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The Most Versatile VoIP Provider: FREE PORTING

An Open Letter to Google: Don’t Do It!

With an obscure post on its support forum, Google has quietly announced that it will discontinue Google Voice support for XMPP on June 18. According to Obihai1 insiders, it will be replaced with a Google-proprietary version of the SIP protocol to which only Obihai has been provided access despite claims from Google staff (without documentation) that the "new Google Voice" will be "standards-compliant" and "should work with many third party solutions."

July 20 Update: Google did it anyway and pulled the plug on their XMPP implementation of Google Voice. See this Nerd Vittles tutorial for the latest fixes for Asterisk without purchasing any new hardware.

If you loved New Coke, this should be a hit. You may recall that Google has attempted similar switcharoos previously only to retreat at the last moment and continue support for "Legacy Google Voice" a.k.a. Google Chat which works with Asterisk® and currently looks like this:


If you happen to have an Obi 200 series device and revisit the same Google Voice Legacy Settings dialog today, what you will now see looks something like this. In addition to the disappearance of the Google Chat option, note the proprietary FQDN in the SIP URI as well as the MAC address accompanying the specific OBi hardware device designation. That’s three clear indicators that this new "service" was engineered to be anything but open.


The important point here is that all existing Google Voice XMPP connections through Asterisk, pygooglevoice, 3CX, OBi 100-series devices, and the Simonics SIP/GV gateway will fail beginning June 18. In its place, we get a new (proprietary) monopoly courtesy of Google and Obihai/Polycom. Can this change? Of course. What are the chances? Not likely. They’re already rolling it out to OBi hardware. And, if you happen to be one of the millions of Asterisk users that has depended upon Google Voice for communications, too bad for you. In fact, when we posted comments on both the Google Voice and Obihai forums warning of the upheaval this would cause in the VoIP community, both comments were promptly removed. So much for transparency and standards compliance. Wouldn’t you think Google would have the decency to at least alert Google Voice users through their registered email addresses that the service was being discontinued after users have relied upon it for almost ten years? Apparently not. A SIP FQDN that begins with a corporate name is not a good sign. It’s anything but standards-compliant. Quoting one of the OBi shills, "Google isn’t obligated to support anything else." And then there’s this from a moderator on the Google Forum:




So where do we go from here? There are several options. None of them are particularly appealing. First, you can port out your Google Voice number to another provider. You’ve got about five weeks to get it done. Second, you can continue to use the existing Google Voice Settings menu (so far) to forward incoming calls to a DID or phone number that you already own. What you lose is the ability to make outbound calls using that Google Voice trunk. Third, you can purchase an OBi 200-series ATA and set up a SIP trunk to process calls from the OBi200 in much the same way that you do today. Aside from the $50 bounty, the only other wrinkle that we’ve found is that FreePBX® currently does not support DIDs of over 50 characters (as are used with the new GV DIDs) so you will need to configure a default inbound route to process incoming calls from your OBi devices or apply the patch that we’ll provide for Incredible PBX® platforms. It should also work with generic FreePBX setups.

We have mixed emotions about documenting this OBi 200-series trunk setup. Other sites have pulled their tutorials arguing that we should boycott Polycom and Obihai devices as well as Google Voice until Google cleans up its act. After all, Polycom has worked with Google for months to design and build this new proprietary setup. It wasn’t an accident. On the other hand, we have championed Google Voice since its inception, and thousands of our followers depend upon Google Voice for their production PBXs. So we’re holding our nose in documenting the setup here. In the meantime, we hope each of you will write and post scathing comments about Google Voice and publish them widely. Do it today! Bad publicity is probably the only thing that will prompt Google to change directions at this juncture.

Continue Reading: Creating an OBi200 Google Voice Trunk to Use with Asterisk

Originally published: Saturday, May 12, 2018

  1. In case you didn’t know, Obihai recently sold out to Polycom. []

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.


whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for all of us.


 

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…

netTALK to the Rescue: Dumping Google Voice for Less Than 10¢ a Day

The Google Voice Adventure with Asterisk® has been disappointing to put it charitably. Google never really saw the benefit of providing free phone service to millions of Americans because they never could quite figure out how to monetize the project. In the meantime, shady call centers were eating them alive with dozens if not hundreds of Google Voice trunks that were placing endless calls around the clock. The final straw was Microsoft deciding to keep Skype proprietary while adding free Google Voice connectivity to its communications products. This meant Microsoft customers had the best of all worlds while Google’s platform had no way to access Skype except through Microsoft’s proprietary client. Google decided to pull the plug on XMPP beginning May 15 of this year and more or less blamed it on abuse by the open source community for using Google’s own open source development toolkit for Google Voice.

