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I don't buy anything but if I'm bored I try to string them along for as much as possible wasting their money. With any luck I'll eventually be blacklisted and not receive any more calls.

Say "Wow, yes, I'm definately interested. Hold on while I go get my credit card information". Then place the call on hold, and go do something else.

If you run Asterisk, there's a great script to have some fun with them.

I just setup an Asterisk PBX box (Fedora) and got myself a Broadvoice account with a local phone number. Where as Skype uses a proprietary protocol, even most SIP based providers don't allow you to use anything but their 'locked' phone/adapters. Broadvoice happily suggests you 'bring your own device', and plays nice with Asterisk.

I just put together a little python script running against Asterisk through AGI (Asterisk's CGI) which pulls weather data from NOAA's site based on a zip code you enter, and speaks it to you. I can call it from any telephone through broadvoice.

~Blake

This is exactly what my company does for VoIP, including Asterisk and SER. Our customers are mostly ISPs and companies replacing PBXs. It can be a tough sell at times, but getting easier as these products mature and more and more ISPs want to offer VoIP to their customers.

However, we still have a quite a customers who want something commercial, such as Cisco Call Manager.

1. But what about your open source, small office/home office/home VOIP setup? It's not gonna work! Until we have some real standards and maturity in the VOIP industry we aren't going to have voice over internet protocal (VOIP) we can really trust to work when we need it.

Look here. The Asterisk PBX is open source. It uses ethernet and SIP protocols. You can get native SIP phones that work with it or use two different styles of converters that allow you to connect any analog phone you want to it. While it is not cheap, it is cheaper to deploy and has many feartures not found in even much more expensive PBX systems.

Asterisk has the ability to handle any call and route it transparently to SIP phones, analog, or software phones.

One of the analog phone converters can be used with a variety of VOIP providers by hooking it up to your sound card. Even has an audio pass-through.

VoIP is based on UDP, and does not easily vork over TCP.

VOIP is not a protocol. SIP and the like are protocols. Not all protocols used for VOIP are UDP. Inter-Asterisk eXchange or IAX is TCP.

Can you do this outside the US or do you need a US billing address?

I live in Asia and I have a US number from Libretel. It costs me $6/month for unlimited incoming calls, and they can be routed to any SIP URL (a FWD number, your own Asterisk box, whatever). Saves my callers a bundle. All I needed was a valid credit card.

A short Gaul whose best friend is Obelix... oh, wait, I thought you meant Asterix, not Asterisk...

Have you looked at asterisk?

We've run Asterisk http://www.asterisk.org/ for about a year now and when we relocated our offices decided that Verizon wanted too much for long distance. I signed us up with Broadvoice http://www.voip-info.org/tiki-index.php?page=Broad voice and configured our phone server to send all long distance calls out over our 1.5 MB DSL line. It's been PERFECT! I did have to upgrade our Asterisk to the latest tarballs, because my older CVS version couldn't register with Broadvoice though. Dirt cheap long-distance.

If people used Asterisk Combined with e164.org free lookups the whole process becomes transparent, and people don't have to go out of their way to make "special" cheap/free calls, it can all be done automatically at the PABX/PBX level and all the person thinks is they've made a call, they don't care how it got to the person.

other voip providers use a something called Session Initation Protocol which is an open standard, compatable with loads of clients even open source ones like kphone.
Even the routing can be done with open projects such as Asterisk. Skype is worthless proprietary tripe compared to these solutions.

I don't get it. I visited Asterisk and I don't see any screenshots of the program running.

I thought all open source programs posted screenshots on their site, thats how people tell how good the software is!

(Yes... I am joking....)

I work for a company setting up a large voip network, and I must say the technology/software is really coming along.

Some of the features that asterisk can do is amazing.

