (known affectionately as a DID). As installed, Incredible PBX includes preconfigured SIP trunks from about a dozen SIP providers. All you’ll need is credentials from the company you wish to use. You can obtain a free DID here. To obtain your own SIP URI, read our tutorial.

2. Trunk Setup for International Calling

We’re going to walk you through setting up a trunk with FreeVoipDeal to handle free international calls to certain countries documented above. This may not be the best fit for you depending upon the international destinations you wish to call. Figure that out first! Then adjust the trunk settings below to match each SIP provider trunk you wish to create. There’s no limit to the number you can have. And, with most of these providers, you pay by the minute for international calls anyway so there is no harm in configuring multiple trunks to take advantage of the best rates calling the countries of your choice. The same applies to all-you-can-eat and "free" trunks except there are varying fees for using the services so you’re probably not going to want a dozen of them even if some of the calls are free after making a periodic deposit. Start with the pink and green entries on the old spreadsheet we referenced for the cheapest historical rates and then visit the actual sites and read the fine print.

To add new trunks to Incredible PBX, use a browser to access the IP address of your server. Login with the default username of admin and the password that you set when your install completed. You can change it with the admin-pw-change script in /root. Once the dashboard appears, click the Connectivity tab and choose Trunks -> Add SIP (chan_sip) Trunk.

For Trunk Name, enter FreeVoipDeal. In the Dialed Number Manipulation Rules section, add a rule for each country code you wish to activate. You can decipher the Country Code for any country at this link. For example, for the United Kingdom, you’d enter a rule like this where 44 is the Country Code and each X represents a required digit in the local area code and phone number. The trailing period means the number includes one or more additional digits. NOTE: DISA calls will not have to be prefixed with 011 to place international calls. Just enter the country code and number to be called. And, we are told that only 441, 442, and perhaps 443 calls to the U.K. are free since those are the designated landline prefixes.




If there are other countries, you wish to support with this trunk provider, you’d click Add More Dial Pattern Fields and insert an additional rule for each country following the example above. If you’ll be using this trunk to make calls in the U.S. and Canada as well, the correct Match Pattern is 1NXXNXXXXXX, and calls will need to be dialed with the 1 to avoid conflicts with international dialing.

Next, we need to enter the Outgoing Settings. For the Trunk Name, enter freevoipdeal. Clear out the entries in Peer Details section and enter the following using your actual FreeVoipDeal credentials for yourusername and yourpassword:

authuser=yourusername
username=yourusername
secret=yourpassword
type=peer
qualify=yes
nat=yes
insecure=port,invite
host=sip.freevoipdeal.com
fromdomain=sip.freevoipdeal.com
dtmfmode=auto
disallow=all
canreinvite=no
allow=alaw&ulaw

Finally, clear out the default entries in User Details and click the Submit Changes button and then red Apply Config button to save your new trunk.

Spoofing Your CallerID. When setting up your FreeVoipDeal account, you can set up one or more numbers to use as your CallerID number on FreeVoipDeal calls. You simply verify the number with a code sent by SMS or phone call from their service. Once you’ve gone through the verification procedure, you can spoof the outbound CallerID on FreeVoipDeal calls using your actual cellphone number. Just add the following entries to your Trunk settings replacing 9991234567 with your cellphone number. Special thanks to @hillclimber on the PIAF Forum for the tip.

fromuser=0019991234567
sendrpid=yes

3. Configuring DISA for International Calling

In the Incredible PBX GUI, we’ll set up DISA by clicking the Applications tab and choosing DISA. Add your new DISA configuration by following this sample. Use a VERY secure password. It’s your phone bill. Once you’ve finished, click the Submit Changes button and then the Apply Config button to save your new DISA setup.



4. Inbound Routing of DISA Calls

Here’s where we lock down your setup so that Incredible PBX only accepts DISA calls from your cellphone number. If you want to allow additional people to use your DISA setup or if you have multiple cellphones, then simply create multiple inbound routes with the 10-digit numbers of each phone to be supported.

In the Incredible PBX GUI, we’ll set up a new Inbound Route by clicking the Connectivity tab and choosing Inbound Routes. If you plan to support multiple phones, then create multiple inbound routes and give each of them a unique Description and CallerID Number that matches the phone number of the cellphone to be supported. Be sure to check the CID Priority Route checkbox and set the correct Destination for your incoming calls. Just fill in the blanks appropriately using this template as a guide. Once you’ve finished, click the Submit button and then the Apply Config button to save your new Inbound Route.



5. Outbound Routing by Country Code

The DISA application is going to obtain the phone number to be dialed and will pass that to the Outbound Routes module. The job of the Outbound Routes module is to examine the phone number passed to it from DISA to figure out which trunk to use to make the outbound call. It then will pass the call to the appropriate trunk which sends the outgoing call on its way to the destination.

For each Dialed Number Manipulation Rule in every Trunk that you set up in Step #2 above, you’ll need a matching Outbound Route if your PBX is used to place calls using multiple trunks. If you’re only using one provider for all of your outbound calls, then we can use a more generic Outbound Route. It’s always a good idea to create the one-to-one match between Outbound Routes and Trunks to make certain that outbound calls are sent to the correct Trunk for processing. So let’s do that using the U.K. trunk we created above.

In the Incredible PBX GUI, we’ll set up a new Outbound Route by clicking the Connectivity tab and choosing Outbound Routes. When the template appears, notice in the far right column that there’s a listing of all your existing Outbound Routes. Calls are actually processed sequentially using the order that these Outbound Routes appear in the list. If there’s no number match in the top route, processing drops to the next route in the list until there is a match AND a successful connection. You can adjust the sequence by dragging the Outbound Routes to a different position in the priority list.

It’s important to use specificity in your Outbound Routes (especially with International calling) to make certain that a call isn’t inadvertently processed by some other trunk. The easiest way to do this is to require the Outbound Route Match Pattern for U.K. calls to be at least 11 digits, e.g. 44XXXXXXXX. (the trailing period is important in that it requires at least one more digit for a match). And we can force a Hangup if the FreeVoipDeal trunk is not available for some reason by adjusting the Destination on Congestion setting. This keeps the call routing from dropping down to the next available Outbound Route in the list if FreeVoipDeal happens to be off-line at some point. So our Outbound Route for U.K. calls should look something like this:



The final step is to move the new Outbound Route for U.K. calls to the top of the Outbound Routes listing in the right column to assure that it is processed first. Once you’ve done that, click the Submit Changes button and then the Apply Config button to save your new Outbound Route AND the adjusted Outbound Route Priority List.

Another alternative in creating Outbound Routes is to use a Dial Prefix that never matches a real phone number to direct calls to a particular trunk. For example, you might use *8 as a dial prefix for FreeVoipDeal calls. By placing *8 in the Prefix column of the Dial Pattern, it will get stripped off before the number is actually passed to the FreeVoipDeal trunk for processing. We actually prefer this setup because it adds an additional layer of security for international calls. If someone were to break into your DISA application by knowing your cellphone number AND your DID AND your DISA password, it’s unlikely they’d also know to prefix outgoing international calls with some arbitrary dial prefix. Just don’t use *8 in case they’re a Nerd Vittles reader. 😉

6. Test, Test, Test!

The easiest way to test the new setup is to place a couple of calls and to watch the Asterisk CLI (asterisk -rvvvvvvvvvv) and see how the calls are processed and who answers at the other end. Then you can apologize for reaching the wrong number.

