 I love you, Dewstank. 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.Net created this bullshit presentation. So we're quite proud of that. Yes, we're very proud of that fact. See if Alex.Net were 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. So let's get started. It's like at 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. For less of fun. What is Asterisk? Asterisk is a free, open-source PBX that runs on Linux. It is the best thing since slice of bread. How many of you don't know what a PBX is? All my phone friends in the front. PBX is a private branch exchange. It's a telephone system, in case you hadn't guessed this 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 is the stable 1, right? 1.0.9 is the stable 1, now? Yeah, OK. Is it? Do we just run the latest bleeding edge version? So why Asterisk? Well, first off, it's free. That's the cool part. Typical PBX for a business will cost you in the five-digit range, just for the hardware. And for the setup, it'll cost you even more. So you can do this yourself. And this is very easy to do yourself. It'll save you a lot of money if you're running a company. And if your hobby is 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 get interconnect with other Asterisk boxes and perform your own network. It's configurable to 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. Yet, we're working on the dish dial tone module. I think, pardon? 2.0, yes, or 2.0. 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 Asterisk will run on surprisingly out-of-date stuff.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. They stress Asterisk on various hardware. Someone running a 133-megators penny in one with 16 megs of RAM got three 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 gigahertz P4 with 512 megs 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. You can do, you know, this is 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 got told that a lot. Are we pardoned? 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 Ron 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 secondhand 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 now. 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 VoIP 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 phony buy at the drugstore that falls apart before you even get it home. But it's a VoIP phone. So if you use a cheap VoIP phone, that'll work. And right in the mid-drains, 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 asterisks 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 Q at the telephone over there if you'd ask your questions via telephone. I 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 asterisks native protocol called EECS. And that's called the EECC. And I think I have one here. Somewhere. Actually it's in the, where did we put the blue box? We're not going to waste swag later so get your dialing fingers ready. This is the EECC, 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 bucks. You can also get the Sephora SPA2002. We've got those here running and these are about 70 bucks. You can plug two vinyl ratchet or vana ratchet. There's ESpeak SIF instead of EECS. You can plug two analog telephones into these. You get two dial tones and that's about 70 bucks. You can get the Grandstream handy tone 286 for about 65 bucks. That only does one I think that speaks SIF. And the old Cisco ATA186, which is the one that Vonage used to send out, is about 50 bucks online. You can also, there's tons more of these. There's one, the adapter that Packet 8 used to send out, the DTA310. You can find those about 30 bucks used online and although you can't really get them working as they are, you can re-flash the firmware and it becomes just generic SIP 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 TDM400B. 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 in the range of many ways you like. The bottom one is a T1 card and you can, there are two versions of that. There's a 405 and 410P and the 46411, the 411, 406 have echo cancellation built into them. And so you can bring four primary-rate ISDN interfaces into that and have 20 to 90-60 channels of voice going out of your asterisk box. Or if you've got a whole bunch of analog phones, you can get a channel bank and put the channel bank into the T1 for your stations so you can have 96 stations running off a one asterisk box with four channel banks. So, why aren't you talking, Ratchet? Let's get into some of the protocols that asterisk uses 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 finally it's cleared up in frustration. Okay, for those of you who didn't hear, that's me. It's apparently the Linksys PAP2 is the same way as the Sepura 2000. And they can or can't, except for, they cannot be flashed from. Anyway, getting back to interacting asterisk. Well, let me go adjust this. Our call for interconnecting voiceover IP calls is called SIP. It stands for Session Initiation Protocol. And it's a Session, as there's a sibling protocol only. It only signals and sets up calls. What it does is it contacts your box and say, hey, I have a court call for you. You might wanna 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. Real-Time Transport Protocol. It was developed by the ITF, not ITU. It uses URLs instead of telephone numbers. For example, if you won't connect to 311-555-2368, you'll enter it into, you'll say SIP, StromCarlson.com, which doesn't work, I've tried. It tended to be a peer-to-peer protocol. It's not like a general server. It's supposed to, like, my box connects to Stroms and it doesn't actually have to connect to it. It doesn't have to contact anybody for it to work. There's 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, Broadway's, essentially all the major providers use SIP. It's the exception of H&T called Vantage. It uses