We’ve never been one to sit around crying about spilt milk when there are plenty of other excellent choices available to the VoIP community. Today we begin our exploration of alternatives with a look at all-you-can-eat VoIP. There still are a few pure VoIP service plans available, but every one that we’ve tried leaves a bait-and-switch aftertaste. The first year may be reasonable, but once they’ve got you hooked, look out. Quite literally, they have your number. For this reason, we’ve chosen a hardware hybrid approach that still relies upon VoIP for the actual calls. Below the stratospheric pricing of the Bell Sisters, Comcast, and Vonage, there still are several wallet-friendly, all-you-can-eat VoIP products to choose from including netTALK, Ooma, and magicJack.

We know. Nothing beats free even with a little pain. But we think you’ll love today’s alternative especially given its expanded feature set and modest long-term cost. Up front hardware cost including service for the first year is about $100. Amortizing the hardware and service costs over three years reduces your investment for unlimited U.S./Canada/E911 phone service to roughly $1 a week. After recovery of your $100 hardware investment, the cost is $29.95 a year which works out to less than 10¢ a day… forever. This compares quite favorably to today’s best all-you-can-eat VoIP deal. AxVoice charges $99 for equivalent first year service and then the price escalates to over $150 for subsequent years. It doesn’t take a math major to figure out that’s 5 times the netTALK pricing beginning in Year 2.

That brings us back to Google. If you had several million happy customers already using your VoIP service and you saw a small company that was still in business charging $30 a year for a VoIP feature set that was better than yours, wouldn’t you think you might try to cash in on $100 million a year in new revenue rather than flushing the project down the toilet while shafting the open source developers that got you the customers in the first place??

Earth to Google: Few on the planet are ever going to use a web browser with Hangouts to make traditional phone calls regardless of how many places you plaster the Hangouts logo. Before you hire another Marketing Genius, read a good book or two. The well-deserved 2013 Lame Foot of the Year Award goes to Google. </rant>

Overview. Today we’ll be pairing an old friend, the OBi110, with the unlimited calling options provided by netTALK. When we’re finished, you’ll have a drop-in replacement for Google Voice on your Asterisk server that provides unlimited calling within the U.S. and Canada, plus free calling to other netTALK and OBi users around the world, plus free 911 emergency service for you and your family, plus voicemail delivery by email, and fax support. And you can keep your existing phone number! All of the existing PBX in a Flash and Incredible PBX features still work exactly as they do today without worrying about Google pulling the rug out from under you… again. With the OBiON app for iOS or Android, you can make free calls from your cellphone using today’s netTALK-OBi110 setup. And, if calls from a cellphone aren’t your thing, when you go on vacation to anywhere with an Internet connection, you can slip the netTALK device into your suitcase and plug it in to the Internet at your destination without ever losing the ability to make and receive free calls. We’ll cover all these magic tricks and more today so hang on to your hat. Let’s get started.

Legal Disclaimer. This is not legal advice. Consult your own attorney for that. We have reviewed netTALK’s Terms of Service and find nothing that would preclude your using the services as described in this article so long as the device is used in the United States, usage is under 3,000 minutes per month, and usage is limited to "normal residential or home office usage patterns" without "auto-dialing, continuous or extensive call forwarding, telemarketing, fax broadcasting or fax blasting." Terms of service can and do change from time to time. Review them regularly.

BY IMPLEMENTING THE TIPS IN THIS TUTORIAL, YOU AGREE TO ASSUME ALL RISKS ASSOCIATED WITH THE METHODOLOGY INCLUDING, BUT NOT LIMITED TO, THE LEGAL AND FINANCIAL CONSEQUENCES OF YOUR ACTIONS. IF YOU ARE UNWILLING TO DO SO, STOP READING HERE!

Hardware Requirements. Here’s what you’ll need. First, purchase a netTALK device. You have several choices. The netTALK DUO is still available for under $50 and includes a full year of unlimited calling in the U.S. and Canada. The netTALK DUO II is the newer model (with the same feature set). It sells for about $30 but only includes three months of free calling. The netTALK DUO WiFi is about $60 and adds WiFi support. Additional years of free calling in the U.S. and Canada are $29.95 with a guarantee of no price escalation as long as you continue the service without interruption. You can add free calling to 60+ countries for an additional $10 a month. Unlimited SMS messaging in the U.S. and Canada is an additional $2.50 a month. AT&T charges $20 a month for unlimited SMS messaging, and it only works on a single cell phone.