You can make up your own test methodology, but here’s one that works for us. There are several tests you need to make. First, call your Incredible PBX DID from your authorized cellphone and enter a correct DISA password to see if you get dial tone to make an international call. Then repeat the drill with an invalid password and make sure you don’t get a dial tone. Next, call your Incredible PBX DID from a phone other than your authorized cellphone. You should not get a prompt for a DISA password. Finally, we use the first three digits of a U.K. number to identify a matching NANPA area code. Then, we find hotels in the two matching cities. For example, one might attempt to call a hotel in Bath, England (44 1… ……) and a hotel in Bermuda (441-…-….). The U.K. call should go through, and the Bermuda call should fail. If you pass all three tests with flying colors, you’re good to go.

Using FreeVoipDeal’s MobileVoIP App

FreeVoipDeal also offers a MobileVoIP app that can be used directly on your smartphone (Android, iOS, and Windows phone versions available) using any Wi-Fi, UMTS, 4G/LTE, 3G, GPRS or EDGE connection. The drawback is the lack of the three extra layers of security protection that Incredible PBX using DISA offers. MobileVOIP lets you log in with your registered Betamax credentials and offers the option to use your existing VoIP credit from your smartphone. The downside is that anyone with the app and your credentials can call anywhere and talk for as long as they like on your nickel using any of your registered CallerIDs. You’ve been warned. For more information or to download the app for your mobile device, go here. Remember to dial the "+1″ country code prefix for U.S./Canada calls.

Originally published: Monday, April 24, 2017  Updated: Monday, February 25, 2019


Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.



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.
 



Some Recent Nerd Vittles Articles of Interest…

Adding SIP URI Dialing to Asterisk for Free Worldwide Calling



Since giving up on Google Voice, we’ve been extolling the virtues of SIP URI dialing which gives you unlimited free calls to anyone else in the world that happens to have their own SIP URI address. SIP URIs look very much like email addresses except they’re used to share phone conversations instead of email messages. And, as we’ve mentioned previously, if everyone in the world had their own SIP URI, paying for phone calls would become a thing of the past. We hope you’ll join us in making that happen. As a fallback, give our $50 credit at Skyetel a try.

One of the drawbacks of Asterisk® PBXs using the FreePBX® GUI has been the inability to place outbound SIP URI calls from SIP phones registered as extensions on the PBX. Today we first want to address that shortcoming. Our SIP URI dialing solution for Asterisk should work with any FreePBX-based implementation including Incredible PBX® and Issabel as well as on Raspberry Pi platforms. We’ll wrap things up by providing some tips on obtaining and deploying your own SIP URI at little or no cost and pointing you to some excellent resources that facilitate calling millions of SIP phones around the world at zero cost. All you need is an Internet connection, and we’ll point you to a terrific softphone to begin your adventure.

Let’s begin by examining why SIP URI dialing is a problem with FreePBX. The reason is pretty simple. FreePBX interprets dial strings by matching them against some rules to determine whether you’re making an internal call or a call outside your PBX. It matches internal calls against a list of available internal extensions. External calls are matched against rules defined in your outbound routes which are associated with trunks. Since SIP URI calls don’t match any extension or outbound route, the caller receives a congestion tone.

The traditional workaround has been to define a custom extension using the FreePBX GUI which points to a SIP URI. Then the user can dial the custom extension, and the call will be routed to the defined SIP URI. These custom extensions also can be defined in extensions_custom.conf within the from-internal-custom context. For example, the following dialplan code would let users dial 411 to reach AT&T’s Toll-Free Directory Assistance: exten => 411,1,18005551212@switch.starcompartners.com.

But there’s a better way. Wouldn’t it be nice to be able to dial any SIP URI from a softphone or to store SIP URI addresses in the phonebook of your SIP phone?1 Well, now you can. Before we actually put the dialplan code in place, let us explain how this will work. First, FreePBX still needs to be able to distinguish a SIP URI call from a "regular call." The reason this gets tricky is because Asterisk typically throws away the destination hostname when you place a call. For example, calls to 8005551212 and 8005551212@sip2sip.info are processed by Asterisk in exactly the same way, i.e. dropping the host address before dialing.

Using the new dialplan code in the next section, here’s how calls will be processed:

User dials                    Asterisk processes call as
------------------------      ---------------------------------------------
701                           internal call to local extension 701
4045551212                    external call using NXXNXXXXXX outbound route
2233435945@sip2sip.info       SIP URI call to Lenny by acct at sip2sip.info
lennybgood@sip2sip.info       SIP URI call to alias lennybgood@sip2sip.info 

Cautionary Notes: Our code should work fine with any Asterisk 13 and FreePBX 13 or Incredible PBX deployment on any Linux platform; however, with servers other than Incredible PBX, make sure you have added the following entries to sip_general_custom.conf, or you can configure them in the GUI by making the changes in Settings -> Asterisk SIP Settings -> Chan SIP Settings:

srvlookup=yes
allowguest=yes

You also need to test a traditional outbound call (e.g. 8005551212) immediately after you finish the install procedure. Monitor the Asterisk CLI (asterisk -rvvvvvvvvvv) and observe the first few lines of the log after you place a call. The second line will show SIPDOMAIN which should be either the FQDN of your server or an IP address depending upon how you registered your softphone extension. The first line should display the MyDomain variable. If it is empty or doesn’t match the SIPDOMAIN entry, the outbound call will fail. To fix it, add an entry to the Asterisk database from the Asterisk CLI using syntax like the following: database put MyDomain FQDN 10.0.0.11 or database put MyDomain FQDN sip.me.com where 10.0.0.11 or sip.me.com matches the SIPDOMAIN entry shown on the second line. Then retry your outbound call, and it should complete successfully. We’ve tested this back to the early Asterisk 11 days with FreePBX 2.11 without any problems. If your calls still fail, then you will probably need to remove the new code from your platform until you upgrade to a more current version of Asterisk and FreePBX. The code hasn’t been tested with FreePBX 14 and 15.

Finally, you may want to manually set the CallerID for your outgoing SIP URI calls. From the Asterisk CLI, issue a command for every extension from which you will be placing SIP URI calls, e.g. extension 701 syntax: database put 701 user_sipname "Nerd Uno"

Enabling SIP URI Dialing with FreePBX

To enable SIP URI dialing from phones registered with your Asterisk PBX, we’ll modify the dialplan in order to detect SIP URI dial strings entered into a softphone or retrieved from a phonebook associated with almost any SIP phone. When a SIP URI dial string is detected, we’ll send the call out as requested rather than passing the call through the outbound routes and trunks associated with your PBX. All of this dialplan code is open source and is licensed pursuant to the GPL2 license.

SECURITY ALERT: Never use the SIP URI MOD on a server with a publicly-exposed SIP port as it is possible for some nefarious individual to spoof your FQDN in the headers of a SIP packet and easily gain outbound calling access using your server’s trunk credentials.

FEB. 21 UPDATE: There was a bug in the original code which caused some internal calls to fail including calls to a DISA extension. Simply install the application again, and it will overwrite the previous version.

MAR. 5 UPDATE: A bug was discovered in previous releases that treated 911 and 933 calls as internal calls when, in fact, they should have been routed out using your outbound trunks. Simply install the application again, and it will overwrite the previous version.

MAR. 13 ALERT: This software is not compatible with the Debian, Raspbian, and Ubuntu platforms.

