Search Results for "sip uri" : 350

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 ›

Free Worldwide VoIP Calling with SIP URIs and Issabel 4

Free Worldwide VoIP Calling with SIP URIs and Issabel 4

Thursday, August 24, 2017

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SIP URIs make the VoIP World go ’round. They’re the email-like addresses that carry VoIP calls between SIP servers to reach their destination. But there’s gold in them hills if you know how to use SIP URIs because SIP URI calls are free even if the calls travel all the way around the world. We previously documented how to deploy SIP URI calling with PIAF5 and 3CX, and today we’ll show you how to make SIP URI calls from and… Read More ›

Integrating SIP URIs into XiVO for Free Worldwide Calling

Integrating SIP URIs into XiVO for Free Worldwide Calling

Monday, September 26, 2016

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It’s been a while since we’ve explored SIP URIs and all of the advantages that SIP URI calling brings to your PBX. Number one on that list is FREE calling to and from anyone on the planet so long as both of you have an Internet connection with a SIP phone or a VoIP server such as Incredible PBX for XiVO. SIP URIs are the fundamental building blocks for VoIP technology. Consider this. If everyone in the world had a… Read More ›

The Gotcha-Free PBX: Harnessing SIP URIs for Free Worldwide Calling

The Gotcha-Free PBX: Harnessing SIP URIs for Free Worldwide Calling

Wednesday, March 25, 2015

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View image | gettyimages.com We continue the Incredible PBX for Asterisk-GUI adventure today with a close look at SIP URIs, those email-like addresses that are the fundamental building blocks for VoIP technology. Consider this. If everyone in the world had a SIP address instead of a phone number, every call to every person in the world via the Internet would be free. That pretty much sums up why SIP URIs are important. The syntax for SIP URIs depends a bit… 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 ›

Practicing Safe SIP: Adding SIP URI Connectivity with a Zero Internet Footprint

Practicing Safe SIP: Adding SIP URI Connectivity with a Zero Internet Footprint

Thursday, October 11, 2012

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PBX in a Flash™ has a long (safe) history in the VoIP community, and the major reason is that we constantly preach never directly exposing any ports on your Asterisk® server to the Internet without implementing a WhiteList of safe IP addresses. This Zero Internet Footprint™ design keeps everybody out except a trusted, defined group on your WhiteList. For everyone else, they never see your server. So how do you receive calls? You do it with phone numbers (DIDs) tied… 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 ›