And it's all open source to boot! All you need to do is pay for hardware (servers, and voip cards) and a good coffee machine to keep you awake during all the phone calls you eves drop on using ZapBarge ;)

• Internet: Mediacom 3Mbps down, 256K up cable modem. Quite reliable, down probably for 10 minutes a month, maybe less. About $45 for that.
• VoIP Provider: BinFone Service through Binhost Technologies, a company I'm a part of. We're small but we know our shit, we're cheap, and we have geeks running the entire show. We are more into reselling VoIP but also do individual IAX and SIP accounts. Rates are $0.03/min for USA, $0.05/min for Australia (wife is Australian, we call there a lot). More info here.
• Phone: Grandstream Handytone 486 SIP phone adapter. A very cheap ($65, I believe) phone adapter, but has a web interface, good features, and does what I need it to. It is plugged into the network via CAT5 and into the phone patch block via standard POTS wire.
• IAX Server: I run my own IAX server (Asterisk) in-house. It talks to Binhost's server through the IAX protocol (Asterisk proprietary) which is very efficient. I have an X100P FXO PCI card in it that allows connection to the PSTN (my landline) and a NIC to talk to the network.
• Firewall: All of this sits behind the firewall, a simple Pentium 233 running Slackware 9.1 and using iptables and QoS scripts to regulate traffic. The QoS designates packets by the MAC address of the Grandstream as highest priority so my VoIP packets always get through quickly.

Monthly VoIP service: About $20 for the calls, $5 for the line.

Internet: $45/month

Asterisk: Free

Asterisk server: Free donation

FXO Card: $15 on eBay

VoIP Adapter: $65

Wiring: out of some old box

Firewall: Free donation as well

Landline costs: $17.95/month
So total? $80 in startup, $87.95 monthly for all my phone calls and Internet service. I call that a *deal*.

I try to avoid relying on a product which has a single supplier or is not standards-compliant, even if it does meet the FSF's standards.

Standards compliance is a great thing. Recently I've been working on a VoIP deployment using SIP phones, Asterisk, and SER. One of the things that has impressed us the most about SIP telephony -- as contrasted with earlier VoIP and digital office phone systems -- is that the major vendors' products all interoperate at a basic level (placing and receiving calls) out of the box. This is a big contrast with earlier systems where (e.g.) Nortel sold you a Succession VoIP system, and nobody else's phones would work on it.

Most of these SIP phones are not open-source. Cisco's and Grandstream's phones are the usual binary-only deal -- compliant with open standards, but not even source-available. However, some SIP phones are open-source, notably SNOM phones, which run embedded Linux, and for which you can download an SDK from their Web site and build your own firmware image. Not too terribly surprisingly, SNOM's phones are not the slickest in appearance (that would be Cisco) or the cheapest (Grandstream) but they are, as far as we can tell, the most configurable.

Much the same seems to be true of VoIP gateway systems. Many people with whom I've spoken are using Cisco instruments as their gateway between SIP and the PSTN (conventional phone system). We are using Asterisk. Although it is hardly the easiest software to configure -- it's kind of like the Sendmail of VoIP, minus the security hell -- the Asterisk/Zaptel/Linux system is far more flexible than closed equivalents.

So what does this have to do with the advantages of open source? In a field of open standards, such as SIP telephony, open source can really shine. Open standards mean that there is little space for vendor lock-in, so vendors cannot exclude open source in the usual fashion. Open source is largely immune to the problem of treating standards as "tick-list features", which some appliance developers seem to suffer from: implementing the standard in a slapdash way so that you can mention it in the four-color glossies. ("Do we have, um, this 'SIP' thing?" "Uh ... [type type] ... sure, we do now!")

So how does this contrast with some of open source's notable weaker points, like user interface and graphics software mentioned in the article? It seems to me that open standards and open source both have their strengths in infrastructure as opposed to interface: not the buttons that users push on their desktops, but the underlying systemry that really makes the system (and the network) run. The advantage of Asterisk over proprietary PSTN gateways is much more than the advantage (if any!) of SNOM over Cisco SIP phones. The same is true in other infrastructural roles: the advantage of Apache over Microsoft IIS is much more than the advantage (if any!) of KDE over, say, the Mac OS X interface.

For the user of closed-source end-user systems (be they phones or desktop computers) the presence of open source in the infrastructure means that it can be customized by experts (IT staff or consultants) to the needs of the organization. It also often means that the infrastructure is simply higher-quality, which benefits everyone. The folks who get VoIP phones on their desks at my workplace don't care whether the gateway is Asterisk or Cisco, but they do care if we can implement features they request. Likewise, our Web designers using Dreamweaver benefit more from the fact that we use Apache (since their work is safer than it would be with IIS) than they would by using an open-source end-user tool.