To begin or update your installation, log in to your PBX as root using SSH or Putty and issue these commands:

cd /tmp
wget http://incrediblepbx.com/sipuri-mod.tar.gz
tar zxvf sipuri-mod.tar.gz
rm -f sipuri-mod.tar.gz
./install-sip-uri-mod.sh

Obtaining Your Own SIP URI

There are a number of ways to obtain your own SIP URI. Perhaps the easiest is to set up the open Incredible PBX cloud platform that we introduced several weeks ago. Then you can create as many SIP URIs as you like, and they can be used to perform any task that’s available with Asterisk. If you’re not quite ready to make that leap, a free or almost free SIP URI is available from the following sources. VoIP.ms provides a SIP URI for every subaccount you create. Just set up an internal extension number for the subaccount, and that becomes a SIP URI to connect back to your registered server or SIP phone. In the alternative, VoIP.ms will also provide you with a free iNUM DID which can be reached at the following IP address: 81.201.82.50. CallCentric provides a SIP URI matching your account number which can be reached @in.callcentric.com. CallCentric will also provide you with a free iNUM DID which can be reached at the following IP address: 81.201.82.50. LocalPhone provides the same two options as CallCentric: you can be reached by your account number @localphone.com. Or the LocalPhone-assigned iNUM DID can be reached @81.201.82.50. Then there’s pbxes.org. Your account name can be used for SIP URI access @pbxes.org. And, of course, if you’re a 3CX user, you can set up a SIP URI for each extension on your PBX. Just navigate to the Options tab of the desired extension(s) and enter a unique SIP ID for each extension. The SIP URI becomes SIPID@YOUR-3CX-FQDN. SIP URI calls to 3CX Clients on smartphones are also free! This list is not exhaustive. There are now more than 2,000 VoIP networks that support SIP URI access. Using a SIP URI dialing prefix, call any of the referenced networks @sipbbroker.com.2

Choosing a SIP Phone or Softphone

You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum. For today we’ll get you started with one of our favorite (free) softphones, YateClient. It’s available for almost all desktop platforms. Download YateClient from here. Run YateClient once you’ve installed it and enter the credentials for an extension on your PBX. You’ll need the IP address of your server plus your extension number and its password. Fill in the Yate Client template using the IP address of your PBX as well as your extension credentials. Click OK to save your entries.

Once the Yate softphone shows that it is registered, try a test call to Lenny using one of the following SIP URIs: 2233435945@sip2sip.info or 883510001198938@81.201.82.50. Better yet, try out a few Incredible PBX samples from the public server we previously deployed:

Yahoo News Headlines    - news@demo.nerdvittles.com
Weather by Zip Code     - weather@demo.nerdvittles.com
Directory Assistance    - information@demo.nerdvittles.com
Lenny for Telemarketers - lenny@demo.nerdvittles.com

Originally published: Monday, February 11, 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.
 



  1. Special thanks to Olivier Adler and voip-info.org for their early work on SIP URI dialing with Asterisk. []
  2. Some of our links refer users to sites or service providers when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from these providers to help cover the costs of our blog. We never recommend particular products solely to generate commissions. []

Introducing Skyetel: A VoIP Provider for All Seasons

Having been around the block more times than we can remember, suffice it to say it takes a lot to get us excited about a VoIP provider. Let us tick off some criteria to even get our attention: terrific pricing, failsafe reliability, and first class performance. So just imagine our excitement to discover that an early follower of Nerd Vittles now provides one of the most compelling VoIP services we’ve ever tested with triple redundancy in multiple data centers. And Skyetel now has added what, for some, was the most important piece: support for VoIP servers with dynamic IP addresses. While it’s still beta code, it’s easy to use and reliable. There’s yet another hidden benefit. Incredible PBX coupled with Skyetel makes a perfect platform for redundant servers. We’ll cover it in a future article, but here’s the basic design.

Let’s sweeten the pot a bit more. We were looking for a service provider that could offer a compelling price for the hobbyist and home user while also having the depth to provide millions of minutes to organizations and resellers that actually have such a need. Skyetel now offers Nerd Vittles readers two special offers. First, you can claim a $10 credit for your new account simply by opening a ticket once you sign up. Once you have kicked the tires and are satisfied with the service, you won’t want to miss the Nerd Vittles BOGO offer. Skyetel will match your original deposit up to $250. Deposit $50 and Skyetel will double it. Or plan ahead with a $250 deposit and Skyetel will still double it. That translates into $500 of half-price VoIP service! Once you have funded your account with your money, Skyetel will provide free porting of your DIDs for the first 60 days after you open your account plus a 10% reduction in your current origination rate and DID costs by presenting your last month’s bill.1 Effective 10/1/2023, $25/month minimum spend required. For resellers and high volume users, document your requirements on your Nerd Vittles signup form and let us put you in touch with someone at Skyetel that will make you a deal you can’t refuse. And what does Nerd Vittles get out of this? Glad you asked. We’re delighted to have Skyetel as a platinum sponsor to keep the lights burning and the deals flowing for another decade of articles and open source offerings for our dedicated followers.

Original Skyetel DepositSkyetel Deposit MatchAvailable SIP Service $'s
$20$20$40
$50$50$100
$100$100$200
$200$200$400
$250$250$500

We want to also address the elephant in the room. Some have asked about our relationship with Vitelity, a long time sponsor of Nerd Vittles and our open source projects. They’re alive and well. However, the company has gone through several acquisitions in the past few years, and their focus now has shifted more to the reseller and wholesale market. ALL EXISTING VITELITY CUSTOMERS ARE UNAFFECTED BY THIS CHANGE IN DIRECTION. And we are more than happy to put new resellers and wholesalers in touch with someone at Vitelity that can address your requirements. The good news is that you’ll now have two companies to compare while new home users and small businesses have a viable alternative moving forward.

Skyetel’s State-of-the-Art Network Design

Because Skyetel’s system architecture is radically different from most other VoIP providers, we wanted to spend a minute documenting their setup. Typically, a VoIP provider may offer a failover server in case their primary server fails. But all calls flow through the primary server unless there is a system failure. As we noted previously, Skyetel’s current setup includes three redundant data centers, all of which receive incoming calls while being firewalled from each other. Once you place or receive a call from the Skyetel network, their data center is completely removed from the audio path of the call which flows directly between your server and the outside party. Thus, even if the data center experienced a total system failure in the middle of your call, neither you nor the other party would ever know it. This design also eliminates the potential of a man-in-the-middle attack from your VoIP provider’s server.

Skyetel Pricing Overview

This summary is not intended to be an exhaustive listing of all Skyetel services. Follow this link for a complete summary of fees and services. Traditional DIDs are $1 per month. Toll free numbers an additional 20¢ per month. Outbound conversational calls are $0.012 per minute. DIDs can be SMS/MMS enabled for 10¢ per month. E911 service is $1.50 per month. Incoming conversational calls are a penny a minute. CallerID lookups are $0.004 per call. Voicemail transcription is available for 10¢ per message.

Signing Up for Skyetel Service

So here’s the drill to 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 your free $10 credit to kick the tires. You cannot port in numbers at no cost until you actually fund your account out of your own pocket. Once you have funded your account, open another ticket for the BOGO credit for your account by referencing the Nerd Vittles special offer. You then can initiate your free number porting requests on the portal and request a credit for the porting fees. BOGO credit is limited to one per person/company/address/location. If you want to take advantage of the 10% discount on your current service, attach a copy of your last month’s bill. See footnote 1 for the fine print. If you have high call volume requirements, document these in your Prequalification Form, and we will be in touch. Easy Peasy!

For those that may be concerned that one day, after your credit expires, you could be paying a penny a minute for phone calls, let me provide a little Ma Bell history lesson for you. When my roommate and I were in law school, our typical phone bill often exceeded $200 a month because we both had girlfriends a couple hundred miles up the road. In today’s dollars, that phone bill translates into roughly $1,200 a month. That would have been 120,000 minutes a month at a penny a minute in today’s dollars. So, yes, VoIP is having a profound influence on the AT&T and Verizon Bell Sisters.

