Nerd Deals:

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

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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 ›

The 5-Minute PBX: Introducing PIAF-Green Virtual Machine for VMware ESXi

The 5-Minute PBX: Introducing PIAF-Green Virtual Machine for VMware ESXi

Tuesday, September 24, 2013

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In our never-ending trek to build the Perfect PBX™, we have another installment for you today featuring VMware’s just released vSphere Hypervisor 5.5 (ESXi). And, yes, there’s still a free ESXi version with a free license available here. But, unlike VirtualBox, you’ll need a dedicated (beefy) server on which to install ESXi. Be sure to register on the site and obtain then install the unrestricted license, or you’re SOL after the short eval period. We’ve built an ESXi appliance which… Read More ›

Fall Festivus: Asterisk Text-to-Speech Roundup with a Baker's Dozen New Voices

Fall Festivus: Asterisk Text-to-Speech Roundup with a Baker’s Dozen New Voices

Tuesday, September 17, 2013

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[soundcloud url="http://api.soundcloud.com/tracks/110813303″ params="" width=" 100%" height="166″ iframe="true" /] It’s been a while since we looked at the text-to-speech (TTS) landscape with Asterisk®. So we thought we’d bring you up to date on where we stand with text-to-speech in the free department insofar as Asterisk 11 is concerned. The hands-down winner remains Lefteris Zafiris’ implementation of Google TTS. It rivals any commercial TTS software application and costs you nothing. The good news is, if you’re running Incredible PBX 11, you already… 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 ›

The Music Frontier: Taming Streaming Music on Hold with Asterisk 11

The Music Frontier: Taming Streaming Music on Hold with Asterisk 11

Tuesday, September 3, 2013

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It’s been over 7 years since we first wrote about streaming music on hold with Asterisk®. While we’re energized with Back to School Fever, we decided it was about time for a refresher. And, in honor of TWOfer Tuesday, we also have a terrific new SIP discovery to share. It won’t cost you a dime. For long time readers of Nerd Vittles, you will note that all of the MOH syntax has changed since the early Asterisk days. So today… 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 ›

2013 Greatest Hits: Lenny Returns for an Encore Performance

2013 Greatest Hits: Lenny Returns for an Encore Performance

Monday, August 12, 2013

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Nothing in the VoIP community this year quite captured the hearts and minds of geeks around the world like Brian West’s "Lenny." For anyone that’s ever been dogged by obnoxious telemarketers or that’s had to deal with less than lucid tech support inquiries, Lenny was a godsend. Finally, we all had a place to send those poor souls while getting our daily chuckle listening to the results. If you’re late to the party and missed all the fun, then start… Read More ›

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