other protocols we'll cover later. Does not play well with NAT. It's very hard to actually set up with NAT. You have to start building 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 Stroms will also go, what? So, there's a lot of work around for it. What you have to do is you have to tell it to, 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 clutch. The other protocol, before there was SIP, there was H323. It was developed in 1996 by the ITU. It's really far more signaling to traditional protocols that are used by telephone networks than NAT's, 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 big world. It is a bear to set up. Like I said, major pain in the ass to get working correctly. Hold on. Although, by the way, although this is a telephone talk, please turn your ringers off. Thank you. Come on, I figure that's discriminatory in telephone users. We actually were hanging, Strom was hanging out in Astros one day on IRC.3Done.net for anyone interested. And Jair Jair, who's one of a major contributors in Astros, made the following quote, just don't use H323 and all your problems will be solved. Technically is correct. Now, for Astros, there's a main protocol. And as you can tell, it's based on Astros. It's called Inter-Asterisk Exchange. It was developed by Mark Spencer Digium, the guy who wrote Astros. 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 H323 and said, boy, these are crap. Let's go to my own protocol. It does not suffer from natural traversal issues. For example, I'm behind a firewall I call Strom and it knows that where I'm coming from, it knows where to connect. It doesn't actually say, hey, I have an IP address here, and 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 Astros. It's, but however the support and terminal equipment is very rare. The only PTA that supports it is the XE5 Digium that makes it. I've also noticed that one phone recently started to support it. I think it's called the Netwed 302. You can, it's actually made in China. And the only place you can get it is some place in Delaware. Where's the telephone? Could you shut off your ringer please? I'm testing for later. The nerve of some people. Let's see, what was I? It's a protocol for many PSTN termination providers because 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 up in one of my accounts about a year ago. I haven't touched the iterations of it. It just works. The other protocols are the Media Gateway Control Protocol, MVCP. That's the other protocol I was talking about. It's used by AT&T call vantage. I'm not sure if they still use it, but they did. I had bought an old AT&T firm on eBay and I had to reflash it and do all kinds of annoying things to actually end up 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 the CCP right now. You can reflat, 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 I would have made in about 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. I try to flash over the box crashes. There are two different SCCP implementations for Asterisk and they both kind of suck. Chan SCCP 2 doesn't suck as bad. But Chan's skinny. But that's kind of like being the tallest midget, so. 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. Wow. It's a compliment. So anyway, basically, there are many different Kodaks who you can use to encode voice for Asterisk and we're gonna go over what they are, what they do, and what they sound like. In order to understand Kodaks, you need to understand how digital audio works. How many of you don't really know how digital audio works? Okay, well, I didn't see too many heads 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. This is called, so you have an analog bunch of different slice up, a slice up bit 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 and 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 one and negative one. So depending on how much resolution you wanna give to the values, you can specify more. The way MULA works, what MULA 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 MULA. 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 a second is still way too high for bandwidth. They're greedy and they want to, or they have less bandwidth and they want to be able to send phone calls over the 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 four bits to describe the change from the last value what MULA does and PCM does is it actually describes each value individually. This one it says go down four, go up two and so on. Regardless of the original source resolution, regular DPCM just uses four bits. Adaptive differential PCM uses a varying number of bits that's been depending on the complexity of that sample but it's still 32 kilobits a second so they have to describe somewhere. It sounds almost like MULA 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 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 talk about 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 a second synchronous bandwidth for wireline telephones. North American uses MULA companding. Most of the rest of the world uses ALA 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 robbing which is basically to give you a supervision in band on one of the digital channels you get 56 kilobits a second but for the most part it sounds the same. On mobile phones you only get four to 13 kilobits a 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 MULA to the codec and in order to be able to hear it you have to transcode back from the codec to MULA. So there's an increased CPU load as 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 like the Sepuras speak MULA and then G.726 I think and then some of the other codecs. But yes. But you know I think we're gonna have to like keep track of whether this is on or off. But the. Thank you. But I think the EC only speaks MULA 