Skyetel Endpoint Group Configuration

Unlike many VoIP providers, 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. A typical setup for use with Incredible PBX®, Asterisk®, or FreePBX® would look like the following:

  • Name: MyPBX
  • Priority: 1
  • IP Address: PBX-Public-IP-Address
  • Port: 5060
  • Protocol: UDP
  • Description: server1.incrediblepbx.com

Skyetel DID Configuration

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.

Incredible PBX Firewall Setup for Skyetel

The Travelin’ Man 3 firewall included with all Incredible PBX platforms limits access to your server based upon whitelisted IP addresses of outside providers and users. In order to receive calls from the multiple Skyetel data centers, the following entries need to be included in the whitelist of your PBX. For new installs of Incredible PBX 13-13 for CentOS, the entries already are included. Otherwise, issue the following commands from the Linux CLI and choose the 0 option using the add-ip utility in /root:

  • /root/add-ip Skyetel-NW 52.41.52.34
  • /root/add-ip Skyetel-SW 52.8.201.128
  • /root/add-ip Skyetel-NE 52.60.138.31
  • /root/add-ip Skyetel-SE 50.17.48.216
  • /root/add-ip Skyetel-EU 35.156.192.164

NOTE: If your PBX is sitting behind a NAT-based router, then you will also need to forward UDP port 5060 from your router to the internal IP address of your PBX. Otherwise, incoming calls from Skyetel will fail. You also may need to add a NAT=yes entry to each of the Skyetel trunk configurations using the GUI. The telltale sign that the NAT entry is required will be incoming calls with one-way or no audio.

Incredible PBX Trunk Setups for Skyetel

Because Skyetel uses multiple data centers without trunk registrations, you’ll actually need to configure 6 separate Skyetel trunks in the Incredible PBX GUI. The same setup applies for those using generic FreePBX aggregations. We’ve created a script to create all of the trunks for you. Just issue the following commands. The last command assures that you don’t accidentally run the script a second time which would cause all sorts of issues. Feel free to review the code if you want to learn how to create trunks in FreePBX from the command line.

cd /root
wget http://incrediblepbx.com/add-skyetel
chmod +x add-skyetel
# uncomment next line if your incoming calls all have 10-digit numbers
# sed -i 's|from-trunk|from-pstn-e164-us|' add-skyetel
./add-skyetel
chmod -x add-skyetel

Incredible PBX Inbound Routing for Skyetel

Next we need to tell your PBX how to route incoming calls from Skyetel. Using a browser, log into the IP address of your PBX using your admin credentials. Because there is no trunk registration with Skyetel trunks, you will need to create an Inbound Route for every Skyetel DID. You cannot rely upon a Default inbound route because FreePBX treats the calls as blocked anonymous calls without an Inbound Route pointing to the 11-digit number of each Skyetel DID. From the GUI, choose Connectivity -> Inbound Routes -> Add Inbound Route. For both the Description and DID fields, enter the 11-digit phone number beginning with a 1. Set the Destination for the incoming DID as desired and click Submit. Reload the Dialplan when prompted. Place a test call to each of your DIDs after configuring the Inbound Routes.

Incredible PBX Outbound Routing to Skyetel

If Skyetel will be your primary provider, you can use both 10-digit and 11-digit dialing to process outbound calls through your Skyetel account. From the GUI, choose Connectivity -> Outbound Routes -> Add Outbound Route. For the setup, we recommend the following using the CallerID Number you wish to associate with your outbound calls through Skyetel:

Enter the Dial Patterns under the Dial Patterns tab before saving your outbound route. Here’s what you would enter for 10-digit and 11-digit dialing. If you want to require a dialing prefix to use the Skyetel Outbound Route, enter it in the Prefix field for both dial strings.

Audio Issues with Skyetel

If you experience one-way or no audio on some calls, make sure you have filled in the NAT Settings section in the GUI under Settings -> Asterisk SIP Settings -> General. In addition to adding your external and internal IP addresses there, be sure to add your external IP address in /etc/asterisk/sip_general_custom.conf like the following example and restart Asterisk:

externip=xxx.xxx.xxx.xxx

If you’re using PJSIP trunks or extensions on your PBX, implement this fix as well.

Receiving SMS Messages Through Skyetel

Most Skyetel DIDs support SMS messaging. Once you have purchased one or more DIDs, you can edit each number and, under the SMS &MMS tab, you can redirect incoming SMS messages to an email or SMS destination of your choice using the following example:



Sending SMS Messages Through Skyetel

We’ve created a simple script that will let you send SMS messages from the Linux CLI using your Skyetel DIDs. In order to send SMS messages, you first will need to create a SID key and password in the Skyetel portal. From the Settings icon, choose API Keys -> Create. Once the credentials appear, copy both your SID and Password. Then click SAVE.

Next, from the Linux CLI, issue the following commands to download the sms-skyetel script into in your /root folder. Then edit the file and insert your SID, secret, and DID credentials in the fields at the top of the script. Save the file, and you’re all set.

cd /root
wget http://incrediblepbx.com/sms-skyetel
chmod +x sms-skyetel
nano -w sms-skyetel

To send an SMS message, use the following syntax where 18005551212 is the 11-digit SMS destination: sms-skyetel 18005551212 "Some message"

SMS and MMS Messaging with Postcards

Skyetel now has released a terrific, open source Docker app, Postcards, that lets you build an SMS and MMS messaging platform for your entire organization. Suffice it to say, anything you ever wanted to do with SMS and MMS messaging, you can do with Postcards. We won’t repeat Skyetel’s excellent tutorial, but you certainly need to visit their site and take Postcards for a spin.

NEW: Skyetel Support for Dynamic IP Addresses

You asked for it, and Skyetel has delivered. For Nerd Vittles users running servers with dynamic IP addresses, Skyetel now provides support for your platform. Log into your server as root and cd /usr/src. Then review this tutorial which describes the steps to put the pieces in place. Be advised that this is beta software at this juncture. If you run into issues, please post your questions on the PIAF Forum. Here are the actual steps:

(1) Log in to your Skyetel portal and Add a New Endpoint Group for your server giving it the name and current public IP address of your server.

(2) While still logged in, tap the Gear icon to open Settings dialog and choose API Keys tab.

(3) Add a new API key and write down your new SID and SID password.

(4) If your server is behind a router or firewall, log into that device and map UDP 5060 and UDP 10000-20000 to the private LAN address of your server.

NOTE: If your server is on the Debian, Ubuntu, or Raspbian platform, substitute the following command for the first two yum commands in step #5 below:

apt-get -y install coreutils curl git jq

(5) Log into your server and issue the following commands to install the EndPoint Updater:

yum -y install coreutils curl git epel-release
yum -y --enablerepo=epel install jq
cd /usr/src
git clone https://bitbucket.org/skyetel/ip-endpoint-group-update.git
cd ip-endpoint-group-update
./ip-update-endpointgroup.sh

(6) Fill in your credentials when prompted, and the cron script will be installed to keep your server’s dynamic IP address registered with Skyetel.

Introducing Skyetel’s New Fax Platform

Every time we read an article predicting the demise of fax technology, we have to chuckle. We’ve been reading the articles for about 30 years now, and fax still is the goto solution for many organizations. Can you spell HIPPA? Finally, Skyetel has dipped its toes in the fax waters by offering an easy-to-use fax solution for receipt of traditional and T.38 faxes. Simply purchase a Skyetel DID and configure it for vFax routing. Enter an email address for delivery of the faxes, and you’re done.


Sending faxes from the Skyetel portal still is on the drawing boards, but it’s coming. In the meantime, Incredible Fax™ which is bundled with all Incredible PBX® platforms will let you send faxes ’til the cows come home with our easy-to-use Hylafax/AvantFax implementation.

