I love you too, Stank.
I hope you enjoyed our little time by there.
I'm Strom Carlson, and this is Black Ratchet,
and welcome to Be Your Own Telephone Company with Asterisk.
Apparently Alex thought that rated this a bullshit presentation, so...
Yes, we're very proud of that fact.
See, if Alex thought that we're actually here, we'd give him the payphone,
but I guess he's not, so you lose, Alex. Sorry.
So, let's get started.
Actually, before we get started, I just want to test the phone
to make sure that our dial tone's the right volume.
Alright, so let's get started.
Seems like I've got these cords in the right places.
So, first off, what is Asterisk? What is this damn thing?
How many of you have no clue what Asterisk actually is?
Sweet. You're in for fun. Lots of fun.
What is Asterisk?
Asterisk is a free, open-source PBX that runs on Linux.
It is the best thing since sliced bread.
How many of you don't know what a PBX is?
All my phones have their friends in the front.
PBX is a private brand exchange. It's a telephone system.
In case you hadn't guessed, it's a talk about telephones.
It was originally written by Mark Spencer of Digium back in 1999.
It's been upgraded significantly since then.
It's now version 1.0.8. It's a stable 1.0.
1.0.9 is a stable 1.0?
Yeah. Okay.
We just ran the latest bleeding edge version.
So, why Asterisk? Well, first off, it's free!
And that's the cool part.
A typical PBX for a business will cost you in the 5-digit range just for the hardware.
And when you set it up, it will cost you even more.
So if you can do this yourself, and this is very easy to do yourself,
it will save you a lot of money if you're running a company.
And if you're a hobbyist, it's really cool because you can set up your own phone network
and call each other for free, and we'll get into that.
It runs on commodity PC hardware.
In fact, this little tiny laptop that Ratchet is typing on right now
is our Asterisk PBX for the presentation.
There's broad support for VoIP protocols and hardware.
It's easy to interconnect with other Asterisk boxes and form your own network.
It's configurable to do almost whatever you want it to do.
So you can tweak it to your needs, you can write your own code to interface with things if you need to.
Unfortunately, it won't do your dishes, but it comes close.
We're working on the dish dial tone module.
So let's get into the hardware of Asterisk.
Asterisk will run on surprisingly out-of-date stuff.
If you go to VoIPinfo.org, there's a wiki there with all sorts of information on VoIP and especially Asterisk.
And one of the pages is this page where people post what they've gotten Asterisk to do.
And they stress test Asterisk on various hardware.
Someone running a 133 MHz Pentium 1 with 16 MB of RAM got 3 concurrent SIP calls going at the same time.
And any PC you have lying around will work.
Someone posted that they got a 2.4 GHz P4 with 512 MB of RAM to do 790 simultaneous calls.
So you'll hit your bandwidth limit far more quickly than you'll hit the limit of your hardware.
Here's a sample Asterisk installation. This is actually my rack behind my desk at home.
So here's my Asterisk box in the corner.
Thanks, thanks. I get told that a lot.
I beg your pardon?
Yeah, I did get the rack at Costco, actually.
So there's the Asterisk box chilling into my mail server, my web server, my NAS box, and everything.
There are Sun machines here. Asterisk is running on the PC, but people have gotten it to work on Sun machines running Linux.
And I think someone got it to run on Solaris running on Sun hardware as well.
So, yes.
They all work. We'll have a Q&A session at the end, so we'll do that.
There are many methods for connecting your telephones to Asterisk.
You can get either an IP telephone, or you can connect regular analog telephones to Asterisk.
There are IP telephones you can buy second-hand on the market. They're not all that expensive.
This is the Cisco 7960. This is an IP phone.
You can get those for about $250 online. In fact, I looked on eBay. I think I saw them going used for $150.
If you look for deals, I've gotten one for $120. That's as low as they go right now in current market conditions.
There's the... go ahead.
You don't want to know what I had to do for that.
There's the Polycom IP600, which is considered the other really good phone.
These two are considered probably the premium-deployed telephones.
And that's $250 online on eBay.
There are crappy telephones you can get. You can get the Grandstream Budget Home 100, used about $40.
It's roughly equivalent to that $9.95 phone you buy at the drugstore
that falls apart before you even get it home.
But it's a VoIP phone, so if you need a cheap VoIP phone, that'll work.
And right in the mid-range, there's the Snom 190, which is about $150, $175.
The prices have come down since we threw these slides together.
And, I mean, there are hundreds of manufacturers making phones for Asterisk and just VoIP in general.
So you can look around and find all sorts of stuff.
Most VoIP phones are built the same. We'll get into that, actually, in the protocol section.
So I'm actually going to ask you to hold your questions until the Q&A session at the end.
And we'll have a queue at the telephone over there for you to ask your questions via telephone.
You know the cool stuff's coming later.
You can also get little terminal adapters if you want to plug your analog telephones in.
You can buy one from Digium that speaks Asterisk's native protocol called EECS.
And that's called the EECS-E. And I think I have one here.
This is the EECS-E. A little tidy thing. You plug in an analog telephone in one end and you plug your Ethernet connection in the other end.
And your analog phone is now a VoIP phone. This is $100.
You can also get the Sephora SPA-2002. We've got those here running. And these are about $70.
You can plug two Vana Ratchet.
These speak SIF instead of EECS. You can plug two analog telephones into these and get two dial tones and that's about $70.
You can get the Grandstream HandyTone 286 for about $65. That only does one, I think, that speaks SIF.
And the old Cisco ATA-186, which is the one that Vonage used to send out, is about $50 online.
You can also... there's tons more of these. There's one... the adapter that Packet 8 used to send out, the DTA-310, you can find those about $30 used online.
And although you can't really get them working as they are, you can reflash the firmware and they become just generic SIF adapters.
There's details for that on the website.
You can also buy something called a Zaptel card. And these are cards you put in your asterisk box and you connect analog telephones and telephone lines to these cards.
The one on the top is the TDM-400P. That has four little modules on it.
You can select either FXS, which are dial tones, or FXO, which are telephone line interfaces, and put them on the card and arrange them any way you like.
The bottom one is a T1 card, and you can... there are two versions of FXO, 405 and 410P, and the 411 and 406 have echo cancellation built into them.
And so you can bring four primary rate ISDN interfaces into that and have 96 channels of voice going out of your asterisk box.
Or, you know, if you've got a whole bunch of analog phones, you can get a channel bank and plug the channel bank into the T1 for your stations.
So you can have 96 stations running off of one asterisk box with four channel banks.
So, what are you talking about?
Let's get into some of the protocols that asterisks use to talk to the network.
Those are really hard to find, in my opinion. I mean, I've looked for them because I thought they were really cool, but I could not find them anywhere.
I searched eBay for about a week, and I finally gave up in frustration.
Oh, that's cool.
Okay, for those of you who didn't hear, apparently the Linksys TAP2 is the same way as the Sapura 2000.
It can or can't, except for it cannot be flashed permanently.
Anyway, getting back to interconnecting asterisks, let's go adjust this.
The main protocol for interconnecting voice over IP calls is called SIP. It stands for Session Initiation Protocol.
It's a signaling protocol only. It only signals and sets up calls.
All it does is contact your box and say, hey, I have a good call for you. You might want to connect to it. It's on this IP address.
At which point, it connects to the IP address with a protocol called RTP, which stands for Real-Time Transport Protocol.
