Post Tagged with: "sip"

The Gotcha-Free PBX: GVsip Gateway Service for Google Voice

The Gotcha-Free PBX: GVsip Gateway Service for Google Voice

Friday, April 3, 2015

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We promised you that free Google Voice calling in the U.S. and Canada would soon be available on every Asterisk® platform whether the platform supported Asterisk Motif or not. And today the first of two SIP gateway offerings has arrived. With this new service, you simply create a standard SIP trunk on your Asterisk server of choice, associate your Google Voice account with the GVsip gateway service, and PRESTO! You get secure OAUTH authentication to Google Voice without worrying whether… Read More ›

The Gotcha-Free PBX: Harnessing SIP URIs for Free Worldwide Calling

The Gotcha-Free PBX: Harnessing SIP URIs for Free Worldwide Calling

Wednesday, March 25, 2015

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View image | gettyimages.com We continue the Incredible PBX for Asterisk-GUI adventure today with a close look at SIP URIs, those email-like addresses that are the fundamental building blocks for VoIP technology. Consider this. If everyone in the world had a SIP address instead of a phone number, every call to every person in the world via the Internet would be free. That pretty much sums up why SIP URIs are important. The syntax for SIP URIs depends a bit… Read More ›

Midnight Madness: Introducing Incredible PBX 12 with Asterisk 12 and FreePBX

Midnight Madness: Introducing Incredible PBX 12 with Asterisk 12 and FreePBX

Monday, December 1, 2014

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#83329085 / gettyimages.com The number "12″ always has held mystical prominence in our culture and so it is with Asterisk®. Just over 12 months ago, Digium first introduced Asterisk 12 at AstriCon in Atlanta and heralded a major change in the direction of the product. It was more than a wholesale revamping of the Asterisk feature set. There was a revolutionary new development methodology thanks to the untiring efforts of Matt Jordan and his incredibly talented development team. Unlike Asterisk… Read More ›

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

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

Monday, January 13, 2014

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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… Read More ›

Newbie’s SIP Navigation Guide for Asterisk: Is It Safe?

Newbie’s SIP Navigation Guide for Asterisk: Is It Safe?

Monday, September 9, 2013

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It’s Back to School Time at Nerd Vittles today with a wrap-up of our series exploring the symbiotic relationship between SIP and Asterisk® including the most important consideration of all: SIP Security 101, a quick-and-dirty look at the security implications of using SIP with Asterisk. If you read nothing else before you begin your VoIP adventure, move today’s article to the top of your list. It might save you a personal fortune! Think of it as winning the lottery without… Read More ›

A Second Look at Grandstream’s UCM6100 Asterisk PBX & Some SIP Surprises

A Second Look at Grandstream’s UCM6100 Asterisk PBX & Some SIP Surprises

Tuesday, August 27, 2013

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What a difference a couple months make! For those that are keeping an eye on the UCM6100 Asterisk® PBX from Grandstream, we wanted to provide some additional insights based upon two firmware updates that Grandstream has released since the PBX was first introduced earlier this summer. The short version of this story is Grandstream has addressed most of the open source issues and they’ve resolved well over a hundred bugs. In addition, they’ve published excellent documentation on the PBX in… Read More ›

Practicing Safe SIP: Adding SIP URI and Free DID Connectivity to Asterisk

Practicing Safe SIP: Adding SIP URI and Free DID Connectivity to Asterisk

Monday, August 19, 2013

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Last year, we began our exploration of safe SIP options for Asterisk® by introducing a hybrid solution using VoIP.ms for a registered SIP trunk and IPkall for a free DID. Today, in addition to a free IPkall DID to accept incoming PSTN calls, we have a slightly different approach that provides a direct SIP URI address from Sip2Sip.info for your server. As with the original tutorial, today’s implementation preserves our Zero Internet Footprint™ design for total SIP insulation of your… Read More ›

The SIPaholic’s Dream Come True: Introducing Anveo Direct

The SIPaholic’s Dream Come True: Introducing Anveo Direct

Monday, May 6, 2013

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We’re incredibly happy with the current list of providers that we recommend to PBX in a Flash™ users for VoIP trunking. At the top of our list is Vitelity, a leading VoIP provider that has been a major contributor to the Nerd Vittles and PBX in a Flash projects for many years. But, as often happens, one of our gurus on the PIAF Forum comes up with a terrific discovery that we just can’t wait to pass along. This week… Read More ›