To make a tangent, consider Microsoft. Their tradit

I understand you want to get some formalized training while the getting is good. Don't blame you there.

But, honestly, the absolute best way to learn something like this is to do it. Download asterisk and start playing. You can even connect a free soft-phone (SIP) to it so there is really no up front cost.

IMHO, formalized training is only useful *after* you have already learned a good deal about the subject matter. It gives you a chance to organize all you thoughts that have been plaguing you, and have a compentent teacher answer them for you.

Going in cold to VOIP lab, when you leave you will understand some jargon - but most of the deep concepts will escape you.

Yes, A bunch of fixed wireless nodes wired into their VOIP switches is exactly what they are doing. I have been expecting someone to do this...would have done it myself if I had the facilities & resources to get it started.

It's still a pretty cheesy solution though. What we really need is for the Cellular providers to setup VOIP gateways directly to their private networks (preferably with IAX2 protocol as an option to work with asterisk http://www.asterisk.org/) and then I can broker calls to or from my cell phone, the traditional phone network, or any VIOP network as I please using my asterisk switch. The cell phone provider could charge a small monthly fee to those who want to use the gateway to cover their (relatively minor) costs of providing it and probably make a bit too and everybody could be happy. Are you listening VERIZON...AT&T...NEXTEL? I would think it would be a boon for NEXTEL as their many business & government customers could further integrate their wired and wireless communications making those accounts very happy and almost turnover-proof. Plus NEXTEL could offer services for setting up their clients with this technology integration for a nice hourly rate.

Maybe, depending on exactly what you can live with. There is a bluetooth module for asterisk, that can connect to some cell phones. It is not stable yet, but it in theory you can set a computer with bluetooth up near the window, running asterisk. When you get home you put your cell phone by the window (and on a charger), and Asterisk connects to the phone. Then you call from whereever you can connect to that machine. Either with VOIP from other computers in the house, or a FXS/FXO (I can never remember which is which without a lot of reading) board to your current phones. Not to mention all the VOIP networks asterisk can connect to.

Like I said, it isn't exactly what you asked for, and not all the modules are stable. However it is worth checking out anyway.

you really could automate it like that. "If you have noticed your browser won't browse, press one" "Lost your password..press two" and etc

I don't know what software does that, but it exists, those sorts of automated responses are quite common when you call into places.

Actually, Asterisk would suit the purpose just fine, once you got over the install hurdle.

One can only imagine the fallout within the telecommunications industry if an open-source project like this gained traction

SIP based VoIP, Asterisk compatible if you want to get fancy, uses STUN to traverse nat'ing firewalls. They even sponsor a few SIP clients so it's all free, and you can buy a cheap hardware SIP phone or interface and make the calls from a real phone instead of a PC.

Take a look at Asterisk http://www.asterisk.org/. The wiki http://www.voip-info.org/wiki-Asterisk has more useful information. It is a full VoIP softswitch solution. In addition to SIP, H323 and MGCP it also supports the IAX protocol, which was designed to be NAT friendly. You won't be able to run it point-to-point. You will have to run an Asterisk server somewhere in your network, but since you are already running Linux on the desktop, it should be fairly easy to run it on one of them for a small network. Combine that with a dynamic dns service like dyndns.org and voila!

Vonage and the likes already have momentum. Asterisk and the likes are in position to take over the PBX market. Connect the two automaticly, along with various other networks, and there is enough mass to solve this. Aunt Mary might not understand it now, but when all her relatives tell her to get off SBC because she is the only one in the world(!) they call where they have to pay fees, and she will be forced to listen. Once Aunt Mary realizes that she can call pretty much everyone for less on her VOIP phone, she drops SBC as an extra bill that she doesn't need.

Soon SBC and the like will file for bankruptcy... Not really, they do have DSL, which is a good way to connect. When the notice that customers are switching to Cable internet just to avoid having to pay for an unused voice line, they will drop that all voice/DSL bundling requirements.