It was developed by the ITF, not ITU. It uses URLs instead of telephone numbers.
For example, you won't connect to 3-1-1-5-5-5-2-3-6-8. You'll enter it into SIP, stromcharleston.com, which doesn't work on Thread.
It's intended to be a peer-to-peer protocol. It's not like a general server. It goes to my box next to Strom's.
It doesn't have to contact anybody for it to work. There are certain caveats to that. We'll get into those later.
The main advantage of SIP is that it's very ubiquitous. It's supported by everybody. If you talk about VoIP, anything post-1998, it uses SIP.
Packet 8, Broad Voice, essentially all the major providers use SIP, with the exception of AT&T called Vantage, which I'll cover later.
It does not play well with NAT. It's very hard to actually set up with NAT. You have to start poking holes in your firewall for the RTP.
And, for example, if you start connecting out to another box, say I'm behind NAT and I try to call Strom, it'll say, oh, connect to my box.
I'm at 192.168.0.252, and Strom's loss of VoIP. So there's a lot of workarounds for it.
What you have to do is you have to tell it to, you know, this is my external IP address. Use this for everything.
Don't care about whatever your IP address is. However, in my opinion, it's a bit of a kludge.
The other protocol, before there was SIP, there was H.323. It was developed in 1996 by the ITU.
It's really far more similar to traditional protocols that are used by telephone networks, as opposed to SIP.
Like, for example, SS7 or something. It's more similar to that.
Again, it uses RTP for media transport, and it's used internally a lot by inter-exchange carriers.
And it's very unpopular in the VoIP world, because it isn't a fair setup.
And it's, like I said, a major pain in the ass to get it working correctly.
Hold on.
Although, by the way, although this is a telephone talk, please turn your ringers off. Thank you.
Come on, I figured let's discriminate her against telephone users.
We actually were hanging, Strom was hanging out in Pound Asterisk one day on IRC.3dome.net, for anyone interested.
And JerJer, who's one of the major contributors in Asterisk, made the following quote.
Just don't use H3.3, and all your problems will be solved.
Which, technically, is correct.
Now, for Asterisk, there's a main protocol, and as you can tell, it's based on Asterisk.
It's called intra-asterisk exchange.
It was developed by Mark Spencer of Digium, the guy who wrote Asterisk.
And it covers both signaling and media transport.
It's a very streamlined, very simple protocol.
What Mark wanted to do is he wanted to, you know, took a look at SIP, took a look at H3.23, and said,
Boy, these are crap. Let's go design a mining protocol.
It does not suffer from NAT traversal issues.
For example, I'm behind a firewall I call Strom, and, you know, it knows that where I'm coming from, you know,
it knows where to connect.
It doesn't actually, you know, say, hey, I have an IP address here, which could be wrong.
Data signaling happened via UDP on port 4569.
That's the only thing you have to open up in your firewall as opposed to RTP,
which I think you have to open about 15,000 ports for.
Very well supported by Asterisk.
However, the support in terminal equipment is very rare.
The only ETA that supports it is the ICSI by Digium that makes it.
I've also noticed that one phone recently started to support it.
It's called a NetWeb302.
It's actually made in China.
And the only place you can get it is someplace in Delaware.
Telephone.
Could you shut off the ringer, please?
I'm testing for later.
The nerve of some people.
It's a protocol for many PSTN termination providers because, you know, a lot of PSTN termination providers,
they use Asterisk.
It all works.
Just set it up.
It's fire and forget.
I think I set it up in one of my accounts about a year ago.
I haven't touched configuration settings.
It just works.
The other protocols are the Media Gateway Control Protocol, MGCP.
That's the other protocol I was talking about.
It's used by AT&T called Vantage.
I'm not sure if they still use it, but they did.
I bought an old ETA firm on eBay, and I had to reflash it and do all kinds of annoying things to actually get it working with Asterisk.
But the support is there.
The other thing is Cisco's Skinny Client Control Protocol.
It's used by Cisco phones.
This actually uses SIP.
This doesn't use SCCP right now.
You can choose with the Cisco phones to use either SIP or SCCP.
They're different firmware images.
Not all Cisco phones support SIP.
For example, I have an older version that was made in the early 2000s, and it only uses SCCP.
There's support for it.
It's not the best in the world, but it works.
I mean, I can only have one line on the phone, and three-way calling is an issue.
There are two different SCCP implementations for Asterisk, and they both kind of suck.
Chan SCCP2 doesn't suck as bad.
But that's kind of like being the tallest midget.
Yeah, exactly.
Since Strom is the Kodak Nazi, we'll just sort of choose this one for him.
I'm a Kodak Nazi now, in addition to being a spelling and grammar Nazi.
It's a compliment.
So anyway, basically, there are many different codecs that you can use to encode voice with Asterisk.
And we're going to go over what they are, what they do, and what they sound like.
In order to understand codecs, you need to understand how digital audio works.
How many of you don't really know how digital audio works?
Okay.
Well, I don't want to see too many hands go up, so I'll kind of go over this quickly then.
Basically, you have an analog waveform, and this is divided up into a number of little slices of time.
So you have an analog bunch of different slices of analog audio.
These are pulses of analog amplitudes called pulse-amplitude modulation.
So you've got time division, but it's still analog.
What happens is these values are then quantized into digital values.
This is called pulse-code modulation.
PCM is what you use when you talk on a telephone, on a regular telephone call.
And it's also what you hear when you play a CD.
So what happens is the values are quantized between 1 and negative 1.
So depending on how much resolution you want to give to the values, you can specify more.
The way mu law works, what mu law is, is the companding scheme that the analog telephone network uses.
Whenever you make a regular long-distance call or a regular local call, you're using mu law.
It's 64 kilobits per second, and what happens is it's a logarithmic scale.
So more resolution is given to the lower amplitudes, where most of the voice frequency is,
in order to give you the impression of 16-bit audio.
What actually happens is it's only 8-bit audio, but it sounds far closer to 16 because of the companding.
But however, for some people, 64 kilobits per second is still way too high for bandwidth.
They're greedy, or they have less bandwidth, and they want to be able to send phone calls over less bandwidth.
So there are many different ways of compressing this already compressed audio.
Adaptive differential pulse code modulation, well first, differential pulse code modulation,
uses 4 bits to describe the change from the last value.
What mu law does, and PCM does, is it actually describes each value individually.
This one it says, you know, go down 4, go up 2, and so on.
Regardless of the original source resolution, regular DPCM just uses 4 bits.
Adaptive differential PCM uses a varying number of bits, depending on the complexity of that sample,
but it's still 32 kilobits per second, so they have to scrim somewhere.
It sounds almost like mu law, but not quite.
There's also LPC, and this is the basis for a lot of what you find in cell phone codecs.
It's linear predictive coding, and it basically, it's pretty complex,
but the basic thing is that it uses vocoders to compress speech.
And it tends to work pretty well for speech, but there's a lot of electronic music nowadays
that has the vocoders that have the singing synthesizer effect in the song.
So, actually if you listen closely to talking on a cell phone, you'll begin to hear that.
And you can actually use all these codecs with Asterisk and VoIP.
So, again, voice on the PSTN is 64 kilobits per second, synchronous bandwidth for wireline telephones.
North American uses mu law companding, most of the rest of the world uses a-law companding,
which is almost identical, but slightly different, but for all intents and purposes, they sound exactly the same.