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 MULA. Some codecs require licensing. Oh and one thing I forgot to mention if you're using non-standard bit 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. If there are people who 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 addiction 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, 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 U law or A law compounding. G.726 which is 32 kilobit per second ADPCM. G.729 which is eight kilobit per second CSA CELP. Just conjugate structure and algebraic codexided linear prediction. That requires a license from Digium to purchase. We're gonna demonstrate that codec later. And just in order for me to be able to call Raction's box and have one telephone conversation over the codec, we each had to pay $10. And if you will, that's only for one call. So if you wanna have more than one call using this codec, 10 bucks a call for the licenses. Not for instance. Like, yeah, simultaneous calls. So if you only buy five licenses, you cannot have more than five simultaneous calls running on that codec. No, if it were 10 bucks 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 predictive coding. This is what Skype uses. And 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 Ag Horbis stuff. That's also 32.3 kilobits per second codec-excited linear prediction. And finally, for those of you who are like 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 that is used for free. So apologies if it's not up to your musical standard, 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 it uses this version. Yes. Oh wait, so let me cue it up here. So on your DEF CON 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 LPC10. Yeah. So let's talk about getting Astris to talk to the real phone network because this is all fun and cool but it's gonna be kind of limited in use if you can't talk to your mobile phone or your mother or there are various providers that I've forgotten how many a hundred 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 just there. They're very easygoing. When I first set up my account with them, I'm not sure how they do it now, but when I first set up my account with them, they said, oh, you know, they'll pay you, you know, they worked in 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, Astris can call you and actually reach you on your VoIP line. The only numbers they have are in Michigan, which for me, I guess is okay because no one really calls me on the besides PIM. And they're also not too free-friendly despite being very easygoing. For example, during Strom and the Lucky 225's presentation last year, freaking in the age of voiceover IP was it? Yeah, I think it was something like that. They sent him a seasoned assess letter saying, no, they didn't send me the seasoned assess letter. Who did they send the seasoned assess letter? Was it, was it not there up here? Yeah. It's not there, you stand up. Who's here? It's Mr. Season DeSist. This is the man who's on here. Terminate your account and PNU for life from new phone. I haven't heard a peep out of them. Okay, we're just going to do this. Oh, they stopped carrying CPM this week? So maybe you're aware of that. This is what we did. Just to like after this weekend. So yeah, Astralink 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 in the talk. You can get your inbound termination via toll-free numbers and it delivers A&I, I, I don't think people do not know what A&I, I, I is. It's a class of service. So you can tell whether your calls are coming in. And Astralink does also proper call completion. So, this is why we don't like them anymore. But we'll get into our, our, our beep of Astralink later. There's also a voice call to the next. We have inbound numbers in a large number of great centers. And it's also called the proper call speed, 2.9 cents a minute. Then they drop it down to a low, low price of 3.47. Usually I have to go to a menu. This, apparently this is the first, this is the first time it's happened to me, but some people would have had major issues with it, so if you're really unlucky, you're totally screwed. There's Voidjet, which is really cheap, it's 1.3 cents a minute. The only pro, the problem is it's like that. I actually, I think it's 1.29 now. Oh, is it? Wow. A whole mill. Call or ID delivery is unreliable with Voidjet. You oftentimes, we have to numb them. That's because we all use LPC 10. There's also a company called Broadvoice. They have cheap DIDs in most race centers. After an initial account, they get you a couple bucks a month. Pretty much run by phone freaks, in fact, when you sign up and if you have them, send you the kit with the ATA and everything. They also send you this little pin, which is just logo, but it has a red circle and the slash through it. I actually found out this weekend, they also do T-shirts. Yes, at least with the Bell logo and the slash through it. So if you hear that person, please stand up and get some applause. Well, they lose. We were gonna give you just like, they were gonna get the pay phone too. The problem with, oh, and the other cool thing is that it has you a name label. It can deliver the name for it. So if you wanna do extermination or if you're trying to firewall or something, you've gotta do some, they're pro to service outages. I've got, you know, my dial, my service of one CNN delivery is also on a private show. It tends to work, but it doesn't work all the time. Why don't you explain me, the code, there's a maximum limit with all of the service features. Touchpad, marvels of modern technology. Where I'm at this telephone, basically I want a dial 315-555-355-355-355-355. It goes over the internet and the dial. Astros gateway interface. I'm gonna take him exactly. Basically it's not in your Dundee Airport. Start on the standard