Search Results for "sip" : 502

Finally a 100% Portable PBX: Introducing GoIP, a SIP-GSM Gateway for Asterisk

Finally a 100% Portable PBX: Introducing GoIP, a SIP-GSM Gateway for Asterisk

Monday, September 30, 2013

— 15 comments

How far we have come! The original Asterisk® claim to fame was its ability to interface with proprietary phone systems and legacy telephony hardware, the glue that literally kept companies stuck to their overpriced PBXs. And, just as wired phone systems began to lose their edge, along came the Bell Sisters to introduce cellular communications with billing that began when the phone started ringing and an end to toll-free calling and extra fees for text messaging on top of exorbitantly… 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

— 1 comment

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 ›

Practicing Safe SIP: Adding SIP URI Connectivity with a Zero Internet Footprint

Practicing Safe SIP: Adding SIP URI Connectivity with a Zero Internet Footprint

Thursday, October 11, 2012

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PBX in a Flash™ has a long (safe) history in the VoIP community, and the major reason is that we constantly preach never directly exposing any ports on your Asterisk® server to the Internet without implementing a WhiteList of safe IP addresses. This Zero Internet Footprint™ design keeps everybody out except a trusted, defined group on your WhiteList. For everyone else, they never see your server. So how do you receive calls? You do it with phone numbers (DIDs) tied… Read More ›

YATE in a Flash: Rolling Your Own SIP to Google Voice Gateway for Asterisk

YATE in a Flash: Rolling Your Own SIP to Google Voice Gateway for Asterisk

Monday, June 25, 2012

— 18 comments

A few weeks ago we introduced you to Bill Simon’s SIP to Google Voice Gateway featuring YATE. This let you set up a SIP connection to your Google Voice accounts in about 5 minutes by filling out a simple web form. Today, we take it to the next plateau for those who prefer to do it yourself. With a little assistance from Bill (about 99% of the brainpower behind what you’re about to read), we’re pleased to now offer you… Read More ›

5-Minute VoIP: Deploying a SIP to Google Voice Gateway

5-Minute VoIP: Deploying a SIP to Google Voice Gateway

Monday, June 11, 2012

— 14 comments

Today we’ll walk you through the 5-minute setup process to configure a SIP gateway for any Google Voice number. Once in place, you can make and receive Google Voice calls using not only your 10-digit Google Voice number but also through any SIP device in the world.

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