If you're in 64 kilobits, if you're doing something called bit-probing,
which is basically to give you supervision in band on one of the digital channels, you get 56 kilobits per second.
But for the most part, it sounds the same.
On mobile phones, you only get 4 to 13 kilobits per second for your voice.
Which is why mobile phones sound so horrible.
So, speech compression does have a lot of costs.
First off, if you're running it on your Asterisk box, you have to transcode from mu law to the codec.
And in order to be able to hear it, you have to transcode back from the codec to mu law.
So, there's an increased CPU load that's required.
That box that does 790 simultaneous calls, if you have it do GSM or one of the lower bandwidth codecs,
I think that drops down to maybe 100 simultaneous calls.
There's no guarantee that two pieces of equipment will speak the same codecs.
A lot of times, the Seporas speak mu law and then G.726, I think.
And then some of the other codecs.
But, yes.
But, you know, I think we're going to have to keep track of whether this is on or off.
Thank you.
But I think the EECSY only speaks mu law.
And there are things that just speak certain codecs, and it's a mess.
I mean, there's no guarantee that any pieces of equipment will be able to talk to each other unless you use mu law.
Some codecs require licensing.
Oh, and one thing I forgot to mention.
If you're using non-standard fit rates for some of these codecs, then you're really in trouble.
Some codecs require you to pay licenses to use them.
There's a codec that... actually, we'll get into that in another slide.
And codecs don't handle all kinds of sounds well.
People will generally have trouble understanding certain words.
A friend of mine says that every time he calls up a restaurant to make a reservation, they cannot understand the word north.
It's always interpreted as something else from his mobile phone.
No one ever gets north right.
It's difficult to understand anyone who has poor diction, because these are designed under the assumption that people who are using them tend to enunciate well.
And music on hold in codec land is absolute torture.
You can gouge your eyes out if you really want to.
I mean, I want to do that every time I'm on hold on my mobile phone.
It is that bad sometimes.
There's one we'll demonstrate that is that bad.
There's a documented case in Arkansas.
Apologies to anyone from Arkansas here.
Here's the benefits. Each call uses less bandwidth.
That's it.
I wish there were more, because that slide looks really empty, but that's it.
Here's what Asterisk supports.
G.711, which is 64 kilobit per second ULA or ALAW companding.
G.726, which is 32 kilobit per second ADPCM.
G.729, which is 8 kilobit per second CSACELP, which is Conjugate Structure and Algebraic Code Excited Linear Prediction.
That requires a license from Digium to purchase.
We're going to demonstrate that codec later.
Just in order for me to be able to call Ratchet's box and have one telephone conversation over the codec,
we each had to pay $10.
That's only for one call, so if you want to have more than one call using this codec,
$10 a call for the licenses.
Not per instance.
Yeah, simultaneous calls.
So if you only buy five licenses, you cannot have more than five simultaneous calls running on that codec.
So if it were $10 a call, I think no one would use this thing.
It would be like being back in those days.
Or if it was developed by Microsoft.
Here's the Internet Low Bandwidth Codec, which is a 13.3 kilobit per second linear prediction coding.
This is what Skype uses.
I mean, I've heard people rave about Skype. I hear it. I hear it just like ILBC.
There's also Speaks, which is an open source codec. I think it's based on like the Ogg Vorbis stuff.
And that's also 13.3 kilobit per second codec-sided linear prediction.
And finally, for those of you who are on dial-up, there's LPC10, which is 2.4 kilobit per second.
It sounds more ghastly than you can possibly imagine.
Yes, that's the one.
So in order to do a comparison of all these codecs, we figured we'd play some audio for you using each of the codecs.
And so I decided, you know, I could do just talking, but that would get boring after about 25 seconds.
So as proof that you can find anything on Craigslist, I found this song that this band let us use for free.
So apologies if it's not up to your musical standard or your musical taste, but it's free, so be happy.
I like it.
I like it, too.
It gets stuck in my head every time I hear it, and it's really annoying when he uses this version.
Yes.
Wait, so let me, um, let me cue it up here.
Okay.
If they ever get a recording contract, buy their album. We promise we'll give them a bargain.
Yeah, we promise.
Also, if you really like the song, I think there's a copy on your asterisk, not on your asterisk, on your Defcon CD.
Also on your Defcon CD, there's the full copy of that song in every codec we just demonstrated, so you can listen to them and compare for yourself.
We won't be responsible if you listen to the whole thing in LPC 10.
Yeah.
So let's talk about getting Asterisk to talk to the real phone network, because this is all fun and cool,
but it's going to be kind of limited in use if you can't talk to, you know, your mobile phone or your mother or somebody on the street.
There are various providers that, um, that you can use to terminate the public telephone network, and we're going to go into one of them.
It's called Nuphone. Nuphone, uh, well, you know more about Nuphone than I do. Why don't you tell them?
The plus about Nuphone is that they have cheap rates, not as cheap as some other voice-over ID providers, but pretty decent rates.
They're very geared toward Asterisk. I believe they support this only under a nice public high-pride,
saying, you know, I want to be assisted by, you know, a supporter or some other ADA.
One of the things that I like about them, and at least my experience with my first provider, is that you can set your caller ID outcomes,
which has some interesting, you know, applications to it. I'll cover that later.
Um, and also the really cool thing about Nuphone, and I think it's their best feature, is they have insanely cheap provisioning.
I never thought anywhere in my life I would ever have to find 800 numbers for my name, but I do.
I, seriously, I click them and they send you an email saying, here's your 800 numbers.
You can actually do vanity 800 numbers for them, like if we wanted to buy 800 Asterisks, I don't know how many characters it is.
I've forgotten how many 800 numbers I own.
Anyway, it takes a little longer for that, and they charge you a fee, but, you know, for any old 800 number that you want, it's just, you know, poof, it's yours.
They're very easygoing. When I first set up my account with them, I'm not sure how they do that, but when I first set up my account with them,
I said, oh, you know, we tell you, you know, we send you some money, what happens when that money's out, are you going to shut off my service?
He said, no, we'll just send you a reminder email, you know, we'll just, you know, send it off.
You won't turn off your service or anything, we'll just send you some money.
They also have something called caller ID number for delivery.
How many people are familiar with SS7 and the difference between CPN and CID?
Oh, good amount of people do.
Anyway, CPN is sort of like caller ID on steroids.
It's a little more speedy than caller ID.
You can actually do 9 times out of 10 unless you know what you're doing.
CPN is usually very free-flow and you can't change it.
Of course, you know, you're probably laughing at me.
It's, caller ID on steroids is a little more free-flow and it's harder to mistake it, theoretically.
They also have, quite recently, they have a blocker on deletion projects.
Now, for the longest time, what would happen is, you know, like,
your call would not go through or you'd get an intercept saying, like,
the number you've been calling has been changed, so and so.
It wouldn't actually work like that.
And for someone like me, who's phone-free and enjoys calling intercepts,
it was a bit of a bummer.
But, you know, a couple weeks ago, oh, a couple months ago, by now,
they actually turned it on.
I called with a payphone out in Boston and I got, you know,
I'm sorry, the phone number you've reached is not accepted.
I literally picked up the phone and called him and said,
who funds this proper conversation?
And he was jumping out like a sleight of hand.
He was too gross on the net.
It was very exciting for me.
