Post Tagged with: "voip"

A Sobering Look at Asterisk and the 2019 VoIP Landscape

A Sobering Look at Asterisk and the 2019 VoIP Landscape

Monday, December 17, 2018

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Every six months or so we like to gaze into our crystal ball for a quick look at the VoIP landscape. 2018 has been quite the transformative year with the acquisition of Digium® and Asterisk® by Sangoma®. Unfortunately, as we predicted, the Digium layoffs have already begun, and 2019 may only get worse. While we have no inside information, we wouldn’t be surprised to see Digium’s headquarters in Huntsville closed within six months in an effort to balance the books.… Read More ›

Spam Phone Call Blocker and CNAM Caching for FreePBX

Spam Phone Call Blocker and CNAM Caching for FreePBX

Monday, November 26, 2018

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Blocking spam phone calls has been a challenge to put it charitably. Thanks to some earlier work by Stewart Nelson on the DSLR forum as well as Stewart’s considerable hand-holding in the development of today’s tutorial, we want to introduce a new approach to blocking these calls. The way it works is first time callers that pass the TrueCNAM SPAM check will be prompted to "press 5 to connect." Since most spam calls sit in a queue for several seconds… Read More ›

R.I.P. GVSIP: A Final Farewell to Google Voice

R.I.P. GVSIP: A Final Farewell to Google Voice

Friday, November 16, 2018

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It’s been a death by a thousand cuts, but today marks the end of the Google Voice era with Asterisk®. Since Google removed XMPP support and transitioned to their new GVSIP platform, many have held out hope that Google hadn’t moved to a purely commercial platform with their ObiHai deal. Yesterday, the head of the Google Voice project requested that all Asterisk GVSIP implementations be discontinued citing Google’s Terms of Service. We hinted this was coming back in July and… Read More ›

Free Asterisk Voicemail Transcription with IBM Watson STT

Free Asterisk Voicemail Transcription with IBM Watson STT

Monday, November 12, 2018

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There are many commercial voicemail transcription services for Asterisk® PBXs, but none hold a candle to the speech-to-text (STT) quality of the IBM Cloud offering known as Watson® STT, formerly known as Bluemix TTS. Despite a recent price increase that takes effect in December, the pricing remains competitive. On the Standard Pricing Plan, voicemail transcription is 2¢ per minute. Or you can try things out on the LITE plan which offers 100 minutes a month at no cost. When the… Read More ›

FusionPBX on Steroids: Text-to-Speech Apps Have Arrived

FusionPBX on Steroids: Text-to-Speech Apps Have Arrived

Monday, September 24, 2018

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And you thought you needed an Asterisk® PBX for your users to enjoy FREE text-to-speech applications such as current News Headlines and Weather reports from the convenience of their telephone. Well, move over Asterisk. FusionPBX™ for FreeSWITCH™ now offers virtually identical functionality with all of the terrific advantages that FusionPBX provides: reliability, updates, performance, security and an unmatched UC platform with no rivals. To get started, make sure you have completed the steps in our FusionPBX introductory tutorial. Intuitive support… Read More ›

Back to School: Introducing FusionPBX for FreeSWITCH

Back to School: Introducing FusionPBX for FreeSWITCH

Monday, September 3, 2018

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It’s been quite a week with the surprise acquisition of Digium® and Asterisk® by Sangoma®. It became official on Wednesday, September 5. You can read all about it here, and you can read our cautious optimism here. As with the recent Google Voice transformation, we hope it serves as a gentle reminder to the VoIP community not to put all your eggs in one basket. With the start of the new school year, we could think of no better time… Read More ›

Double-NAT Blues: Tackling Asterisk’s Thorniest Problems

Double-NAT Blues: Tackling Asterisk’s Thorniest Problems

Monday, August 20, 2018

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Whether you’re new to VoIP technology or an Old Timer, nothing is quite as frustrating as wrestling with one-way audio and no audio on SIP calls either because of poorly designed NAT-based routers or poorly implemented SIP ALG solutions on low-end residential routers. To make matters worse, you get to deal with calls originating behind not one, but two, NAT-based routers neither of which complies with the basic SIP Rules of the Road. In a perfect world, SIP and RTP… Read More ›

VoIP 101: Developing a Cost-Effective SIP Strategy

VoIP 101: Developing a Cost-Effective SIP Strategy

Monday, June 11, 2018

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In the lead up to the demise of Google Voice XMPP service next week, we wanted to offer what we have found to be a cost-effective SIP strategy which takes advantage of the best of all worlds. We would divide SIP offerings into five broad categories: business-class unlimited SIP trunks, Old Faithful SIP providers, Mom-and-Pop SIP services, dirt-cheap termination services, and Gee Whiz SIP providers. As we have said many times, the beauty of setting up an Asterisk® PBX such… Read More ›