Implementing the New Spam Call Filter

One of the most often requested features for any PBX is spam call filtering. Skyetel takes it to the next level by dealing with the spammers before the calls ever reach your PBX. For each of your Skyetel phone numbers, click on the Features tab and set the Spam Call Filter as desired.

Recording and Transcribing Skyetel Calls

As with spam call filtering, recording and/or transcribing Skyetel calls is only a click away. For each of your Skyetel phone numbers, click on the Features tab and set the option desired for Recording and/or Transcribing calls. Recordings and Transcriptions can be managed from your Skyetel Dashboard. Storage is free for up to 30 days, after which they are deleted.

Skyetel Monitoring of Endpoint Health

In addition to monitoring and reporting the health of all Skyetel services in your web portal, this latest addition allows you to configure Skyetel to not only monitor the State of every registered endpoint but also its Health with realtime metrics of the Latency, Packet Loss, and Jitter of each of your endpoints. Simply check the Network QOS options desired.

Skyetel Expansion for Canadian Users


Here’s some great news for our Canadian friends. Skyetel has been listening!

  • Porting to Skyetel in Canada now is significantly easier and faster
  • Awesome reductions in audio round trip times
  • Epic reductions in time-to-deliver
  • Faster response times to technical issues (and fewer of them!)
  • Audio for Canadian calls will now originate from Canadian data centers
  • SMS and MMS available on Canadian ported numbers

Originally published: Thursday, November 1, 2018  Updated: Wednesday, June 12, 2019


Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.



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. In the unlikely event that Skyetel cannot provide a 10% reduction in your current origination rate and/or DID costs, Skyetel will give you an additional $50 credit to use with the Skyetel service. []

Skyetel Smorgasborg: SMS Blasting, SMS Dictator, and more




Just in time for Santa, we’ve got a great treat for those of you that have taken advantage of the Nerd Vittles special offer from Skyetel which gets you a $50 credit on their powerful VoIP platform. Today we’re adding not one, but three, SMS messaging utilities to the Incredible PBX UC platform. Effective 10/1/2023, $25/month minimum spend required. In addition to a command line utility to send SMS messages, we’re also introducing SMS Message Blasting which lets you send an SMS message to as many recipients as you would like. It’s perfect for sports team and community group messaging. To round out the trifecta, we’ve updated our SMS Dictator utility by integrating Skyetel messaging with IBM’s powerful voice recognition software.1 Simply dial S-M-S (767) from any extension on your PBX and dictate an SMS message to send to a recipient of your choice. Gone are the days of wrestling with Google’s ever-changing voice recognition platform. Good riddance!

To get started, you’ll need to have an IBM Watson account with an APIkey for their Speech-to-Text (STT) engine. Next, you will need a Skyetel SMS-enabled DID. Before we install today’s SMS scripts, it should be noted that SMS messages must be sent from the PBX registered as the Skyetel Endpoint Group for the SMS-enabled DID specified in the Skyetel SMS scripts. So let’s begin with the configuration steps to put all the pieces in place.

Getting Started with IBM Watson STT Service

We’ve created a separate tutorial to walk you through obtaining and configuring your IBM Watson credentials. Start there.

Now let’s get IBM’s Speech to Text service activated. Log back in to the IBM Cloud. Click on the (upper left) Menu icon and select Dashboard. Click on the Speech to Text app. Choose a Region to deploy in, choose your Organization from the pull-down menu, and select STT as your Space. Choose the Standard Pricing Plan or LITE Plan. Then click Create. When Speech to Text Portal opens, click the Service Credentials tab. In the Actions column, click View Credentials and copy down your STT APIkey. Then logout of IBM Watson.

Getting Started with Skyetel Messaging

If you haven’t already signed up for a Skyetel account, read our tutorial and take advantage of the $50 coupon for free service. Sign up for a DID and activate the SMS feature for your number. Create an Endpoint Group with the public IP address of your PBX. Then edit your phone number and link it to the Endpoint Group of your server. If you want to forward incoming SMS messages to either an email address or to your smartphone’s messaging service, configure it under the SMS & MMS tab. Finally, click on the settings icon beside your account name in the upper right corner of the Skyetel portal and then click the API Keys tab. Click the Create button and copy down your SID and SECRET for Skyetel’s API service. This secret is not retrievable once you close the window so put the credentials in a safe place for subsequent use. Then logout of the Skyetel portal.

Installing the SMS Components on Your PBX

There are three separate applications which we will install on your PBX: (1) a stand-alone utility that lets you send SMS messages from the Linux CLI by entering a recipients 11-digit phone number and an SMS message surrounded by quotes, (2) an SMS message blasting utility that lets you send a previously prepared SMS message to a group of recipients whose 11-digit SMS numbers have been entered into a text file, and (3) the SMS Dictator application which lets you pick up any phone on your PBX and dial S-M-S (767) to dictate a message and send it to a recipient whose number you’ve key in from your phone. For those not residing in North America, the number of phone number digits can easily be changed in all of the scripts. After we install the three applications, we’ll edit each of the scripts to insert your IBM STT and Skyetel API credentials. Then you’re ready to start messaging.

First, let’s install the stand-alone and message blasting SMS utilities. Log into your server as root and issue the following commands:

cd /root
mkdir sms-skyetel
cd sms-skyetel
wget http://incrediblepbx.com/smsblast-skyetel.tgz
tar zxvf smsblast-skyetel.tgz
rm -f smsblast-skyetel.tgz

Next, let’s install the SMS Dictator application while still logged into your server:

cd /var/lib/asterisk/agi-bin
wget http://incrediblepbx.com/sms-767-skyetel.tgz
tar zxvf sms-767-skyetel.tgz
rm -f sms-767-skyetel.tgz
./install-sms767-dialplan.sh

Configuring the Skyetel SMS Components

While still positioned in the agi-bin directory, edit smsgen.sh. Insert apikey as your API_USERNAME and your actual STT APIkey as API_PASSWORD in the fields provided. Insert your Skyetel SID, SECRET, and 11-digit DID in the fields provided. Then save the file.

Next, change directories to /root/sms-skyetel and edit BOTH sms-skyetel and smsblast and insert your Skyetel credentials and DID in the fields provided at the top of both files.

Finally, when you’re ready to use the message blasting application (smsblast), first insert your SMS message in the smsmsg.txt file. Then insert the list of SMS numbers in smslist.txt.

Testing the Skyetel SMS Components

To try out the SMS Dictator application, dial S-M-S (767) from a phone connected to your PBX. When prompted, enter the 11-digit number of the SMS recipient. When prompted, dictate the message to be sent and press #.

To try out the stand-alone SMS application, navigate to /root/sms-skyetel and issue the following command using the 11-digit number of the SMS recipient followed by a space and an SMS message to be sent surrounded by quotes: ./sms-skyetel 18005551212 "Howdy."

To try out the message blasting SMS application, navigate to /root/sms-skyetel. Enter the message to be sent in smsmsg.txt and enter the list of SMS numbers in smslist.txt. Kick off the message blast by entering the command: ./smsblast.

Originally published: Monday, December 10, 2018


Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.



Need help with Asterisk? Join our new MeWe Support Site.


 

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. Skyetel outbound SMS messages are billed at 1¢/message plus a monthly SMS surcharge of 10¢ per SMS-enabled DID. With IBM’s STT service, users have a choice of the LITE tier providing 100 minutes a month of free transcription or the STANDARD tier providing unlimited message transcription at a cost of 2¢/minute. []

VoIP 101: Developing a Cost-Effective SIP Strategy

In the lead up to the demise of Google Voice XMPP service next week, we wanted to offer what we have found to be a cost-effective SIP strategy which takes advantage of the best of all worlds. We would divide SIP offerings into five broad categories: business-class unlimited SIP trunks, Old Faithful SIP providers, Mom-and-Pop SIP services, dirt-cheap termination services, and Gee Whiz SIP providers. As we have said many times, the beauty of setting up an Asterisk® PBX such as Incredible PBX® is you don’t have to put all your VoIP eggs in one basket. In our particular case, that has included a mix of Google Voice trunks plus all five of the SIP categories above. Today we want to document why we’ve personally made the selections we’ve made and hope that it provides a roadmap for your own VoIP setup while encouraging you to venture out of your safe zone and try some new VoIP options.