CONSTED, directed for dial numbers, which are basically numbers you would call on,
you would set up on the PSTN, so mom, dad, you know, your sister,
someone who doesn't have an asterisk can call you and actually reach you on your woodline.
The only numbers they have are in Michigan, which for me, I guess, is okay
because no one really calls me on woodlines besides him.
And they're also not too freak-friendly, despite being very easygoing.
For example, during Strong and Lucky 55's presentation last year on freaking in the age of voice over IP,
was it?
Yeah, I think it was something like that.
They sent them a cease and desist letter saying, no, they didn't send me a cease and desist letter.
Who did they send a cease and desist letter?
Was it not there yet?
It's not there yet. Stand up.
Here's Mr. Cease and Desist.
This is the man who's on Jeremy McNamara's shit list.
He basically said, you know, if you do this, I will terminate your account and ban you for life from New Phone.
I haven't heard a peep out of him.
Okay, we're just going to...
Oh, they stopped carrying CPM this week.
Oh, great.
This is what we did.
We used to like Astrolink until this weekend.
Astrolink, for the most part, for me, has been reliable, although this week has been a whole other bag of forms which we'll get into later.
You can get your inbound termination via toll-free numbers.
And, of course, AMI-II. How many of you do not know what AMI-II is?
AMI-II is the class of service.
So you can tell whether your calls are coming in from an AMI phone or a mobile phone or a pay phone, what type of pay phone.
You can see whether your calls are coming in from a person or two.
And Astrolink does also proper call completion.
So if you get the PPM at the port and your numbers have changed, you can do that.
The con of the AMI-II interface is that it's a little bit clunky.
It's actually gotten better. I was actually able to use it with Firefox for the first time this week.
But I had to try it for a while, so it didn't quite work right with Firefox.
And right now, at least, they're experiencing each two different problems with some customers, namely us.
This is why we don't like them anymore.
But we'll get into our beef with Astrolink later.
There's also Voice Calls Connect, which I use to do my inbound calls.
They give you a local numbering in almost every center you want.
You get unlimited internet minutes on inbound EECS calls.
And you have inbound numbers for a large number of great centers.
And it's also called a proper call program.
The problem is, it's one of the most expensive EECS providers for outbound information.
New Phone is about two cents a minute.
Astrolink is also two cents a minute.
Voice Calls Connect is, I think, it used to be 2.9 cents a minute.
Then they dropped it down to the low, low price of 2.4 cents a minute or something like that.
Wow, that's not ideal.
I guess when you go landline long distance, it's five cents a minute, like my inbound call.
There's something else you can totally keep.
When you've got providers that do 1.3 cents a minute, you're getting better.
And the other thing is that inbound numbers with Voice Calls Connect,
my number that's on my business card is a Voice Calls Connect number.
And for some reason, usually I have to go to a menu.
You know, thank you for calling for strong cards.
My PBX is strong.
I like to be clear.
I like to show you that I appreciate what you're doing.
And I had someone trying to call me, and they kept pressing 2, and it would just hang up on them.
Because my PBX never heard the 2.
So I had to make a move directly to my mobile phone this weekend.
Apparently, this is the first sentence.
This is the first sentence that happened to me, but some people are having major issues with it.
So if you're really unlucky, you're totally screwed.
Here's VoiceJet.
It's really cheap.
It's 1.3 cents a minute.
Actually, I think it's 1.29 now.
Oh, is it?
Wow.
A whole mil when they dropped it.
Hey, it adds up.
Yeah, it does.
It does.
Caller ID delivery is unreliable with VoiceJet.
You oftentimes get the number where I found the call, instead of the number that you set it to.
There's no e-mail service that's appropriate, and there's no proper call completion.
If you call a number that's been disconnected, and the phone company that you're giving it to says,
this number has been disconnected, all you'll hear is the phone ringing and ringing and ringing.
So you'll never know why your mother has changed her phone number in your situation.
She doesn't love me!
Exactly, see?
It's because we all use LPCs now.
There's also a company called Broadvoice.
They have cheap PIDs in most race centers.
They're pretty much run by phone freaks.
In fact, when you sign up, you have them send you the kit with the ATA and everything.
They also send you this little pin, which is this logo,
with the red circle and the slash through it.
I actually found out this weekend they also do T-shirts.
Yes, at least one person here at DEF CON is walking around with a Broadvoice pin,
with the Bell logo and the slash through it.
So if you hear that person, please stand up and get some T-shirts.
He's not here? What the hell?
They lose.
You lose.
We were going to give you just like $100,000.
They were going to get the payphone, too.
Oh, and the other cool thing is that you get a caller ID with name for inbound calls.
Most of these other carriers will just deliver the number,
but these people will actually do a name level and deliver the name for you.
So we'll see not just that 311-555-2368 is calling you,
but that it's from Mabel's fish restaurant or something.
The problem is that it's SIF only.
So if you want to do EECS termination or if you're trying to firewall or something,
you've got to do some sort of EECS proxy
if you want it to be reliable.
They're prone to service outages.
I've got, you know, my dial type is the 1-plus-8-4-7 number.
It just goes over broad voice.
But sometimes those things can work.
Phone support is kind of slow, although it is 24-7.
So I guess if you call at, you know, 3 a.m., they will answer like that.
No, they won't.
Yeah, I try.
They'll have to wake up first.
Then they'll, you know, drink their coffee.
And then the other thing is CNN delivery is also unreliable.
So, you know, caller ID name tends to work, but it doesn't work all the time.
So now let's get into building your asterisk network.
Why don't you explain enum, Ben?
Okay, Brandon.
Enum was the first way to actually translate code numbers over to URLs in SIF.
It's based on DNS.
It allows you to read numbers and query.
And what it does is it basically bypasses the public switch telephone network.
So, you know, I know that 3-1-1-5-5-5-2-3-6-8 is a voice number.
How can I do that?
I don't know.
So the type I can use is supposed to be used by telecom companies.
However, there's a program out there.
I think this is the largest one.
I may be wrong.
It's called 2-1-2-3-4.org.
It's a free do-it-yourself solution.
What you do is you register your number in 2-1-2-3-4-5-5-2-3-6-8,
and you register it to a URL.
And then someone queries their enum server, then, over the Internet.
But then you realize that local exchanges can't access 2-1-1-5 numbers,
and how many exchanges you've got can't access 2-1-2-3-4-5-6-8.
I think in an area code, there's a maximum limit.
We've always reserved 3-1-2-3-4-7-2-2 million numbers.
And Los Angeles has how many area codes now?
So, a lot.
They also have 78 million special PSTN services.
So you're essentially...
you can offer free termination of $800,
which is basically a PSTN termination service.
You can use this, and you can dial in your numbers.
How e-mail works...
I left it in my car.
Everything's in my car.
Okay, if we all pay attention to the top,
basically, I'm at the top of my phone,
and I want to dial 3-1-1-5-5-5-2-3-6-8.
Oh, yeah.
Touchpad, marvels of modern technology.
Yes, latest thing in Virginia.
I'm at this telephone.
I basically want to dial 3-1-1-5-5-5-2-3-6-8.
So I pick up my handset, dial it,
and talk to my DVX.
It says, okay, I want to dial 3-1-1-5-5-5-2-3-6-8.
Connect me there somehow.
So what I set up my DVX to do
is talk to a medium circle.
And it says, hey, do you know
what 3-1-1-5-5-5-2-3-6-8 is?
