Post Tagged with: "sip phone"

The 5-Minute Wonder: OpenSIPS Server Takes the Cake

The 5-Minute Wonder: OpenSIPS Server Takes the Cake

Monday, May 13, 2019

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We covered Kamailio in our Part I article. And we’ve skipped writing about SIP server contestants two, three, and four because they each had a healthy dose of insurmountable problems… at least for us. So today we’re pleased to present Part V in our SIP server series. And, as the headline exclaims, with OpenSIPS we’ve found a platform that finally is worthy of your attention. Our requirements were fairly straightforward. We wanted an open source SIP server to which we… Read More ›

Meet Linphone: Free Worldwide Calling to Anybody with SIP

Meet Linphone: Free Worldwide Calling to Anybody with SIP

Monday, April 29, 2019

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Earlier this year we demonstrated how to set up a publicly-accessible Asterisk® server to enable free worldwide calling using SIP URIs which are email-like addresses for VoIP and video calls. But not everyone has an Asterisk server so today’s tutorial extends free calling to everyone with a Windows or Linux PC, a Mac, or any smartphone or tablet. All you need is a desktop computer with wired or wireless Internet access or, on a smartphone or tablet, a cell data… Read More ›

Interconnecting a Mobile PBX to the Asterisk Mothership

Interconnecting a Mobile PBX to the Asterisk Mothership

Monday, April 22, 2019

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The Holy Grail for a mobile VoIP solution is a simple way to connect back to your primary Asterisk® PBX via Wi-Fi from anywhere in the world to make and receive calls as if you never left. Let’s tick off the potential problems. First, many home-based PBXs are sitting behind NAT-based routers. Second, almost all remote Wi-Fi connections are made through a NAT-based router. Third, chances are the remote hosting platform blocks outgoing email from downstream servers such as a… 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

— 3 comments

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 – replaced

SIP Happens! Deploying a Publicly-Accessible Asterisk PBX – replaced

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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SECURITY ALERT: https://securityboulevard.com/2019/06/rce-using-caller-id-multiple-vulnerabilities-in-fusionpbx/ 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… Read More ›

Back to School: Introducing FusionPBX for FreeSWITCH

Back to School: Introducing FusionPBX for FreeSWITCH

Monday, September 3, 2018

— 12 comments

SECURITY ALERT: https://securityboulevard.com/2019/06/rce-using-caller-id-multiple-vulnerabilities-in-fusionpbx/ 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… Read More ›

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