The all-you-can-eat business plans, which we previously have covered, make little sense for most home and small business users. Then there are the rock-solid, long term pay-as-you-go providers such as Vitelity and CallCentric that make perfect sense as your primary DID and SIP provider. While they may not always be the cheapest VoIP providers, the tradeoff is dependability and long-term reliability for your VoIP platform. In the case of Vitelity, it turns out the Nerd Vittles DID special (detailed below) from our Platinum Sponsor is perhaps one of the best VoIP deals on the planet.

The third category of SIP providers and our personal favorite is what we would call the mom-and-pop providers. These are typically one or two-person operations that offer incredible deals on all-you-can-eat VoIP plans for home users. Included in this category are Vestalink (available to existing customers only), Future-Nine and CircleNet. VestaLink originally began as OBiVoice and morphed over trademark issues. While the service is no longer available to new customers, it remains the best bargain at $72 for two years of unlimited inbound and outbound residential calling services. A close second goes to Future-Nine and their "Future 5 Grey" plan which provides 1,500 inbound and 1,500 outbound minutes a month for only $5. You can sign up here. Be sure to read the Terms of Services carefully, especially item #18. The New Kid on the Block is CircleNet. In addition to very attractive pay-by-the-minute offerings of $.005 per minute to most of the U.S. and Canada, they also have an $8 a month all-you-can-eat plan for residential customers that includes a very reasonable 5,000 minutes a month for calls to the following countries: United States, Canada, Australia, Bangladesh, Belgium, Brazil, Chile, Cyprus, Denmark, Finland, France, Germany, Greece , Guam, Hungary, India,Ireland, Italy, Japan, Latvia, Mexico, Netherlands, New Zealand, Norway, Poland, Puerto Rico, Singapore, Spain, Sweden, Taiwan, Thailand, United Kingdom, and Vatican City. Just let them know that you plan to use it with an Asterisk-based PBX. CircleNet also is offering Nerd Vittles readers a free month of the $8/month service to kick the tires. Simply send an email to sales@circlenet.us with your valid email address to take advantage of the offer. One free trial per customer/email address. CircleNet also offers a $15 a month business plan with even more minutes.

A fourth class of VoIP providers is the dirt-cheap termination services including Anveo Direct, TelecomsXchange, V1VoIP and the Betamax companies for low-cost international calling. These providers make terrific additions for supplementing your other VoIP services. TelecomsXchange is our personal favorite because of the special deal they have extended to Incredible PBX users. You get access to 300 VoIP wholesalers and can read about their services in this Nerd Vittles article. V1VoIP also has some terrific deals with 15¢/mo. DIDs from 13,000 Rate Centers and incoming and outgoing U.S. call pricing as low as $.003 per minute (not a typo!). Anveo Direct was perhaps the first provider to offer wholesale pricing to consumers, and they remain a terrific service both for DID and origination services with T.38 fax support as well as many of the lowest cost SIP terminations worldwide featuring user-configurable least-cost routing. Check out their pricing and rates here.

Finally, there are the SIP providers such as VoIP.ms that offer a rich collection of special features that you won’t find in many places and certainly not under the same roof. These features include SMS messaging, SIP URI proxying and iNUM for free worldwide calling, and fax support. Every one of these features is free when you sign up for an account at VoIP.ms. We encourage you to take advantage of these little known free services to enhance your PBX.

Putting It All Together. Now that we’ve covered the options, let’s go over how we would actually implement this. For the inbound trunk and primary DID, we’d recommend a SIP trunk from either Vitelity, VoIP.ms, or CallCentric. If you have multiple, simultaneous inbound calls, then the Nerd Vittles Vitelity special below can’t be beat because it provides four call paths. In addition, you get SMS support on the same trunk. Many people now assume your primary number supports SMS. We actually get dozens of unsolicited SMS messages on our home number from schools, churches, and political groups. If incoming call volume isn’t an issue, then VoIP.ms and CallCentric also offer a free iNUM number for your account. And VoIP.ms throws in a SIP URI as well.

For outbound calling for home and SOHO deployments, we recommend at least one of the mom-and-pop, all-you-can-eat providers: Future-Nine or CircleNet. If international calling is a requirement, you can’t beat the CircleNet offering. In addition to using your primary incoming provider, we also recommend you set up SIP accounts with a couple of the dirt-cheap termination providers. These don’t cost you anything other than a modest deposit unless you actually use them to place calls. And, when your primary outbound service has an outage, your PBX will never miss a beat.

The icing on the cake always has been several Google Voice trunks which work well for IVRs, Stealth AutoAttendants with DISA support, and faxing. While this may change with the demise of XMPP support, it appears that Bill Simon’s SIP Gateway to Google Voice will live on. With the Nerd Vittles sign-up link, you can migrate your existing Google Voice XMPP connections to the Simonics gateway for $4.99 each should the need arise. Enjoy!

Originally published: Monday, June 11, 2018


CircleNet SIP Setup for FreePBX/IncrediblePBX/VitalPBX/Issabel:

username=acct-id
type=friend
trustrpid=yes
sendrpid=yes
secret=acct-pword
qualify=yes
nat=yes
insecure=port,invite
host=sip.circlenet.biz
fromuser=acct-id
context=from-trunk
disallow=all
allow=ulaw

Registration String: acct-id:acct-pword@sip.circlenet.biz:5060/did-num

Future-Nine SIP Setup for FreePBX/IncrediblePBX/VitalPBX/Issabel:

username=acct-num
type=friend
trustrpid=yes
sendrpid=yes
secret=acct-pword
qualify=yes
nat=yes
insecure=port,invite
host=incoming.future-nine.com
fromuser=acct-num
context=from-trunk
canreinvite=no
disallow=all
allow=ulaw

Registration String: acct-num:acct-pword@incoming.future-nine.com/acct-num


Need help with Asterisk? Visit the PIAF 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.
 



Some Recent Nerd Vittles Articles of Interest…

Creating an OBi200 Google Voice Trunk to Use with Asterisk


Since Asterisk® will no longer be able to "talk" to Google Voice after June 17, we promised to hold our nose and document how to salvage your Google Voice trunks. Our exercise for today is to show you how to deploy an OBi 200-series device which can speak the new Google Voice language and use it as a traditional SIP bridge between Google Voice’s proprietary SIP platform and your Asterisk server. We will skip the editorializing on why Google is making a terrible mistake by discarding XMPP and forcing users to a proprietary solution necessitating a hardware purchase without first offering an open standards solution as Google’s Community Manager promised here. Promises, of course, don’t keep your phones ringing. For the whole story, see our article from last Saturday. For today, you’ll need to shell out $50 for an OBi 200 device. Once you have it in hand, feel free to read on and we’ll get you back in business. For security reasons, it should be noted that today’s setup assumes you are running an Incredible PBX® server and OBi device locally behind a NAT-based router. This will work equally well with the Incredible PBX-enhanced versions of Issabel and VitalPBX. We’ll leave it to the FreePBX® folks to figure out a solution for their proprietary distro.