As geeks it is our responsibility to socity to make sure it happens. So start your own VOIP expiriment at home, and use it once in a while. Long distance telephone is obsolete, but nobody has realized it yet.

At my company we use open source projects such as BIND (for an ENUM / DNS based call routing directory) around the edges of our VoiceXML / VOIP IVR hosting service, but not in our core platform.

Originally we did use early open source VOIP projects such as OpenH323. OpenH323 was great, but it needed to be replaced as we moved to SIP and required reliability beyond what OpenH323 offered.

Asterisk is in a similar place - it is a great project that has seen some great early success in voip. I have heard that Vonage, for example, uses it in their voicemail system. I also use it at home and we have several projects at work in the research phase that incorporate it.

Asterisk is not reliable enough for our production environment today - reboots every few weeks to few months are common. As a project it is similar to where Linux was 5+ years ago - plenty of momentum but not quite ready for mission critical use. I have no doubt Asterisk will become as pervasive and reliable as Linux and other leading open source projects have though. Asterisk is an extremely flexible, easy to work with project; and the people involved are also easy to work with and know telephony very well.

Right.

The world would be far better off if everyone installed Linux on a spare computer so they could run Asterisk. You then just need to buy a bunch of hardware, and then either spend a few hundred dollars each on WiFi phones, or spend tens of hours recabling your house.

Oh, and then you get to configure the mess, after learning all about such eccentricities as G.711, G.723.1, GSM, IAX, and SIP, SCCP, plus a whole lot of other defacto telephony standards and Ways Of Doing Things that were obviously developed in a cave.

Once you solve the echo problem, all you gotta do is make DUNDi work, and you can finally call other people Just Like You. Or, you can sign up with any of dozens of shady small VOIP telephone companies and pay a few tenths of a cent per minute to talk to regular people via a SIP, IAX, h.323, or MGCP connection.

Sweet.

Alternatively, one could always download and install Skype. I understand that it does work fairly well, and is easy for mere mortals to use. It seems that Siemens now has an easy way for you to use their handsets with it. Neat.

The FCC had ruled previously that "pure VoIP" was free from regulation because calls originated and terminated over the Internet, but regulation of VoIP calls that terminate on publicly switched networks had yet to be addressed. These calls and services will now be treated in the same way.

So does this just apply to the big corps or to everybody?

If I hook up my old-school phones in the house with VOIP TA's and terminate them at an Asterisk PBX, onto my POTS line (which I'm about to do anyway), do I get to forgo the state taxes on those lines?

This company is probably nothing more than someone running Asterisk, using Nufone for the PSTN service.

A simple php script will dump a callfile into /var/spool/asterisk/outgoing and bridge the two calls together.

Then all you need to do is write something to manage user accounts, and accept paypal payments and bam. You've got camophone.com.

This whole configuration could probably be whipped up in a day.

Very Cool =)

If they're ever stuck for which project they should contribute to, send em on over to have a looksie at asterisk, The Wishlists and bounties on the asterisk wiki, and the asterisk bug tracker! [/shameless plug]

Seriously though....

However, Voice over IP and even open controlled analog/digital converted PBX systems (like Asterisk), will be able to converge into a single, re-assignable open standard.

If you are comfortable with interfacing your servlet engines with your phone system, Voice over IP (and H.323 standards) will allow you to do so.

Offtopic, My Ass.

There's no special wiring involved anymore, the terminals (phones) are computers in their own right, connected to the enterprise IT network, speaking IP.

It's not an island, it's part of the modern IT infrastructure.

See Asterix, which works with three VoIP protocols.

Personally, I'm intrigued by software like Asterix and its capabilities, but I have absolutely no telephony knowledge and I'm not really sure where to start, like what kind of hardware I'd need in order to set this up with POTS. Lots of modems? Special cards for the phones in the office?

With Voicepulse Connect http://www.voicepulse.com/ I pay $7.99/mo for an unlimited minute incoming phone number and 2.95 cents/min for outgoing calls (local or long-distance). Even with a wife and two daughters my call accounting tells me I would spend less just paying by the drink than my SBC local analog line at $34 (taxes, fees and caller-id included) by almost $20/mo!
In short, you'd be surprised how few minutes you really do use.
On top of that you might want to consider Voicepulse connect because I now get multiple incoming calls and multiple outgoing calls at no extra cost other than that the meter runs for outgoing calls.