In addition to your Asterisk server, the other piece for today’s puzzle is OBiHai’s OBi110, a terrific analog telephone adapter that we’ll use to connect the netTALK adapter to your Asterisk server. If you want to connect a Google Voice account for a few more months, it can do that as well. It also supports a connection to another SIP provider of your choice for redundancy. For today, our focus is getting a Google Voice replacement service in place for your Asterisk server. You can scour the Internet to add the other pieces. The OBi110 is available through Amazon for under $50.1

Installing and Configuring the netTALK Duo

Before your netTALK Duo will work, it has to be registered on the netTALK web site. Locate your temporary username and password for the NetTalk DUO inside the box. Log into the web site and click Start Activation. Plug in your credentials and click LOGIN. Fill out the registration information and create a username and password for your new account. Then press CONTINUE. Complete the E911 information and click SAVE. Select a phone number and ASSIGN it to your account. Now plug a plain-old phone into your netTalk Duo, connect the device to your LAN, and then connect the power adapter. Some routers are problematic. Be sure SIP ALG is disabled on your router. It took about 5 minutes for ours to change from alternating green and red lights to a solid green light and the one-ring call indicating that the device is operating properly. Once you get the solid green light, make a call to the device and from the device. Nothing else works if the netTALK can’t make calls! Once it’s working, you can unplug the phone and use it to configure the OBi110 in the next section.

Installing and Configuring the OBi110

There are a number of steps to the OBi110 setup, but it isn’t difficult. If you can handle slice-and-bake cookies, you can do this. Just follow the recipe and don’t skip any steps. We’ll be configuring the OBi110 in two phases using the OBiTalk web site first and then using the OBi110’s built-in web server. Plug the OBi110 into your LAN and then attach the power adapter. Plug a POTS phone into the PHONE port of your OBi110. Once the OBi110 has booted, pick up the phone and make sure you have a dial tone. Then hang up.

IMPORTANT: Make sure that you restore the OBi to its factory default settings if you have previously used the device! ALWAYS keep your OBi110 behind a hardware-based firewall with NO Internet port exposure!

Now head over to the OBi web portal and set up an account if you don’t already have one. From the OBi Dashboard, click ADD DEVICE. Uncheck the box to set up a Google Voice account. You can do that later if desired. Now pick up the phone connected to the OBi110 and dial **5 plus the 4-digit number shown in your browser. This will identify your device to OBiTalk. Your OBi110 will appear in a dialog box for confirmation. Click CONFIRM promptly, or start over.

In the Device Configuration window that appears, add a Device Display Name, Webpage Admin PW, OBi Attendant PIN, and your TimeZone. SAVE your settings. The OBi110 should now appear in the OBi Dashboard with its assigned OBi number and speed dial number together with a Green status icon signifying it’s working.

Now is a good time to download the OBiON app to your iOS device or Android phone. Launch the app and login with your OBiTalk account information. In the OBi Dashboard, you will note that your softphone now has appeared and was assigned a 9-digit OBiTALK number. Write it down. You’ll need it in a minute to complete the OBi110 setup. Click on the Edit icon for the softphone and assign your OBi110 as the OBi Voice Gateway. SAVE your settings.

For the remainder of the OBi110 setup, we’ll be using the web interface built into the OBi110. If you don’t know the IP address of your OBi110, pick up the phone connected to your OBi and dial ***1.

1. Use your browser to log into the OBi110’s web interface. Log in with admin:admin as the username:password.

2. Once you’re logged into your OBi110’s web interface, the Setup Wizard will display. Expand the first five headings in the left column by clicking on the + icons for Status, System Management, Service Providers, Voice Services, and Physical Interfaces. Then expand ITSP Profile B under Service Providers.

3. Download the latest firmware from here to your desktop. Currently it’s 1.3.0 (Build: 2824). Install it on your device: Device Update -> Firmware Update. Your OBi110 will restart after loading the new firmware.

4. Disable ALL AutoProvisioning: Auto Provisioning -> Firmware Updates, ITSP Provisioning, OBiTalk Provisioning. Then Submit and Reboot.

This keeps external forces from stepping on your setup once it’s working. If something breaks down the road, you can manually provision your device once you know what’s broken.

5. While not absolutely necessary, we recommend you set a static IP address for your OBi110: Network Settings -> Internet Settings. Submit and Reboot. Using your browser, log back into the new IP address.