And it says, yes or no.
And essentially, in this example,
it says, yes, it's a bell at telephone.com.
So it registers that to my DVX,
at which point my DVX goes,
ooh, I guess I'll go over to the internet.com
rather than picking up a local line.
And it goes over the internet,
and I dial 3-1-1-5-5-5-2-3-6-8
down here, which they also
don't do at 3-1-1-5-5-5-2-3-6-8.
They're also completely bypassing the BSDN.
Email is a very...
has a lot of problems in my opinion.
It's a very top-down way
of looking at things.
It's centrally managed, centrally served,
and centrally centralized.
Basically, they wanted developers
exactly like DNS and have, you know,
15 root servers all over the world,
which is great, but no one actually
has done it yet.
And it's not being used
by any BSDN developers.
As far as we know.
Does someone want to tell me otherwise?
My idea is how
logically they save you money, you know.
I can charge you 5 cents a minute
for, you know,
and it's nowhere near
where it would last.
We're working towards a really nice solution
that's starting to get there, but at the same point,
you know...
Something like that, yeah.
So last April,
what someone set out with
is a fully peer-to-peer
E164 solution.
It's a fully peer-to-peer
E164 solution.
Basically, you know,
it sets up your own
telephone network for fresh
for your telephone company.
You know,
5, 5, 5, 1, 2, 3, 4.
So,
we've got this DBS here,
and it talks to this node over here,
and what this node does,
it only has one node,
and what it does is
it reaches that node,
and that node talks to only this one,
and it reaches the DBS node,
it's got a dial,
3, 5, 5, 2, 5, 6,
at which point,
it scrolls back and says,
hey, 411 service,
this is all it.
Basically,
it's a very good
peer-to-peer setup,
which has some problems with
Dundee, in my opinion, also.
To be honest, I mean,
essentially, what I could start doing
is I could start advertising, hey,
all the numbers to the White House,
send them to me,
so I can, you know,
start World War III,
scalability,
we're not sure how it's going to be
still relatively small,
but I've never seen a product
like Nutella.
Look how many problems we had with that.
It's not officially a standard yet,
we just submitted it to IETF,
but as we all know, standards cost money,
and these people don't have it.
Right now, it's only officially
in DBS,
and we can submit it,
and run it against 109,
or 107, or whatever,
and get some support in it,
and in my opinion,
it works fine,
and also in my opinion,
I think the eye of the beholder
should be nice.
Quality assurance,
what quality assurance
do you charge
for your
for your phone call
going out over your WAN,
for example,
I don't have quality assurance
for my WAN,
or for example,
for my WAN.
There's two ways of doing it.
Can we do this?
No, no.
Okay, we'll do it that way.
Sorry, I forgot to dial
something in my PBX earlier.
Basically, there are two ways
you can use QoS here.
There's priority QoS,
which can give you a way back
to priority over any other
factor,
where you can basically
reserve your WAN
and turn it away
and get the rest of it for it.
But for everything that's not QoS,
you get the rest of the WAN.
I'm sure there's more
about that, but I'm a phone person,
not a router person.
Many residential routers are now voice aware
and will do the sort of thing right out of the box.
And you can also
if you're a control freak,
or if you want to do it on a larger scale,
then you can connect your router
and say, hey, I've got a super router phone
and I've sort of gotten working
and I'm definitely not a super router person.
So now let's get into the
cool part of what you can do
with Asterisk.
And by the way, if you're watching on TV
in the hotel,
we've got a special thing coming up for you.
Asterisk Gateway Interface.
Do you want to talk about that?
Experts on certain subjects?
Yes, Tag Team, exactly.
So,
Asterisk Gateway Interface.
What it is, is an interface for adding
functionality to Asterisk.
It basically uses standard in and standard out.
And it's very cross-language.
You infer, you see, you PHP,
Lisp, PCL,
anything that uses standard in and standard out,
you can use it.
And basically, since Asterisk is the best thing
since sliced bread, I find AGI to be the best thing
since sliced bread.
Because you can basically do anything you want with it
given enough time.
Here's a simple AGI for a minute.
Wow!
What it is, is basically,
I don't know if you can actually see it
up on the screen, I apologize.
What it does, is basically...
It's not in your Dendy notebook?
Oh, great.
This is what
apparently electricity parts
don't work.
So, basically,
a lot of my presentation of this part
hasn't been done.
Starting from here,
basically, what it is, is
start on a standard in point,
standard in and standard out.
Basically, what Asterisk does is
it reads all the variables in
and checks all of it.
It checks to make sure, 200 times per day,
5 times per day,
that we're not sure about the quality
and it means that
you just keep on going.
Basically, what happens is
after you read all the variables in
and make sure they're OK,
what it does is, it streams a file
that says, your caller ID is...
and it says,
you recheck your caller ID.
So, here's how it works.
I've backed up here,
but the one we're really interested in
is AGI caller ID, which says 311-555-2368.
And then it says,
after it's done feeding all the variables,
it says 200 results here,
it means I'm done in 41 minutes.
It says, OK, stream file,
you know, AGI caller ID.
Then it says,
OK, I'm done, what do you want me to do?
OK, stream pages,
311-555-2368,
AGI command that asterisk understands,
and it goes,
you know, says 3, 1, 1,
200 results here,
OK, I'm done. And at which point it says,
hang up, I'm done.
It takes it back to me,
sends back, you know, 200 results here,
I'm done now, it hangs up.
Yes, exactly,
AGI caller ID readback, ENAC.
Now,
that's a lot of work to do
in what's called asterisk AGI,
which is a Perl module for asterisk
in writing AGI scripts.
It simplifies a lot of AGI programming.
Essentially, it takes out a lot of the dirty work
and takes away everyone's doing the work
that you don't have to.
It allows the AGI to control
the simple object interface in Perl.
The problem is,
I subscribed to the mailing list
a few months ago,
and it's been dead silent
for the last few months ago,
and someone said,
is this mailing list still active?
At which point I'm going to be
not able to work anymore.
Maybe I should set up another thing,
or maybe a few people
should sign up to it
and start using it yourselves.
What it is, is it allows
the AGI to easily integrate with Perl,
which easily integrates
with almost anything in the known universe.
And it's available at
asterisk.uniter.net
which is a very strange German word
for like the longest time.
Here's the same program.
As you can tell, it gets a lot easier.
I think we can test this out at least 100%
to about 10 lines or so.
So this is, you know,
it reads in, you know,
starts up a new object
called AGI.
You know that entire, you know,
10 line thing, check the variables,
make sure they're okay.
It does it all automatically.
And the other feature that the other
one didn't have is you did not check
whether or not the person had a caller ID
in the first place.
So you basically, if you didn't have a caller ID,
you did a file file saying,
the caller ID is, and then it would hang up.
Or what this one does is it checks
to see if your caller ID is, you know,
what this would do if you didn't have a caller ID,
it would just hang up on you.
And as I say, wow, it's easier to do.
It checks to see if your caller ID
is on touch-tone.
Basically, everyone knows how to use it,
including your mom.
However, you know, it's very limited
on what you can do with it.
For example, you have to use the sign-up function
to use it, or, you know,
you can only use it if you have five members.
The XML, it's extensible.
WebPXML is extensible.
VoiceXML is basically an XML
that allows for scripting
with a voice interface.
It's extensible.
