Post Tagged with: "freepbx"

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 ›

Taking a Page from Asterisk: How Far We Have Come

Taking a Page from Asterisk: How Far We Have Come

Monday, July 22, 2013

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We’ve never written about paging technology before, and this is one of those areas of VoIP telephony where it certainly paid to wait. What a difference a few years makes! At least in the Asterisk® context, SIP-based paging traditionally involved issuing a Page command with a list of extensions in your dialplan. The wrinkle was that each VoIP phone manufacturer had its own SIP header to trigger autoanswer on its phones. And, without autoanswer, paging becomes next to worthless with… Read More ›

Triple Treat: Some Asterisk Utilities to Brighten Your Summer

Triple Treat: Some Asterisk Utilities to Brighten Your Summer

Wednesday, June 26, 2013

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[purehtml id=12] If you live and breathe Asterisk® but don’t visit the PIAF Forum regularly, you’re missing one of the best VoIP resources on the Internet. To get everyone in the Independence Day mood, we thought we’d share a few of the new goodies that have appeared on the PIAF Forum since The Great Crash of 2013. Although each of these utilities was designed to support PBX in a Flash™ and Incredible PBX™ systems, with a little tweaking, they’ll work… Read More ›

Here We Go Again: Getting Ready for the Next Google Voice Train Wreck

Here We Go Again: Getting Ready for the Next Google Voice Train Wreck

Monday, June 3, 2013

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Self-inflicted wounds are nothing new in the technology business, but Google spent much of last week working hard to take top honors for what is clearly one of the most selfish and short-sighted moves ever in the telecommunications marketplace. Less than a week after extolling the values of open source technology during Google I/O 20131, Google wasted little time performing a complete 180 by deep sixing further support of the open source XMPP protocol for messaging and VoIP communications. With… 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 ›

It's Baaaaack: Skype Returns to PBX in a Flash with Asterisk 11 and FreeSwitch

It’s Baaaaack: Skype Returns to PBX in a Flash with Asterisk 11 and FreeSwitch

Friday, April 26, 2013

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It’s been a long road, but we finally have a reliable Skype™ implementation for Asterisk® 11. Ironically, it uses components from ArchLinux™ as well as FreeSwitch™. But, hey, it works! It sounds great. And it lets you talk (for free) to over a half billion of your closest friends around the world. So today we present a Skype solution that works with a full-featured PBX. It can run on any Windows®, Mac®, or Linux® Desktop using the PIAF-Green™ Virtual Machine… Read More ›

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