Another alternative is to permanently lock the DHCP-assigned IP address to the OBi110 using the web interface of your router.

6. Open the SIP profile under ITSP Profile B. Here you’ll need to insert the IP address of your Asterisk server in BOTH the ProxyServer and X_AccessList fields. Also add a check mark for X_SpoofCallerID. Before you can add these entries, you’ll need to uncheck the Default checkbox beside each entry. This applies to all further steps as well. After making the three entries, click Submit and Reboot.

7. Open the SP2 Service window. For X_ServProvProfile and X_CodecProfile, change the settings to B. Change X_InboundCallRoute to LI. Add a check mark for X_KeepAliveEnable. Change X_KeepAliveServerPort and X_UserAgentPort to 5061.

In the SIP Credentials section, change AuthUserName to obitrunk. Make up a secure password and insert it in the AuthPassword field. Remember the password! We’ll need it to configure your Asterisk trunk in a minute. For the URI entry, use the following with the actual IP address of your Asterisk server: obitrunk@192.168.0.82. Double-check all nine entries carefully and then click Submit and Reboot.

8. In the OBiTalk Service Settings window, change the InboundCallRoute to an entry that looks like this: {pp(ob290999999),li}. We recommend you cut-and-paste our example and then replace 290999999 with the 9-digit OBiTalk number that was assigned to your softphone above. A punctuation error here will block your softphone from ever working. Click Submit and Reboot.

9. Finally, we need to configure the LINE Port. For the InboundCallRoute, insert the following using the 10-digit phone number assigned to your netTALK Duo: SP2(6781234567). For the SilenceTimeThreshold, set the number of seconds you want the OBi110 to wait before disconnecting a call where nobody at the other end of the call says anything. We recommend 600 which is 10 minutes. Click Submit and Reboot.

10. Now it’s time to connect your netTALK Duo to your OBi110. Unplug any phone connected to the netTALK Duo. Using a telephone cable, connect the PHONE port of the netTALK Duo to the LINE port of the OBi110. Never plug the netTALK Duo into the PHONE port of the OBi110, or your OBi is (burnt) toast!!!

11. Test your configuration. Pick up the phone that’s still connected to the OBi110 and dial either a 10-digit or 11-digit number of someone you love: 8005551212 or 18005551212. Do the same thing using the OBiON app on your cellphone or tablet. Be patient! OBiON connections are not instantaneous. Your connections have to be authenticated through OBiHai’s servers before they go through.

Interconnecting Asterisk with the OBi110

There basically are three pieces you need to add to Asterisk so that it can communicate with your netTALK Duo and OBi110. You need a Trunk to which the OBi110 will register. You need an Inbound Route to tell Asterisk how to handle incoming calls from the netTALK Duo phone number. And you need an Outbound Route to tell Asterisk which outgoing calls should be routed out through the netTALK Duo. We’re assuming you will be using the netTALK Duo as your primary trunk for outbound AND emergency calls. We’re also assuming you will not be making international calls. Finally, we’re assuming you are using FreePBX 2.11 with either PBX in a Flash or with one of the Incredible PBX builds on the CentOS 6.5, Raspbian, or Ubuntu platforms. Other FreePBX 2.11 setups should work in much the same way. If any of these assumptions don’t apply, you’ll obviously need to make the necessary adjustments for your environment.

Trunk Configuration. To set up the obitrunk under FreePBX 2.11, log into FreePBX and choose Connectivity -> Trunks -> Add SIP Trunk. For the Trunk Name, use obitrunk. For Outbound Caller ID, enter the 10-digit phone number assigned to your netTALK Duo. For Maximum Channels, use 1. For Dialed Number Manipulation Rules, add the following Match Patterns: 1NXXNXXXXXX, NXXNXXXXXX, and 911.

In Outgoing Settings, use obitrunk for Trunk Name and enter the following PEER Details:
type=peer
host=dynamic
port=5061
disallow=all
allow=ulaw
dtmfmode=rfc2833

In Incoming Settings, enter your actual 10-digit netTalk phone number in the User Context field: 6781234567. Enter the following USER Details replacing mypassword with the password you set up in OBi110 step #7 SIP credentials above and adjusting the permit entry to match your LAN subnet:
type=friend
secret=mypassword
host=dynamic
context=from-trunk
canreinvite=no
nat=yes
port=5061
qualify=yes
dtmfmode=rfc2833
disallow=all
allow=ulaw
deny=0.0.0.0/0.0.0.0
permit=192.168.0.0/255.255.255.0

Click Submit Changes and Apply Config to save your entries.