The catch? You gotta run Asterisk http://www.asterisk.org/ and get at least one FXS port card from Digium http://www.digium.com/.

Anything over $15/mo is robbery in my opinion

How about setting up your own Asterisk server (yes, it's Linux, but it works, get over it). Then you can use VoicePulse Connect! to get a cheap rate for an incoming line.

Well thats all well but broadvoice charges $20/mo. I've been using them since August and have been very happy.

Broadvoice+Sipura from voxilla+asterisk=awsome home phone system.

• Local access lists. Every serious SMTP MTA supports access control based on IP address, reverse DNS, attested address (HELO), and so forth.
• DNSBLs and other sorts of published blocklists. A DNSBL is nothing but a site's IP-address access list, published over the DNS so that others can use it.
• Protocol enforcement techniques such as greylisting. Greylisting tests that the sending host is willing to make the effort of retransmitting, as required by the protocol.
• Content filtering. Even a server-side antivirus program is a content filter; much more so the statistical filters often used today.
• Multi-site statistical tools. Vernon Schryver's DCC and Vipul's Razor come to mind.
• Traffic limiting. ISPs can restrict the number of SMTP messages a host can send per day or hour.

Many of these techniques can be adapted to VoIP systems. I am surprised that SER and Asterisk do not already support DNSBLs -- even if there is no call for them yet, we will certainly need published lists of abusive hosts or networks within a few years.

The flexibility with which one can express access restrictions is an important part of any system's security. My workplace is just starting a VoIP deployment. I want to be able to say things like:

• No single outside host may make calls to more than 50 different destinations in a day.
• No host may send more than ten pending SIP invites at any time. (Prevent predictive dialing!)
• No host may send SIP IMs to more than 20 addresses in the same minute.
• After an inbound call is completed, the recipient can dial *666 on our Asterisk PBX to report it as an abusive call. If five different addresses report abusive calls from the same originator, that originator is flagged and blocked for 24 hours.

You must have missed the eailer post about Asterisk the opensource pbx...

>Server software that runs on Linux for those of us that like a standard back office.
like asterisk
It supports many VoIP standards, pots, BRI, PRI, etc...

Within five years, the telco world will have changed.

We will observe a strong fragmentation of the telecommunications world as many small companies will try to get their share of this multi-bilion dollars market. And just because of the low entry cost (look at asterisk, Convedia, Ubiquity, Appium, and many other players way too numerous to list here), you don't have to be a huge company to deliver services in that emerging market of VoIP services (here, by VoIP services, I don't only mean providers, but also secondary services like voice recognition, IVRs, vertical markets services, unified messaging, value-added access resellers, etc.). Maybe after, the market will reconsolidate though.

VoIP is to telco what PC was to computing, what the Amiga Video Toaster was to TV productions, what Napster was to RIAA, what iPod was to MP3 music, what Internet was to information access, what Word, Excel and Powerpoint was to corporations, ... It's a disruptive technology.

It's a fact; those who can't adapt to their changing environment will disappear. And new dominant players will take their place in a new order...

I wonder what my phone (ok, communication device) will look like and will allow me to do in 5 to 10 years from now.

I don't know about other VoIP providers, but Packet8.net has been great for me. I've had friends use the phone at my house and explain to them that they need to dial 1+area code+number and then when they get off the phone I tell them the call went over the internet.

Usually, they are surprised that it wasn't a "real" phone conversation. I have sold a lot of people on it because it's only 20 bucks a month. I'm switching to BroadVoice when they have area codes in my state, because they give you the SIP username/password so you can use Asterisk Linux PBX.

Chris

If a product is closed source and proprietary, then that should be all you need to know about it.