Everything we’re covering below will work just as well using any of the OBi 200-series devices. We’ve simply chosen to use an OBi202 in our examples today because it supports an extra phone port. But an OBi200 works just as well if you will only be deploying Google Voice trunks (up to 3 and perhaps more) for your PBX. They retail for approximately $50 and are readily available at Amazon through the link in the right column which also provides a few shekels for Nerd Vittles to keep the lights on. As mentioned last week, Obihai crippled the OBi 110-series devices which will no longer work with the new Google Voice setup. Such a fine company that we once praised for producing our Device of the Year. And don’t worry. If you ever visit their forum, you can expect a cheery reception from the Obihai forum moderator. Here’s the response we got1 when raising concerns about the demise of Google Voice XMPP:



Registering Your OBi2x Device with OBiTALK

A Quick Start Guide accompanies your OBi hardware. Following along in the tutorial will get your OBi set up using a free (so far) OBiTALK account. When you get to Step 5, you’ll be ready to set up your Google Voice account by clicking the Google Voice Set-Up button.

Before you begin the Google Voice setup, we strongly recommend that you plug a POTS phone into your OBi device and dial ***6 to update your firmware to the latest release. Depending upon where you purchased your device, it may or may not have the latest firmware which is required to communicate with Google Voice on or after June 17.

We also recommend that you dial ***1 and obtain the DHCP-assigned IP address for your OBi. You’ll need this in a few minutes. And, while you’re at it, be sure to set the OBi up behind a NAT-based router to protect it from intrusion. Once someone gains access to your OBi, they’ve essentially got the keys to your telecom castle. So always deploy an OBi behind a hardware-based firewall that is on the same private LAN as your Asterisk PBX. Finally, on your router, be sure to reserve the DHCP-assigned IP address of your OBi for permanent use by the OBi hardware. Otherwise, the IP address of your OBi may change, and this will break the SIP gateway connection to your Asterisk server.

Finally, a word about the new OBi setup. All of your settings are now stored and managed in the OBiTALK cloud. Obihai then pushes the configuration to your OBi device. To put it charitably, this usually works but sometimes it doesn’t, and you end up with a quirky OBi setup that looks correct in the cloud but simply doesn’t work. We’ve found the simplest solution is to unplug the device and then restart it. Then check all of your cloud-based settings when the OBi device comes back to life to be sure none of your settings disappeared. Sometimes they do! In the old days, you had the option of configuring your OBi device locally; however, Obihai (now Polycom) has disabled that functionality with the new Google Voice setup presumably to disguise what they are doing under the covers to connect to Google.

Configuring a Google Voice Trunk on OBi200

To give credit where credit is due, configuring a Google Voice trunk on the OBi 200-series devices is dead simple. Login into your OBiTALK account, click on your OBi device, and then click the Google Voice Set-Up button.

Enter your Google Voice credentials when prompted, give Obihai permission to control your Google Voice account, and you’re done. Within a few seconds, the connections dialog box should show Google Voice connected on service provider SP1.

If you haven’t already done so, plug a POTS phone into your OBi device and place a call to somebody by dialing a 10-digit number. Then use another phone and call the Google Voice number you assigned to your OBi device. The POTS phone should ring. Don’t continue until you get these calls working in both directions. You’d be wasting your time.

Now we need to adjust the destination for incoming calls to your OBi device and redirect them from the POTS phone to the SP3 trunk we’ll be using to connect to your Asterisk server. We’ll leave SP2 unoccupied in case you wish to add another Google Voice trunk down the road.

To make this change, click the OBi Expert Configuration button at the bottom of the Device Configuration window. Then click OK to confirm that you know what you’re doing. Next click the Enter OBi Expert button at the top of the next form. In the left column, click Voice Services and then SP1 Service. The fifth parameter is called X_InboundCallRoute. Beside it, uncheck both the OBiTALK Settings and Device Default checkboxes. Now enter sp3(6781234567) in the Value field for X_InboundCallRoute where 6781234567 is your actual Google Voice phone number (DID). Scroll to the bottom and click the Submit button.

Finally, at the top of the left column of the form, click Return to OBi Dashboard.

Configuring OBi SIP Trunk for Asterisk

1. Login to your OBi Dashboard using a web browser . After signing up for an account and registering your OBi device, click on the OBi 200 device in the My OBi Devices list.

2. In the Device Configuration dialog, click OBi Expert Configuration button. When prompted whether you’re sure, click OK.

3. In the OBi Expert Configuration Menu, click Enter OBi Expert button.

4. In the Production Information (left) column, click Service Providers.

5. In the Service Providers listing, click ITSP Profile C General.

6. For each of these fields, uncheck OBiTALK Settings and then uncheck Device Default:

  • General:Name
  • Service Provider Info:Name
  • Service Provider Info:URL

7. Fill in the ! field Values as shown below using the private IP address of your PBX:



8. Click Submit button after checking your entries carefully.

9. In the Service Providers listing on the left, click ITSP Profile C SIP.

10. In the ITSP Profile, enter the private IP address of your PBX in the Proxy Server, Registrar Server, and Outbound Proxy fields after first unchecking both the OBiTALK Settings and Device Default checkboxes.

11. Scroll down the form to X_SpoofCallerID and uncheck both the OBiTALK Settings and Device Default checkboxes. Then check the Value field for X_SpoofCallerID.

12. Scroll down the form to X_DiscoverPublicAddress and uncheck both the OBiTALK Settings and Device Default checkboxes. Then uncheck the Value field for X_Discover PublicAddress.

13. Click Submit button after checking your entries beside the 5 red exclamation points.

14. In the Production Information (left) column, click Voice Services

15. In the Voice Services listing on the left, click SP3 Service.

16. In the SP3 Service Profile, fill in the 5 fields in which the OBiTALK Settings checkbox is unchecked. The AuthUsername and AuthPassword entries will be used to authenticate to your PBX so be sure to choose a very secure password. It’s your phone bill. The URI field actually makes the trunk connection to your PBX so replace the 192.168.0.82 entry shown with the actual IP address of your PBX.

17. In the SIP Credentials section of the form, make certain that X_EnforceRequestUserID is unchecked. If not, uncheck both the OBiTALK Settings and Device Default checkboxes and then uncheck X_EnforceRequestUserID.

18. If you do not want to pass the CallerID number with your calls, in the Calling Features section of the form, be sure to check AnonymousCallEnable after unchecking both the OBiTALK Settings and Device Default checkboxes.

19. In the Service Providers listing on the left, click ITSP Profile A SIP.

20. Be sure X_SpoofCallerID is checked.

21. Click Submit button after checking your entries carefully.

Configuring Incredible PBX GUI for an OBi200

On the Incredible PBX side, log into the GUI using a web browser. We’ll be adding a SIP trunk, an outbound route, and an inbound route to process calls to and from the OBi device.

Add a SIP Trunk with a Trunk Name matching whatever you used in your OBi SIP credentials, e.g. obi200 or obi202. Plug in your Outbound CallerID to match your Google Voice phone number. In the Dialed Number Manipulation Rules tab, add a Match Pattern of NXXNXXXXXX. In the SIP Settings tab for Outgoing, the Trunk Name should match whatever you used on the OBi side, e.g. obi200 or obi202. In the PEER DETAILS, enter the following using the default username and password you assigned on the OBi side. Normally, port 5061 is the default port assigned on the OBi side. If you get a failed registration, try 5060 and then 5062 and 5063. Click Submit and reload your dialplan when finished.

type=friend
defaultuser=obi200
secret=your-password
qualify=yes
port=5061
nat=yes
host=dynamic
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
insecure=port,invite

For Outbound Call Routing, we recommend an Outbound Route using the 624 (OBI) prefix and 10-digit numbers. For example, if a user dials 624-888-1234567, your Incredible PBX server would place a call using the OBi’s Google Voice trunk to 1-888-1234567. When your Outbound Route setup looks like the following, click Submit and reload your dialplan.



