Search Results for "sip" : 502

Best of Both Worlds: Safely Marrying Asterisk to OpenSIPS

Best of Both Worlds: Safely Marrying Asterisk to OpenSIPS

Monday, May 20, 2019

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Last week we introduced OpenSIPS, the multi-functional, multi-purpose signaling SIP server which can fulfill almost any communications function one can dream up except the unified communications tasks typically performed with a PBX such as Asterisk®. Today we want to marry the two platforms to give you the best of both worlds. For Incredible PBX® users, the primary advantage of adding an OpenSIPS front end is the elimination of the complexities associated with interacting with your PBX from remote sites with… Read More ›

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 ›

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 - 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 ›

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 ›

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 ›

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