The problem with Asterisk is that it has no
relatively new open-source program
called Asterisk PDF, which says
it has Asterisk CIFS.
So I spent about two weeks
trying to get this compiled.
It's a high-resolution
demonstration.
There's also
commercial offerings. For example, Microsoft
has a new VXML program
where there's, like, at least two or three
commercial ones that I know of that
are also free
open-source.
VXML is still, you know,
you need to use.
Think back
to that Texas speech program you played with
in, like, 1987.
That program is called
Cepstral.
Cepstral is a free
free-for-download
free-for-download program
that they've also announced
however it's not free.
Something like that.
You can play with a
free version of Cepstral, but
it's also a sound file
and every fifth or tenth word or something goes
please register me.
Yeah, and it is free
for download, like you said, and you can make sure
if you ever have, like, you know, one of the programs
and you want to make sure that it works with your program
you can set it up and, you know, you just have to do
that please register me every time it works
or something.
The other thing for output is
reports. You can get your stuff for free.
The other person is Allison
and she has huge support
for Asterisk.
It's for pay and she has lots of
fan sayings available on the internet.
Asterisk also comes with a whole ton of sayings
and she says all sorts of crazy stuff
in the LA area.
A friend of mine actually
just wanted to test her
and so he sent her
increasingly offensive scripts
and I think he finally found her
a little bit after...
yeah, a long time.
Well, maybe she'll read
Richard Nixon's Social Security number,
I don't know.
Basically,
we have a...
I'm not a weather
fan, my name is Ron.
Phone!
I'll show you
my phone voice in a second.
And now
a demonstration by Scott Carlson
that we actually do.
Good afternoon.
Thank you for calling DEF CON My Phone.
Script getting tested,
Blamer approved.
To hear what's coming up in the next hour,
press 1. To search for an event,
press 2. To need feedback from
DEF CON My Phone team, press 0.
And so on.
If any of you
want to write down silly things on slips of
paper and
leave them up here,
I'll do them for you.
Here's a slightly more complex
thing, this actually uses
input and output.
Basically, what it does is
I've got three ways of doing it.
One of those is
using CGI, and basically
it lets you forever
wait for you to enter a file.
Wait for you to enter a digit.
Wait for it to say
who's entered a digit.
And what it says is
after you enter a digit,
wait for five seconds,
and if you look at one, it says
it has five words on it.
If you enter two,
it has seven words on it.
If you enter three, it has a thousand words on it.
This is really cool.
Intermission!
Okay, so we have
some cool swag to give away.
GGM has seen fit to give us
props. Apparently, Charles has
become very friendly with Mark Spencer,
and he sent us
very friendly
So GGM sent us this whole box
of really cool stuff. There's these little
CDs with asterisks on them, and they're in the
shape of an asterisk.
There's asterisk stickers.
Yes, if you put
the sticker on your car, more people
will want it.
I also have
bin racks here, because an old-school
freak.com sticker isn't good.
I have two very
kick-ass sites. You see some of this
in forums almost every day.
Asterisk T-shirts.
So I need four volunteers who want to
come up here and dial phones.
Actually, no. Let's have them answer
questions.
Name a pre-divestiture Bell
operating company.
Pre-divestiture, not 9X.
Pre-divestiture?
Pre-divestiture.
Operating company.
GTE was not a Bell company.
I'm taking over now. Name a post-divestiture
9X regional Bell operating company.
For example, someone already said it.
Raise your hands, people.
Come up.
So, let me wind this thing out of the way.
Each of you sit in front of a telephone.
So, let me prop up this list in front of you here.
This is a touch-tone speed dialing contest,
and for some of you who were in the
game area earlier, we kind of did this
just to work the bugs out.
So, basically, I have a list of numbers here
that I've programmed into my PBX,
and the person who dials these numbers
the fastest and most accurately
will bring my desk phone here
and get a prize.
So, let me set this list up here.
Yeah, I'll explain the rules in just a second.
Unfortunately, these don't do pulse dialing,
otherwise we would do a switch-hook
Can you all read that?
Pardon? Sir, can you read that?
Okay, so what you're going to do is
you're going to pick a phone. Each phone is dialed
one plus the area code plus the number.
After one second, if you dial correctly,
you'll hear another dial tone, dial the next number,
and so on and so on.
If you miss-dial a digit, say you dial
two instead of one, you'll get a reorder tone
after one second, at which point
you will want to dial the number again.
If you skip a digit,
you'll hear nothing after one second.
You may want to press a couple digits
until you get a reorder.
If you get a reorder that
you can't break by dialing,
if you get the reorder where you can't redial the number,
there's rules, there's rules.
If you get the reorder that does not go away
when you start dialing digits,
you screwed up and you have to hang up and start the beginning again.
So,
ready,
set,
go.
Beg pardon?
He wins.
The guy in the Quest shirt.
What size t-shirt do you wear, sir?
What size t-shirt do you wear?
Large? Okay.
He gets to leave.
We need one more contestant.
The guy in the Bell shirt.
You want to switch phones, Natas?
Okay.
We like preferential treatment.
So,
do you understand the rules, sir?
Okay.
On your mark,
go touch the handset.
Don't pick up the handset until I say go.
You can touch the handset.
One plus the area code plus the number, yes.
On your mark, get set,
dial.
Good news, everyone,
I found the numbers.
You can get them in the
capstoneextensions.com.
You know,
Natas, you're really starting to make these sounds
that I know you're...
You flash the hook
and get a call, all right?
He put his own call on hold.
That's what we're...
Yeah, let's do some
bid revenue.
Let's get some free stickers, too.
One of you still got a call on hold.
Call, please.
Natas, would you like an old-school free sticker?
Would you like an old-school free sticker?
He runs the site
for those who don't know.
And what size T-shirt do you wear?
Excellent.
Let's do this. Let me plug the
telephone in so we can hear the winner.
I think we'll do one more,
and then we have to get on with the presentation.
Let's make this fair.
Let's clear this out
and have four all-new people.
If you guys do compete,
come up to me after the show. You'll get swag.
You'll get swag.
You'll get swag. You're welcome.
Okay, new rule. Don't flash so fast
that you put a call on hold.
Yes.
Okay, so,
let's get some volunteers. How about this guy?
You made some funny comments.
Okay.
I heard a patch.
Yes.
Okay, we have one more.
I thought that was my line, Ratchet.
Okay.
Okay, do you guys understand
the rules from last time, or do I need to repeat them?
Okay, yes.
Okay, and now you're able to
listen to your caller.
So, on your mark,
get set,
dial.
Okay.
Don't leave now.
If you want free stuff, don't leave now.
We have various telephonic pieces of equipment
from Star Collection and more asterisks
after the show.
Good afternoon.
Cheers.
Oh, a princess phone.
A princess rotary?
It's little, it's lovely,
but, however, it does not like
what it's a consumer sale model
made out of.
Digium Zap-Tel cards do support rotary phones.
In fact, you can navigate
IPR menus of rotary phones with a Zap-Tel card.
However, the only thing that does
support rotary dial are the card
and the EC.
I think the EC supports rotary dialing.
I'm not sure.
It does? Okay, cool.
So, yeah, this thing does support it.
Unfortunately, no.
This one actually belongs to Mark Spencer
and I had to give it back to him.
Anyway, I found my two
complex, my slightly more complex
and my caller ID.
So, here we go.
Caller ID 1.
This is slightly more complicated.