Inbound Route Configuration. If you already have a default inbound route on your Asterisk server, then you can skip this step unless you want incoming calls from the netTALK DID routed in a special way. To create an inbound route for the netTALK phone number, choose Connectivity -> Inbound Routes -> Add Incoming Route. For the Description, enter netTALK-OBi110. For the DID Number, enter your 10-digit netTALK phone number. For CID Lookup Source, choose CallerID Superfecta if you’re using this module. For Set Destination, choose how you want FreePBX to route the incoming calls, i.e. an extension, ring group, IVR, etc.

Click Submit and Apply Config to save your entries.

Outbound Route Configuration. If you want all 10-digit, 11-digit, and 911 calls placed from your Asterisk server to be routed out through the netTALK Duo, then you’d Add a Route under Connectivity -> Outbound Routes that looks something like the following. Don’t forget to move this Outbound Route (in the right column) to the TOP of your list of Outbound Routes to make certain it is processed first by FreePBX.

For Route Name, use obiout. For Dial Patterns, use the same ones you used in your Trunk setup above: 1NXXNXXXXXX, NXXNXXXXXX, and 911. For Trunk Sequence, select obitrunk.

Click Submit Changes and Apply Config to save your entries.

While still in Outbound Routes, drag obiout to the top of the outbound routes list in the right column. Then click Apply Config again to save your trunk processing sequence.

Verifying Connectivity. Let’s be sure everything works. First, log back into the IP address of your OBi110 and verify that System Status -> SP2 Service Status shows the OBi110 is registered to your Asterisk server. Next place a 10-digit call using an extension on your Asterisk server and monitor the Asterisk CLI to make certain that the call went out using the netTALK Duo trunk and was completely successfully. Finally, use your cellphone to call the number assigned to your netTALK Duo. The call should ring on the devices you configured in the Inbound Route above. Enjoy your new freedom from Google Voice!

Special Thanks. We want to express our appreciation to ObiHai for an excellent Administrator’s Guide and to the numerous individuals who have wrestled with the OBi110 setup over the years. This includes Adrian Li, Ad_Hominem and MichiganTelephone on the OBiTalk Forum as well as the reference articles which now are available here.

Originally published: Tuesday, January 7, 2014



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


whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for all of us.


 

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…

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

Paradise Found: The Holy Grail of Mobile Communications » Meet the OBi202

Much has been written about the quest for Unified Communications where all real-time communications services are transparently accessible by phone, by email, or via the web using multiple devices and various media types. But the Holy Grail of Business Communications is a bit different from our perspective. For the modern business person, business telecommunications comes down to a 3-way time slice between a home or home office, a real office with a real phone, and a cellphone whenever the person morphs into a road warrior or telecommuter. What the business person really wants is transparent integration of his or her smartphone into existing home and office phone setups. In other words, when we’re at home with a cell phone, we want to answer incoming cell phone calls on a house phone rather than scrambling to find a ringing cellphone on the other side of the house. And when we’re at the office, we want incoming cellphone calls to either ring on our desktop phone or be redirected to the office PBX when we’re unavailable. For many businesses, the only phone number that a customer ever has is the business person’s smartphone number. So, when we place outbound calls from home, or the office, or the cellphone, we want the customer to always see the cellphone number in the CallerID display. For accounting purposes, we’d also like all of the calls to be recorded in the cellphone log so that we can actually track call activity without reviewing logs in three different places. Well, we’ve finally got it!

Google Voice™ solved some of this cellphone integration with its new service. You can actually have calls ring on multiple devices simultaneously including your cellphone, your office phone, and your home phone. But there are several limitations. First, it’s only available in the United States. Second, some folks just don’t want Google knowing everything about your call history. Third, there’s a flexibility issue when using Google Voice to forward calls to your home or office. Without a lot of hands-on daily management, the incoming calls get forwarded to your alternate numbers whether you’re there or not. So, for example, if you’re on the road, you probably don’t want incoming business calls to your cellphone picked up by either your spouse or the office switchboard because there’s no way to easily route the calls back to you.

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And that brings us to a pair of terrific new products from ObiHai. Last year we were provided a prerelease version of the OBi202™ at no cost, but frankly we hadn’t had time to play with it until a business opportunity presented the perfect case for the OBi202. In addition to being a terrific standalone VoIP phone system, the OBi202 also supports an analog fax machine using Google Voice as well as T.38 digital faxing with a CallCentric trunk. And then we discovered the OBiBT™ Bluetooth® Adapter which we actually purchased. Lo and behold, for under $100, the Holy Grail of Mobile Phone Communications was staring us in the face. So let’s begin with a quick summary of how all of this works, and then we’ll show you how to do it yourself. As with all of our Amazon links, you are obviously free to buy products wherever you like. Where prices are competitive or availability is a factor, we often recommend Amazon because Amazon provides financial support to Nerd Vittles through its referral links. That really does help keep the lights on so thanks in advance for your understanding.

