Post Tagged with: "sip"

The New Gold Standard: Incredible PBX 13-13.10 for Raspbian

The New Gold Standard: Incredible PBX 13-13.10 for Raspbian

Monday, March 11, 2019

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Today we are pleased to introduce the 2019 update for Incredible PBX® and the Raspberry Pi® featuring 70+ new FreePBX® GPL modules and a native Skyetel SIP trunking platform with a $50 service credit if you sign up before the end of March. In addition to dozens of under-the-covers tweaks, there also are new backup and restore utilities which should ease the pain of backups and future migrations. In fact, today’s build was created using those tools because the image… Read More ›

Keep On Trunkin’: Free International VoIP Calling Returns

Keep On Trunkin’: Free International VoIP Calling Returns

Monday, February 25, 2019

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Today we’re taking a fresh look at the international calling marketplace by updating the best VoIP deals available. FreeVoipDeal once again takes the prize with the best selection of "free" international calling destinations at the lowest prices. Below we’ll provide a quick tutorial to transform your Incredible PBX server into an international calling platform at minimal cost. Here’s How It Works. For every 10 euros ($10.72) you deposit into your account, you’ll get 300 minutes a week of free calls… Read More ›

Adding SIP URI Dialing to Asterisk for Free Worldwide Calling

Adding SIP URI Dialing to Asterisk for Free Worldwide Calling

Monday, February 11, 2019

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Since giving up on Google Voice, we’ve been extolling the virtues of SIP URI dialing which gives you unlimited free calls to anyone else in the world that happens to have their own SIP URI address. SIP URIs look very much like email addresses except they’re used to share phone conversations instead of email messages. And, as we’ve mentioned previously, if everyone in the world had their own SIP URI, paying for phone calls would become a thing of the… Read More ›

SIP Happens! Deploying a Publicly-Accessible Asterisk PBX

SIP Happens! Deploying a Publicly-Accessible Asterisk PBX

Monday, January 28, 2019

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We’ve previously documented the benefits of SIP URI calling. Because the calls are free from and to anywhere in the world, the use case is compelling. The drawbacks, particularly with Asterisk® servers, have primarily centered around the security implications of exposing SIP on a publicly-accessible server. Today we want to take a fresh look at a possible SIP implementation for Asterisk based upon the pioneering work of Dr. Lin Song back in the PBX in a Flash heyday. We’ve embellished… Read More ›

SIP Happens! And Kamailio Makes It Easy, Part I

SIP Happens! And Kamailio Makes It Easy, Part I

Monday, January 14, 2019

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If ever there was a Swiss Army Knife for SIP, Kamailio (a.k.a. OpenSER) is the hands-down winner. The flexibility of this open source SIP server is legendary. Whether it’s secure communications, insulation from brute force attacks, load balancing, failover, WebRTC, or the return of shared line appearances on your office phone system, Kamailio can handle it while processing thousands of call setups per second on minimal hardware platforms. Our plan for today is to walk you through setting up a… 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 ›

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 ›