It sounds like it's going to your AIS script.
I actually replaced
all of the default sounds with JMR
because we used a lot of the
Dell System and Output back in the 1960s
but I really like it more.
You can hear some
slightly more complex script
but I'm going to avoid it for now.
No, baby, give it to me.
Yes, yes, that's it.
And option 3.
This says...
Enter your object now.
Enter your...
This says...
Enter...
This is what's from the
sound file.
I ripped it off from the
information course.
Thank you for making it here.
Send your caller ID to whatever
you want.
After you send your caller ID...
Initial address message.
It was made by
Automatic Electric, which was
GT's manufacturing company.
But it's now discontinued as far as I know.
And ironically, Automatic Electric
is now part of this.
But anyway, you only find the GT5s
typically in former GT territory
and some independent
companies' territories.
However, in the GT5,
Automatic Electric
is an all-around unique
network.
So they decided to test out
each other's networking one day
through this home phone and
decided to list your friends.
I mean, how many times did you
have to call?
You need to create
your neighbor's credit card.
You find your neighbor's credit card
and pay him back.
You don't know where you're calling from, right?
You're obviously
using the same
phone you found your neighbor's card.
I'm not going to take your number.
You just send any caller ID
you can find, any CPM
number.
Oh, you must be selling something to connect
Let's get into the technical details of this later.
You can also, however, do color-ID spoofing most of the time with two lines in your extensions.com file and ask for a CPS.
So, I'm just going to grab some data for you.
We have a lot of theories about color-ID spoofing.
So, I have some.
It's based on the cost of your own number, of some number you want to talk more about.
And then you'll telephone the company, and they'll send you whatever name is associated with that number.
Yeah, it depends on what database you're using, but for the most part, it's fairly easy.
I think I've got a photograph of me doing that.
So, you can see how it works for yourself, your local exchange, your local website.
It's called Remembering a Missing Database.
And then they send that name to you.
It allows you to, if you're doing scanning, you can prioritize the more missing phone numbers.
Here are some real color-ID names.
We've got a company called C.I.A.
It's a really, really cheap reverse lookup.
And you can figure out stuff for yourself.
I'm straight.
I love you strong.
I love you too.
Here's an example of me backspoofing.
You need your credit card, personally.
This time, if it didn't work, you go to Google.
SuperColorID is something that I hacked up in a matter of about five minutes.
It extrapolates tons of useless info from a telephone number.
It gives the name and address from YPH.com.
And it does get you information from the Telecordia location that you've got.
In my office, it works on the telemedicine device, telemedicine terminal.
But if you've got some screen or some window you want to look at, you can do that too.
This is a version of it that has a site, oldschooltrees.com, that doesn't use the word.
That just grabs the address from the web.
Here's a photo of me using SuperColorID,
and looking from Gladstone's Corned Fish restaurant at Pacific Palisades,
to my own phone number.
Yes.
Wow, people are making their own versions for their own country.
Someone's got an Amazon copy from Australia, that works in Australia.
So, sweet stuff.
So you can see here,
it does the number,
the address,
so this spins up your webpages,
this is the calling party name, that tells you who's the phone company,
and here is all the details,
how the solution is served out,
how many information, prefixes,
where the switch is, what time the switch is,
and, for example,
if it's useless for anybody but their own country,
you can just enter, like,
Apple, or Google, or anything like this.
What?
Um...
No, it does not.
You can rig radio podcasts.
Yeah, why don't you explain this?
What radio does is,
they have a hunt group,
that's what it was,
you call one number, and it hunts for a free line.
So, for example, you call 5-8000,
and the reality is,
it hunts for a free line,
which is, you know,
8-000-1, 8-000-2,
8-000-3,
8-000-3,
8-000-3,
8-000-3,
8-000-3,
8-000-3,
8-000-3,
8-000-3,
8-000-3,
8-000-3,
8-000-3,
and what happens is,
after the hunt is over,
I think our provider for
Defcon by phone claimed that we can have
probably over 600 calls
going on at the same time.
We do have a DS1,
otherwise known as a T1,
and a DS3,
and a DS4,
and a DS5,
and a DS6,
and what it basically does,
you tell Asterisk,
generate a bunch of
different radio stations,
to connect to a certain radio station.
So, what happens,
call file one,
sorry I'm call number two,
call file three,
sorry I'm call ID number three,
now I've got it set up to work,
and it leads out bound calls,
and you can do more,
generate call file number four,
and actually stuff gets done,
call number five, call number six, call number seven,
and you're looking at call number nine,
and all of a sudden,
the calls that are currently in the queue
after the call is over,
you say, hey, oh my god, I won,
and I say yes,
and all of a sudden,
I'm sure radio station guy number one
is going to be highly confused,
and he says, well, they don't know
what's going to fix this,
that's why I actually found something out this week,
and they solved the problem,
and I'm not going to tell you what's going to happen.
Yeah, American Idol,
or some, if you watch TV,
I don't watch TV anymore,
and I said, well,
gee, this is important for the voting
mind, for the worst guy that was ever
listed on the list,
and I said, well,
if you do that, and they get a record contract,
radio will get about 1.1% worse.
What's your
NMAT5 voting number? Let's just do this.
On your
exchange, right?
Enter the IP address
you wish to scan.
Oh, that's it.
You know how to push my button.
Please wait.
Host found with
17 ports open.
Echo on port 7.
Disguise on port 9.
Date time on port 13.
Threat on port 17.
Curtain on port 19.
SMTP on
port 25.
Name server on port 14.
Domain on port 15.
HTTP on
port 80.
And that's it.
On Friday,
just fixing bugs and
dealing with things.
You don't have to.
All of him,
he thought of all the phrases
except the recording.
After recording everything,
I got really crazy and recorded
a whole bunch of crazy introductory phrases.
Okay, so,
you're basically ignoring how weird it is
to see phone calls
and recordings of yourself.
It still freaks me out.
So, there's a search
to be able to ignore calls,
there's a reminder, and again,
you need to register.
How does it work?
There's a MySQL database,
and there's a whole bunch of audio clips.
And pretty much,
I spent a week in front of my computer
dictating the entire text
of the schedule.
There's registration,
reminders,
and there's a web interface
internally.
To hear a listing of shows
that are coming up in the next hour,
press 1.
Please enter the first 5 letters
of the title of the show you'd like to see,
followed by the talent sign.
Presentation ID 56.
Be your own telephone company.
With Asterix.
Presented by Strom Carlson and Black Ratchett.
In Parthenon 3 and 4.
Starting at
Saturday, July 30, 2005
at 7 o'clock p.m.
Running for
110 minutes.
To hear more about this event,
press 1.
To skip to the next event,
press 2.
If you hear more about the event,
it brings you back to the entire description of the talk
as printed in the schedule.
Shouts to Strom!
If any of you want to hire me
for your voiceovers, please come see me.
I'll give you my contact information.
But don't give me 48 pages of show descriptions, please.
Prefer sanity.
No, I think I can handle that.
It's cheaper than buying that damn stuff for us.
The problem?
Dictating the whole schedule is a pain in the ass.
I didn't realize how long it would take
to dictate the whole thing in.
We were rushing.
It was supposed to be less than a week ago.
I'm not used to it.
I'm not used to it.
We had some jitter issues last week.
There were so many problems.
We couldn't nail all calls at the same time.
And we had jitter problems.
We had to switch to SIP.