The way The Holy Grail works in our SOHO setup goes something like this. I carry a Bluetooth-enabled, Galaxy Note II smartphone. While away from the house, the smartphone works like any other cellphone. I can make and receive calls as well as email and SMS messages. The only difference is that I have the phone connected through a Google Voice number so that incoming SMS messages are also delivered as Gmail messages. When I return home, the OBi202 with the OBiBT adapter senses that a smartphone is within range. You can pair and prioritize up to 10 of them. Incoming calls still ring on the cellphone, but they also ring on some POTS cordless phones scattered around the office. The POTS phones are connected to the first of two phone connections on the OBi202. CallerID actually shows the same thing as the cellphone CallerID. And incoming SMS messages also appear in the CallerID display of the cordless phones. If an incoming call is not answered in two rings, the OBi202 transfers the call to our Incredible PBX™ running on a Raspberry Pi®. It then processes the call through an AutoAttendant and delivers the call either to all of the house phones or to the desired person in the house. If there’s no answer, the call is handled by the voicemail system in Incredible PBX, and the message is also emailed to the desired recipient.

While at home base, outbound calls from the POTS phones in the office are always placed through my cellphone using the Bluetooth connection in the OBi202. Depending upon how you set up your Google Voice interaction with your cellphone, outbound calls will show either your cell phone number or Google Voice number as the CallerID. When we leave the office, the office phones no longer ring, just the Galaxy Note II. And outbound calls from the SOHO cordless phones are handled using a preconfigured SIP provider or Google Voice trunk in the OBi202 instead of via Bluetooth and the smartphone.

This may sound trivial to some of you. Suffice it to say, it’s not. You won’t find any commercial PBX that can do it. And the Asterisk Dev Team has been working on a Bluetooth connector called chan_mobile for as long as we can remember. It still doesn’t work reliably. You can follow the progress of our half dozen chan_mobile pioneers here.

Getting Started with the OBi202. Before you can tackle Bluetooth, you need to get a perfectly functioning OBi202. Plug it in with a network cable behind your router which must provide a DHCP address to the device. Plug a POTS phone into PHONE 1. Now make a test call to OBiTALK by dialing **9 222 222 222. Next, decipher the IP address of your device by dialing ***1. Make sure your device is running the latest software by dialing ***6. Using a browser, go to http://www.obitalk.com. Create an account and then log in. Choose Add Device and follow the prompts to get your new device registered. If you want to use Google Voice, now is the time to set up your account. Choose Configure Voice Service Providers, choose your provider, and specify what phone port to use for the service. By default, both phone ports will work with whatever service provider you first configure. If you want to register your OBi202 as an extension on your Asterisk® server, now’s the time to do that as well. We also recommend you create an account with VoIP.ms and obtain a free INUM trunk. You can read how to set this up and why in this Nerd Vittles article. The advantage of having this trunk is that you can use it to route calls between your OBi202 and your Asterisk server at no cost. Just create and then register separate subaccounts on VoIP.ms for both your Asterisk server and your OBi202. Build a trunk and an inbound route on your Asterisk server to route calls from your INUM DID to wherever you’d like incoming INUM calls to go, e.g. an extension, a ring group, or an IVR. INUM DIDs look like this: 88351000XXXXXXX where the last seven digits are your personal number. Use SP4 on your OBi202 to set up your VoIP.ms subaccount. Be sure all of the accounts you create get properly registered.

Configuring Bluetooth on the OBi202. Setting up the Bluetooth functionality is straightforward. Plug in the OBiBT. Dial *28 from a phone connected to the OBi202. Within two minutes, open the Bluetooth network settings menu on your smartphone and pair it with the OBi202. If prompted for a passcode, it’s 0000. Refresh your OBi Dashboard, and click on Edit BT icon in the Voice Service Providers frame. Set the device up as shown above. Click the Submit button.

At this point, incoming calls on your cellphone will also ring on the POTS phones connected to your OBi202. And calls that you place using a phone connected to the OBi202 will be routed out through your cellphone. This may be sufficient for many of you. We wanted the added functionality of routing inbound calls to our PBX when there was no answer on the OBi202-connected phones. At least with AT&T and StraightTalk, two rings is about the most you can allow without risking a voicemail pickup through your cellphone provider. Here’s how to set it up.