end point, and we'll standard in the standard out. And basically what Astros uses is all kinds of different ratings. Think back to that Texas speech program you played within like 1987, either 10th order or something goes, please register me. Astros also comes with a whole ton of sayings. She says all sorts of crazy stuff. Yeah, it tends to share some of the things you want. Yes, you are thinking what you are thinking and you're probably right. I'm 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 limit after. Well, maybe she'll read Richard Nixon's Social Security never out of it. I'm not a weather pattern, my name is strong. I'll show you my phone voice in a second. Thank you for calling DEF CON my phone. Kitty tested, blame her. Here's what's coming up in the next hour, press one. To search for an event, press two. To be feedbacked with DEF CON by 14, press zero. If any of you want to write down silly things on slips of paper and leave them up here, I'll do one for you. You guys left boroughs and alfalfa and this actually uses up. Okay, so we have some cool slag to give away. Giggium has seen fit to give us two gobs of stuff. Apparently strong and become very friendly with Mark Spencer. The G-Sector's very friendly. So Giggium sent us a whole box of really cool stuff. There's these little CDs with asterisk 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 you. I also have been registered as an oldschoolfreak.com stickers for two very kick-ass sites in my opinion. You can see some different forms almost as we're meeting up. 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 bill operating company. Pre-divestiture, not 9x. Pre-divestiture? Pre-divestiture. Operating company. GTE was not a bell company. How about this? I'm taking over now. Name a post-divestiture 9x regional bell operating company. For example, someone already said it. Raise your hands, people. You. Come up. You in the back. So, let me mind this thing out the way. Each of you is sitting in front of the telephone. So, let me prop up this list in front of you here. This is a touchstone 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 of it. 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 price. So, step this list up here. The rule's in just a second. Unfortunately, these don't do post-dialing. Otherwise, it would do a switch hook tap in competition. Can you all read that? Are you pardoned? Sir, can you read that? Okay. So, what you're gonna do is you're gonna put the phone. Each phone is dialed one plus the area code plus the number. After one second, if you dial directly, 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 re-dial 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 had to hang up to start the beginning again. So, ready, set, go. Big 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 wanna switch phones, Natas? You understand the rules, sir? Okay, on your mark, you'll touch the handset, don't pick up the handset, you can touch the handset. One plus the area code plus the number, yes. On your mark, get set, dial. He put his own call on hold. Yeah, let's use some bender of an old school free sticker, too. One of you still got a call on hold. Would you like an old school free sticker? Would you like an old school free sticker? What size t-shirt do you wear? Let's do this, let me plug the telephone in so we can hear the winner. I think we'll do one more, then we have to get on with the presentation. Let's make this fair, let's make this fair, let's clear this out, and have four all-new people. If you guys who competed come up to me after the show, you'll get swag, you'll get swag, you'll get swag. You're welcome. Okay, new rules don't flash so kept in the morning. Wow, he actually gave him the money. Good for you. I think that guy in the gray hat around. Okay, so, let's get some volunteers. How about this guy? I heard a patch. Does that come up? Yes. Okay, we have one more. I thought that was my line, Ratchet. 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... So, on your mark, get set, dial. Oh, we should play the touch tone music, but I wouldn't spread the touch tone thing. Well, your iPod is also next to the door. Yes. A princess rotary? Digitum's aptile cards do support rotary phones. In fact, you can navigate IVR menus with rotary phones with a zaptile card. However, the only thing that does support rotary dial are the... I think the e-sports thing does support rotary dialing. I'm not sure. It does? Okay, cool. So, yeah, this thing... This thing does support it. Unfortunately, no. This one actually belongs to Mark Spencer, and I had to give it back to him. How about you conflict my slightly more conflict? Yes, thank you. Enter your option. Initial address message. It was made by Automatic Electric Group, which was GTE'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 GTE-5s typically in some independent... It's the only reason I would ever want to live in GTE territory. Let's get into the technical details of this later. You might be running out of time. Color ID is always right, right? You can also... You can also have her do color ID spoofing most of the time with two lines in your extension. It allows you to... If you're doing scanning, you can prioritize the more of the spoofing. I'm straight. Here's an example of me fax spoofing. It's credit card personal. This time it didn't work. It didn't go through to the Christmas. Super color ID is something that I happen to get. It extrapolates tons of useless info from a telephone number. It does an address from whitepages.com, and it does switch it to some screen or some window you want to run. Here's a photo of me using super color IDs spoofing from Gladstone's for fish trash bar in the Pacific. Useless for anybody with phone screens? No, it does not. I think our provider for DEF CON by phone claimed that