And it's really difficult to keep our schedule in sync
with DEF CON's schedule.
We went and asked them to please
give us all your scheduled updates
to inform our users
and everyone who calls in about changes.
I don't remember
getting my email address
out of the entire agenda.
So thank you, everyone.
And also this.
We came up to them and said,
hey, our schedule really sucks.
Is there someplace we can go to get updates?
Oh, go to updates.defcon.org.
So I was like, okay, I'll sit down, fix everything.
Not a single schedule change on there.
I was...
It's causing nightmares.
I don't have nightmares for you.
We spent all of Monday, or all of Friday
trying to deal with the fact that they pushed
the entire track date back an hour.
But we hadn't anticipated that.
I'm sorry?
What we should do next year
is we're probably going to set up something
where the schedule is actually
thrown together in our system.
And that's the idea I had.
The schedule is thrown together in our system
so we automatically know about it
and have everything synchronized.
And as a disclaimer, we all love DEF CON.
We're not trying to piss anyone off,
but this is what happens.
When we're at DEF CON reminders,
there's no way to detect whether users
are rejected by scale.
So they don't get thrown into the reminder.
Some users are brain-deficient
and forget their thing.
And you've got to call from somebody.
You know who you are.
Rejector.net.
We'll be back next year.
We'll be back next year.
The problem with Crack 2 is that
they said, we'll move everything up and down,
OK, that's fine. I update the schedule.
And they said, OK, let's make that time
so everything is on schedule by the end.
I cried when I finally found out.
There was no way to do that.
I want to get to the Q&A
because I'm sure there are a lot of you
who wrote the Q&A.
The CDF cards kind of flake out.
They're problems because asterisk is TTL
but all folks know how to do the asterisk
So, if you want to ask a question, there is a telephone there.
Queue up in front of the telephone.
And for those of you watching on TV,
call the following number, area code 213-415-1047,
and you will be in the Q&A queue.
Do not call this number if you are sitting here in the room.
If you see someone who is on their phone and calling the number
and holding and asking a question, grab their phone and throw it to me,
and we won't give it back to them.
Will you please pick up and dial 8-1-1-0?
Oh, shoot. Sorry about that.
Please pick up that phone.
Hello?
I have a question.
Really?
Okay, please, everyone calm down. We are trying to get to make questions.
The codec that was demonstrated earlier, you can do rock and roll, funk,
with a lot of bass, and a yelling guy,
and a very quiet dude with his band going.
The band was really good to get that music, like,
I don't know if codecs are nice for speech, but the way I see it,
if you portrait test a codec, you see what they really sound like
under all circumstances.
Because not all speech is perfect.
So if you portrait test a codec,
you get a clearer picture of what it really sounds like.
Yeah, it's a good test. I know it's not designed for music,
but there is music on hold.
What happens when you're driving in your car,
and you have tons of background noise, and you're trying to use one of these?
Same thing.
Okay.
Thank you very much.
Thank you.
Let's see if anyone's calling in from the TV.
Agent logged in.
Hello?
Hello?
Black call!
Let's hang up and get the next person in line.
Agent logged in.
Calling in? Wow!
More people at home!
If you're watching on TV call, okay, let's get you in.
Okay, two questions, actually.
One, for all the color-ID speech and stuff you guys were talking about,
do you need a DS1, an MP3, or a PN?
No, I've got it running on my DSL connection.
Okay, so just with the VoIP provider...
Yeah, your standard VoIP call, even with MULA,
only requires 80 kilobits per second of bandwidth plus overhead.
So if you've got more than 80 kilobits per second connection in both directions, you're fine.
Okay.
Okay, never mind.
The other question I had is, what is the difference between a phone company and a telephone company?
How many phones do you want to use?
You can get generally about the same as a phone company, between three and five telephone sets.
Run, Forrest, run!
Isn't this scavenger hunt over yet?
No.
There's a giant bee problem with DEF CONNORS here.
Thank you.
Hello, you're with Asterisk.
Hey, how you doing?
Hey, how you doing?
Hey!
There's a little delay.
What's your question, sir?
My question on DEF CONNORS.
Are those the only ones that are supported?
For interfacing your phones with Asterisk?
Yeah.
No.
No, they're not the only ones supported.
I believe you're...
Yeah.
No, they're not the only ones supported.
I don't know which one it is.
I think it's something called VoiceTel that uses the same hardware.
It was the same older version of it.
It's supported.
It works.
But...
Good luck getting it working.
So, are those PCI cards?
Yes, they're PCI.
And I believe there's also one with ISA for you to use with...
I believe you're running...
Okay.
...DEF CONNORS.
DEF CONNORS, right?
Yes.
This is actually on the DEF CON by VoiceTel.
But we're going to move on to the next question.
Okay, let's get the next question from a real person.
Marty?
Hey, could you...
For the most part, Asterisk is about as secure as a regular telephone,
although the new version of the ETH protocol now does encryption.
I haven't played with it yet.
It's only about two weeks old.
But there is now encryption built into the TVS head version of the ETH protocol.
Cool. Okay.
Let's take a call from...
Hello?
Oh, this was always right here.
The reason those payphones take an incredibly long time to call...
Also, it works from my mobile phone.
So it's not our...
The other fun thing about the payphones out in the lobby
is that they do not transmit a payphone code.
So therefore, you can actually use DEF CON by phone from them.
All payphones are banned because we don't feel like being charged a fee.
However, thanks to Goldetel, whoever supports them,
they have a very unincorrect standard on their payphones
and they treat them as being real payphones.
Hello, caller?
Grab his phone!
...and O'Reilly makes great books.
This one's apparently like 570 something pages.
I'm looking for a good book with a starfish on the cover.
And secondly...
Sure, I'm broke. I'll accept the money.
You won't pay me. I've got it down for you.
Alright, thank you.
Call in somebody from...
Hello, party?
Hello?
Hello.
Up to four people can be in the queue at any given time.
And how did you figure out that you're the inventor?
Because we're dialing our internal Dundee net number.
We got skills, yo.
Yeah. Actually, you can set a password for this, but for convenience,
since none of you from the outside world can dial on our Dundee plan,
we figured it would just be easier if we could just answer the question
instead of getting some password thing.
Yes, essentially, this allows you to set up your own phone sets.
Yes. Thank you much.
I'm wearing...
I've got this really, really nice butt set.
Have you played with any of the 802.11c voice phones?
I haven't played with any of the 802.11c voice phones.
There are phones that'll act like cordless phones, except they'll talk to your Wi-Fi network.
I wasn't kidding when I told you I had a really nice butt set.
7920? Okay, so 7920.
If you go to voice-info.org wiki, they have information on all of those.
I haven't played with the Wi-Fi phone set.
Yeah, yeah, you can do voice talk on BDSU.
Yes, I set them up for my own thing.
I have a nice Bluetooth headset. I have Wi-Fi hotspot. It comes right in hand.
We've got to go wrap this up. Thank you all for coming.
I'm going to send out shouts to binrev.com.
You can find Strom and I there on forums, binrev.com.
Anything else? Oldschoolfreak.com.
We have stickers that we can give people. Just follow us out.
We'll talk to your questions and do whatever.
Hang on just a second. You can also, before you go,
if you have any questions and you can't get to us directly,
my telephone number is on my website, so call me up there.
Further reading of resources, on to the slideshow on your TV.
Thank you for coming, everybody.