From the OBi202 Device Configuration Menu, click on the blue OBi Expert Configuration button. Acknowledge that you know what you’re doing and then click on the blue Enter OBi Expert button. In the left column under Voice Service, click OBiBlueTooth. Edit the Calling Features section and make it look like what’s shown above, replacing xxxxxxx with your personal INUM DID assigned from VoIP.ms. Click the Submit button when you’re finished. Now incoming calls will ring twice on your OBi202-connected phones and then be transferred to the INUM DID configured in Asterisk.

You can check the status of your OBi202 at any time by launching OBi Expert and clicking System Status. Enjoy!

Originally published: Monday, February 4, 2013


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 Forum. It’s the best Asterisk tech support site in the business, and it’s all free! Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you.


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


whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.


 

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…

Meet the OBi110: A Permanent Google Voice Fix for Asterisk

We’re going to take a little time off for Spring Break and leave you with a terrific new tutorial from our good friend, Tom King. But first, despite pitching Google Voice as one of Asterisk’s Top 10 Tricks as recently as last October, Digium® apparently has had a change of heart. Our frustration with Asterisk® and Digium over the tepid support for Google Voice™ continues to build with the discovery that the latest (several) releases of Asterisk 10 break Google Voice connectivity entirely. The default Asterisk 10 install in PBX in a Flash™ continues to work just fine. The Digium response can be summed up in two words: "Oh Well." They’re apparently too busy doing Amazing New Things™ to worry about keeping your one-month-old PBX functioning reliably. So… we’ve pretty much given up on Digium’s attitude toward Google Voice ever changing. It’s simply not a priority for them which, of course, is their prerogative. But it also means everyone needs to start considering other alternatives if Google Voice reliability matters to you.

So today we start down a new path for our users and readers as well as the rest of the VoIP community. We hope to have a FreeSwitch® announcement soon to reliably handle Google Voice and Skype for Asterisk-based servers. These two functions have worked flawlessly with FreeSwitch since Anthony Minessale and Brian West first released them a couple years ago. In the meantime, reliability of Google Voice in Asterisk continues its downward spiral with almost monthly nightmares. The latest debacle is a month old today. Happy Birthday! 🙄

There’s another alternative as well. Sherman Scholten at OBiHai tells us they are poised to release the OBi202 with all the usual OBi110 goodies plus T.38 real-time faxing over IP plus support for PPPOE, VLANs, and up to 4 SIP or Google Voice trunks. Add a firewall with DRDOS attack protection and VPN pass-through plus some amazing PBX-like functionality for management of collaborative calling, and you really couldn’t ask for much more in a product which will retail for under $100. OBiHai has been kind enough to send us a complimentary unit, and we’ll have a full review for you soon.

In the meantime, we have a short term answer for anyone that depends upon Google Voice to perform tasks (such as making phone calls) where reliability matters. It’s the under $50 OBi110. You’ll find a link to buy one while supporting Nerd Vittles in the right column. And today we’ll show you how to set it up to use with Asterisk and PBX in a Flash™ so that Google Voice calls flow into and out of your server reliably and transparently without worrying about who may have "improved" things while you were sleeping.

PIAF2 Preliminaries. If you’re currently using PBX in a Flash 2 for your Google Voice needs, then the first thing you need to do is remove any Google Voice trunks you’ve activated using the Google Voice module in FreePBX. Once you’ve done that, you’ll also want to disable the jabber and gtalk modules in Asterisk. This has no impact upon the separate gvoice command line utility which will continue to work fine with the speech-to-text apps that we’ve released over the last month. The Google Voice for Python project is well supported and (fortunately) is separate and apart from the Asterisk project. We’ve also documented on the PIAF Forums how to keep gvoice running reliably on your server.

To disable Google Voice in Asterisk, log into your server as root and edit modules.conf in /etc/asterisk. Change the two lines in the [modules] context for these two modules by changing the word load to noload. Then save your changes and restart Asterisk: amportal restart.

noload => res_jabber.so
noload => chan_gtalk.so

Step2. Once you have your OBi110 in hand, the rest of the process to get it handling inbound and outbound Google Voice calls for Asterisk is simple as long as you don’t skip any steps. Just download Tom King’s new tutorial and follow along. You’ll be up and running in under 15 minutes with a reliable, independent alternative for Google Voice calling with Asterisk. Enjoy!