For Inbound Call Routing, create an Inbound Route specifying a DID Number to match your Google Voice number. Choose a Call Destination to meet your own requirements, e.g. an extension, ring group, or IVR. Then click Submit and reload your dialplan.

Now you’re ready to test an outgoing call by dialing the OBi prefix (624) plus a 10-digit number. Then place a call to your Google Voice number using your cellphone and be sure Asterisk routes it to the destination you specified in your inbound route above.

Configuring VitalPBX to Use an OBi200

Truth be told, we weren’t bright enough to figure out how to configure the VitalPBX Trunk using credentials so we simply set up the SIP trunk using IP address authentication with the IP address of the OBi device. It works just as well and just goes to prove there’s always more than one way to skin a cat. So here’s the Trunk configuration on the VitalPBX side. The only entry you will need to change is the Host IP address for your OBi device. If you don’t know it, plug a phone into the OBi and dial ***1.

NOTE: For the Username and Description fields below, be sure to match what you used on the OBi side (above) for your SIP credentials, i.e. obi200 or obi202. If they don’t match on both devices, you won’t get a successful connection. Our apologies for mixing apples and oranges in the screenshots.



For Outbound Call Routing, we recommend an Outbound Route using the 624 (OBI) prefix and 10-digit numbers. For example, if a user dials 624-888-1234567, the VitalPBX server would place a call using the OBi’s Google Voice trunk to 1-888-1234567. Here’s the Outbound Route setup to make that happen:



For Inbound Call Routing, go to PBX:External:Inbound Routes and add an inbound route and destination for calls from your 10-digit Google Voice number. Or you can use the Default Inbound Route which we explained in our previous VitalPBX tutorial. Basically, you set up an Inbound Route with a Description and Routing Method of Default. All the other fields should be left as is except for the Inbound Destination. For the destination, you can choose an IVR, Extension, Ring Group, etc. to meet your own requirements.

Originally published: Monday, May 14, 2018


Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.


SPECIAL TREAT: If you could use one or more free DIDs in the U.S. with unlimited inbound calls and unlimited simultaneous channels, then today’s your lucky day. TelecomsXChange and Bluebird Communications have a few hundred thousand DIDs to give away so you better hurry. You have your choice of DID locations including New York, New Jersey, California, Texas, and Iowa. The DIDs support Voice, Fax, Video, and even Text Messaging (by request). The only requirement at your end is a dedicated IP address for your VoIP server. Once you receive your welcome email with your number, be sure to whitelist the provider’s IP address in your firewall. For Incredible PBX servers, use add-ip to whitelist the UDP SIP port, 5060, using the IP address provided in your welcoming email.

Here’s the link to order your DIDs.

Your DID Trunk Setup in your favorite GUI should look like this:

Trunk Name: IPC
Peer Details:
type=friend
qualify=yes
host={IP address provided in welcome email}
context=from-trunk

Your Inbound Route should specify the 10-digit DID. Enjoy!



Need help with Asterisk? Visit the PBX in a Flash 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. You can always find a little humor in insults if you dig deep enough. Ironically and unbeknownst to our pal, Steve, it was Sherman Scholten and his OBi development team that were among the first Google Voice "freeloaders." Only years later after Google Voice was integrated into FreeSwitch did Josh Culp at Digium perfect a clean way to integrate Google Voice into the Asterisk platform. []

Obivoice = OBi Heaven: Dumping Google Voice for Less Than 10¢ a Day

What a difference a week makes! When we wrote last week’s article about netTALK and their terrific pricing, we were pleased to report that at least one company could offer a drop-in replacement for Google Voice without breaking the bank. But, alas, all is not well in netTALK Land. For openers, the Better Business Bureau revoked their accreditation last June because of failure to respond to or resolve technical complaints. And a recent SEC Filing paints a fairly bleak picture of the company’s financial condition. Special thanks to Gershom1624 for his sleuthing efforts. This merely reinforces the difficulty of providing reliable, unlimited VoIP service at the $2.50 a month price point. But we firmly believe $2.50 is the magic price point, and it is achievable with some safeguards for the provider, i.e. residential service, no call centers, no 10,000 minutes-a-month customers. My mom loved the telephone, but she never spent 5 hours a day on the telephone. There also has to be some tradeoff in the level of support customers can expect. If customers tie up expensive support reps with multiple calls, the pricing matrix falls apart very quickly. And that brings us to this week.


Let’s review the Wish List for those that missed last week’s article. We want a drop-in replacement for Google Voice on both the OBi110 (stand-alone with any POTS telephone) and Asterisk® (PBX) platforms. It needs to provide unlimited (within reason) calling in the U.S. and Canada. It needs a feature set that is fairly comparable to Google Voice. It needs to include E911 service because the federal government says so. We don’t care much about support as long as the setup process is well-documented, the service is reliable, and calls sound great. Charging for support requests to resolve issues that aren’t the company’s fault is perfectly fine with us. But the price point for unlimited calling needs to be $2.50 a month, i.e. $30 a year or $60 every two years for the math-challenged. We’d prefer no tips, taxes, or fees. We want to keep our existing number. And, lest we forget, the company must promise to stay in business and never raise prices… forever.

Suppose we could find you a company that, with a 2-year commitment, could provide all of the above (minus the last sentence) plus fax support including a web page to send outgoing faxes from attachments, free calling and a mobile app for your iOS and Android devices, Visual Voicemail with voicemail transcription as well as email delivery of voicemail messages, call forwarding, call waiting, CallerID spoofing for any number you own, and unbelievable customer service. Not sure about the service? How about a 30-day free trial with 60 free minutes?

Let us introduce you to Obivoice. Don’t be alarmed by the one-year price of $40. The two-year price is just $60. But it doesn’t cost you a nickel to sign up and try the service. Obivoice is a pure SIP provider so the setup with PBX in a Flash™ or an OBi110™ takes only a couple minutes. Here’s the SIP trunk setup for PBX in a Flash using FreePBX®. All you need is your SIP credentials and phone number once you’ve signed up for an account. Plug in your 10-digit phone number in the Outbound CallerID and Register String, replace 1234 with your Account Number in the username, fromuser, and Register String, and replace yourpassword with your real Password in the secret and Register String.

Next, build yourself an Inbound Route with your 10-digit DID and point it to your favorite PBX destination. Finally, create an Outbound Route using obivoice as the Trunk Sequence, and you’re all set. It doesn’t get any easier than that.

We don’t think you will but, if you need assistance setting this up, head over to the PIAF Forum where there’s a lively discussion about Obivoice already.

The OBi110 setup is just as easy. Plug in sms.intelafone.com as the ProxyServer and OutboundProxy in your ITSP Profile, add your SIP credentials in the SP1 Voice Services dialog, and forward (or transfer) your existing Google Voice number to Obivoice. Done! Obivoice’s complete tutorial is available here.

Let us close with our own customer service story. We were so excited about this new service when it was announced yesterday that we actually clicked the wrong button and signed up for the wrong plan. Of course, it only takes a minute to get that sinking feeling in your stomach when you know you’ve screwed up. So late yesterday (Sunday night!) I opened a support ticket and asked to either cancel the wrong plan so that I could reenlist or to transfer to the $60 two-year plan. At 1:30 a.m. this morning, I got an email back from customer service indicating that the plan had been adjusted and that I had been billed for the price difference. WOW!

Run, don’t walk, to sign up for Obivoice. It’s that great!

p.s. The Obivoice jingle in their YouTube video is as good as their calls. We want it for our Music on Hold!

Originally published: Monday, January 13, 2014



Need help with Asterisk? Visit the PBX in a Flash Forum.


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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.
 



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