we can have a... probably open 600 calls going on some time. Yeah, American Idol or some... If you watch TV at a once... If you do that and they get a record contract, radio will get about 1.1% worse. What's your... And Matt, by phone number, let's just do this. Oh, here it is. On your exchange, right? Enter the IP address you wish to scan. No rocket. You know how to push my button. Please wait. I was found with 17 ports open. I go on port 7. Desire on port 9. Date I'm on port 13. Search on port 17. Coaches on port 19. SMTP on port 25. Names of the port 40D. Domain on port 50D. HTTP on port 80. Amic. He thought of all the phrases except telephone. From the phone. Shows that are coming up in the next hour. Press 1. Search for an event. Press 2. To log on. To search for a show by title, press 1. To search by presentation, please enter to the first five 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. Presented by Tom Carlson and Flatratch. In Barthenau, 3 and 4. Starting at... Saturday, July 30th, 2005. At 7 o'clock, T.M. Running for 110 minutes. To hear more about this event, press 1. You're all here, you know about this event. Maybe hear more about the event and read you back the entire description of the talk as printed on the schedule. Shouts to Strom. If you want to hire me for your voiceovers, please come see me at a reading my content. But don't give me 48 pages of show descriptions. Please. First entity. I don't think I can handle these. Turn your notifications on. I just want to make sure that you can see each of your issues that we call in. And also this, we came up to you and said, hey, our schedule really sucks though. You guys have changed or anything? Is there some place we can go to get updates? Go to updates.com.org. So I was like, okay, I'll, you know, sit down, fix everything. Not a single schedule change on that. I was, oh, it's caused me nightmares. I want to have nightmares for you. We spent all of Monday, all of Friday, trying to peel off the pack. What we should do next year is we're probably going to set up something where the schedule is a disclaimer. We all love the stuff that I update the schedule. And we set the day so everything hopefully is on schedule by the end. I cried when I finally found you. There's always no way to do that. I want to get to Q&A. The CDM cards question there is a telephone there. Queue up in front of the telephone. Plus, if you're watching on TV, call the following number, area code 213-415-1047 and 811-0 because that's a clearer picture of what it really sounds like. The table looks like trying to use music on hold. So what happens when you're driving in your car and you have tons of background noise, you're trying to use one of these? Thank you very much. Thank you. Let's see if anyone's calling in from the TV. Hello? He's in Long Beach. Hello? He's in Long Beach. Oh, okay, let's get you in. No, I've got a running on my DSL connection. Okay, so just the standard voice call, even if you only require 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. Generally about the same as the phone company between 3 and 5 telephones. Run, Forest! Run! There's a giant B problem at DefCon this year. Thank you. He's in Long Beach. I really hope there's someone's calling in. What's your question, sir? For interfacing your phones with Asperger? No, they're not the only ones supported. I hope... Yeah. No, they're not the only ones supported. I don't know what it is. I think it's something called voice tell that uses the same hardware, the same older version of it. It's supported, it works, but good luck in your work. It's ISA for use. Okay. I'm sorry about that. All right. Yes. This is actually on the DefCon by Bones, but we're going to move on to the next question. Let's get the next question from this real person. Yes, for the most part, Asperger is about as secure as a regular telephone, although the new version of the EATS 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 CVS head version of the EATS protocol. Let's take a call from... I'll just put those right here. The other fun thing about the pay phones out in the lobby is that they do not transmit a pay phone code, so therefore you can actually use DefCon by phone from them. All pay phones are banned because we don't feel like being charged the fee. However, thanks to GoldenTel, whoever supports them, they have a very unsupported, uninterrupted standard on their pay phones in this industry, but real pay phones. Hello, caller. Okay, I'll accept the money. You don't pay me this guy enough for you. All right, thank you. We'll take one more question from here, and then we got to pack up. Will we out back? You can launch it. The four people can be in the queue at any given time. And how do we figure out how do we figure out that you're the administrator? Because we're dialing our internal Dundee number. We've 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 you just answered the question instead of getting some passwords back. Yes, essentially, this allows you to set up your own phone section. Yes. Thank you much. We'll talk to your... Anyone I told you I had a really nice process. 2920. Okay, so let's go to 7920. If you go to the void-info.org-weekly, they have information on all of our services. I haven't played with the Wi-Fi function. Yeah, you can do what software on PDSU. Yes, I set them up for my own thing. I have a nice tutored headset, a Wi-Fi hotspot that comes right in hand. We're gonna go wrap this up. Thank you all for coming. I'm going to send out shouts to binrep.com. You can find Strom and I there on forums up binrep.com. Anything else? We have stickers that we can give people. Just follow us out. We'll talk to your questions and do whatever. 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 the slideshow on your CD. Thank you for coming, everybody.