The company for which I work already uses VoIP, but we wouldn't touch Skype with a barge pole. It's our policy that we avoid closed-source software as far as possible, even if that means having to do stuff by hand. We use asterisk for an exchange, together with Zultys hardware IP phones, using SIP. We just have an ISDN-30 line (E1) connected with the appropriate hardware interface card (by Digium) to the asterisk server. The card is multi-span, just in case 30 lines turns out not to be enough. The server is a dual Xeon 2.8, which might be slightly overkill for Asterisk; but it's also running our office software (we pretty much were using LAMP applications before the name was coined) and the E1 card needed a 3V3 PCI slot which is only found on expensive mobos. (There is now a 5V version available ..... d'oh!)

We paid money for the hardware, and we paid in blood, sweat and tears for the software; but nobody can ever take away what we learned.

The backlash against Star38 is the type of friction that can arise between for-profit software companies and hackers who resent the commercialization of technology they believe should remain free.

"In most countercultures, there is an aspect of selling out," said Caleb Sima, co-founder of Spi Dynamics, an online security company. "People who make money off technology are deemed to have sold out. Anyone who has a unique idea and is making money is going to get badgered."

Why are they blaming this on FOSS?

Telemarketers already have the means to do this. Any type of trunked or primary rate phone line and PBX equipment will let you do this.

You can even do this with free software and some relatively cheap PCI card(s).

...watch legislation arrive to clamp down on who can own PBX equipment, and what it can be used for.

Bye bye, Asterisk.

We're not a 501c3 organization this year, and so we don't have the needed infrastructure in order to charge for admission. We do have some CafePress gear available with a $1.00 markup on each item to help us cover some of the costs we're footing from our personal funds. No donations required, though. Besides, it's better when it's free (sex AND conferences)! ;)

We have some excellent speakers coming in this year from Red Hat, Novell (SUSE), the Apache Software Foundation, the Samba Team, and more. We're having an excellent speaker on the Asterisk PBX for Linux and VoIP technologies.

Hosted at The Ohio State University in Columbus, Ohio and with plenty to see and do in the city during the visit, as well! I'm looking forward to seeing a lot of friends this year, and the afterparty should be grand!

It appears that the folks building MySQL are even MORE pro GPL than a rabid /.er! I know it's hard to believe, but unlike the very bad description of the artcle given above (the sky is falling, the sky is falling) the actual text of the article shows that the company is pro GPL. It isn't backing away from the liscence, but tryng to be sure that users of GPL software uphold that very lisence.

Whats interesting is that this affects open source, but not necessarily GPL projects. Asterisk which use a different lisence have removed MySQL libraries because of this conflict.
From their documentation:
"We were recently contacted by MySQL and informed that the MySQL client
libraries are now under GPL license and not LGPL license as before.

Since Asterisk does allow exceptions to GPL, we are removing MySQL support
from standard Asterisk. We will, where appropriate, make it available via
a separate package which will only be usable when Asterisk is used completely
within GPL (i.e. not in conjunction with G.729, OpenH.323, etc). We
apologize for the confusion.

Is this a case of the GPL being a bad thing?

cluge

• Asterisk, the Open Source PBX
  
• OpenBSD, the secure operating system
  
• The OpenBSD PF FAQ
  
• IPsec HOWTO
  

Armed with this information, use OpenBSD to set up firewalls with ALTQ packet prioritizing, PF stateful filtering and IPsec secure VPNs between all endpoints.

Setup Asterisk PBX' behind the firewalls and network them over the VPNs.

Now let them try to monitor your calls.

(No, this doesn't help with calls you terminate with an insecure 3rd party, like a VoIP provider gatewaying your calls out to the PSTN. The "P" in PSTN is for "Public", so you need to treat it as completely insecure and act accordingly.)

Actually, I'm currently researching 2nd-hand (maybe even new) small PBXs & associated phones.

Check out Asterisk - very configurable and you can either use IP phones or soft-phones, either way you're using either your existing cat-5 wiring or 802.11.

VOIP is becoming a big business. If you don't like one provider, try a different one. NuFone is a good one. It works extremely well with Asterisk too.

It also needs to be noted that asterisk is much much more then a voip gateway. As for as Digium's business, it is geared more towards the people that want to connect PRI's and/or analog phones to the system. This is how they make their money.

http://www.asterisk.org/
http://www.digium.com/
http://asterisk.xvoip.com/

So what does this mean for Asterisk?