Originally published: Friday, March 16, 2012


Well, we’re just a few folks shy of 5,000 followers on Google+. See the right column for today’s tally under Google Goodies. That’s less than 10% of our weekly Nerd Vittles fan club. So what are you waiting for? We can’t promise you one of these but, if you become #5000 to put us in your Google+ circles, we do want to hear from you! Please include your mailing address. 😉



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


whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.


 

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…

VoIP My Way: Best VoIP Bargain of the Year Or Is It?

There haven’t been many all-you-can-eat VoIP bargains lately. So we’re delighted to finally have something to talk about besides Google Voice which is once again in flames on the Asterisk® 1.8 platform. As those of you using Google Voice have discovered, one real limitation of the service is the lack of support for 911 emergency dialing. We’ve always recommended that users have a backup alternative, not only for 911 calls, but also as another layer of redundancy particularly if you’re relying upon Google Voice as your primary VoIP provider.

Today’s $40 VoIP My Way offering could fit the bill perfectly. You not only get a DID in your choice of area codes in the U.S., but you also get E911 service PLUS unlimited inbound calling and free outbound calling within the U.S. and Canada. The fine print makes clear that this is a residential offering and typical usage should not exceed 1500 minutes a month. But for $40 for a year of service, you really don’t have a lot to lose… other than your $40. And the call quality is impeccable.

One reason the call quality is excellent is because VoIP My Way is reselling VoipO’s service. Whether you can make the offering work as a business model is certainly a question. VoipO charges a $199/year reseller fee plus $1 a month per DID and another $1 a month for E911 service. That leaves $16 profit for the year assuming no one ever made a call, assuming you never fielded a call for support, and assuming there weren’t any expenses due to fraud. The odds of any of those assumptions panning out are about zero.

VoipO charges resellers a penny a minute for inbound and outbound calls. This is about double the typical VoIP wholesale rate. In its literature, VoipO also suggests that the average usage is 470 minutes a month per VoIP customer. That would cost VoIP My Way over $56.40 for calls each year which would leave VoIP My Way about $40 in the hole per Wedding Special customer. Suffice it to say that all-you-can-eat customers typically aren’t worried about per minute charges so there are likely to be many customers with monthly usage at or above 1,500 minutes. So here’s the math on a 1,500 minute a month customer: $15.00 for calls + $1 for DID + $1 for E911 service which adds up to a $204 annual cost for each $40 customer. Hopefully, the other plan offerings including the business plans will make up for the almost certain loss of money on the Wedding Special.

We mention all of this so that you can judge the risk of service interruption at some point down the road. Also keep in mind that VoIP My Way offers a 30-day, 300 minute evaluation with a money-back guarantee. And PayPal gives you 45 days from the date of the order to open a claim in the Dispute Resolution Center. Finally, we should note that VoIP My Way posted a comment on VoipO’s reseller forum indicating at least some interest in "selling off the voip company." When we questioned the posting as others have as well, the following response was provided. None of the claims have been verified by us incidentally.

It was thought a few days ago I was getting hit hard by orders and a lot of fraudulent ones at that. The soon to be misses started to get a annoyed so I put a feeler out. Thats about as far as it went i had a few people contact me but no one was able to handle the amount of clients [100] I had at the time.

Since then I have taken on one temp employee to help field emails and calls to help stream line things a little more.

We still have in our business model the ability to have our wholesale provider take over our clients if we ever decide to get out of the industry or our investors to buy us out. Either way the current clients will keep their current rates and numbers.

Setting Up VoIP My Way. If you’re still with us, appreciating the risks, the rest is all good news. VoIP My Way’s documentation for setting up the service with either FreePBX and Asterisk or an OBi device is excellent. After receiving our credentials, it took less than 5 minutes to configure a trunk, inbound route, and outbound route. And, as mentioned, call quality is excellent. Here is the Trunk setup that worked for us with PBX in a Flash:

You’ll note that the Dial Rules support both 7-digit dialing through the default outbound route as well as prefix dialing to direct calls prefixed with a 6 to the VoIPMyWay trunk.

Finally, we want to close by wishing VoIP My Way the very best of luck in their business venture. We’ve contributed our $40, and our call volume on this trunk will help VoIP My Way’s bottom line. We promise. Hopefully, there will be at least a few more like us.

Originally published: Tuesday, March 22, 2011


Need help with Asterisk? Visit the PBX in a Flash Forum or Wiki.
Or Try the New, Free PBX in a Flash Conference Bridge.



whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.


 

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…