DEAL Millions of Free SIP Calls

wardmundy

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This week's Nerd Vittles article documents a SIP setup that can be integrated into any Asterisk platform to provide literally millions of free calls in and out of your PBX. Pay particular attention to the sections on Linphone clients, sipbroker.com, and iNUM plus the hundreds of worldwide PSTN gateways that both services provide. You and your friends/relatives may never pay for a personal phone call again. And it's pure open source code!

Meet Linphone: Free Worldwide Calling to Anybody with SIP

 

wardmundy

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Actually, we've already set up a Linphone SIP server a.k.a. flexisip. If you'd like us to set you up a SIP account, PM the following to me privately:

Full Name
SIP URI: [email protected]
Password desired
Can your SIP URI be shared here on the forum? Yes/No

This will give you a SIP URI that can be used to receive and make calls exactly like the Linphone SIP server documented in the Nerd Vittles article. (we hope)
 
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mainenotarynet

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Mr Ward, didn't Incredible do this (provide sip names for users) way back when iNum or ITAD was out? - I seem to remember getting one years and years ago. I have/had an ITAD (1380, I think) but that has long been dead (at least the website is) I have an INum too (voip.ms), that may be another "directory" for us also.
 

kenn10

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The bad thing about setting up a directory for SIP URI calling is that the telemarketers will seize on this to get around the new STIR/SHAKEN technology the phone companies are putting in place. When you can make SIP calls for under a penny per minute, do we really need this technology? SIP URI calling is a back door for sleazebags to annoy us.

One side issue with the new STIR/SHAKEN technology will be that you cannot send a number across a SIP trunk that is not registered on your account with that provider. While it will stop masquerading as someone else, it will also limit how we set up our PBX's. to use multiple SIP providers for outbound calling. I suspect it may well prevent sending an outside caller's caller-id to a cell phone on find me/follow me arrangements. We'll have to see how the new technology evolves.

Free is great for the poor (who probably don't have the money to buy the technology to use it) but a bigger boon for the sleazy robo-callers and telemarketers. I don't want that to see that happen.
 
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wardmundy

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I would never recommend creating a public SIP URI directory. I was thinking more along the lines of a private list for a limited number of folks that have been regulars here for a very long time.
 

wardmundy

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Despite a good old college try to implement the Flexisip server from the CentOS-packaged RPM, we've thrown in the towel and will leave it to Linphone to provide the server platform. Hopefully, theirs is more stable. Zero server support anywhere that we could find. Flexisip is dual-licensed just like Asterisk, and they keep solutions close to the vest (unless you pay) just like the early Asterisk days.

We were chugging along all fat, dumb, and happy thinking we had a solid implementation, and it did actually work which is pretty amazing given what we discovered. When we turned on the journal, we were met by an endless stream of what you see below. The referenced Linphone database is nowhere to be found in the config file so this is obviously hidden in the source code. Pulseaudio is a monumental train wreck as well. No clue at all on the crashes. So the journal log grows by roughly 500MB per day.

We now plan to leave Flexisip running and use the platform as a honey pot and nothing more. And, boy, is it a doozy of a honey pot. The SIP attacks in the transaction log almost match the journal in size and frequency with no simple way to block them. The log structure rules out Fail2Ban. Lots of creeps in the world from our favorite countries. So we're gonna pass on this one.



p.s. We love the Linphone clients and Linphone's server platform. So life is good!
 
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