Slashdot Mirror

Do you think I'm suddenly going to freak out on VOIP because the US government might start listening in on my calls? I'm actually suprised that they're not already (they seem twitchy about that stuff right now), though this may be a political version of "it's easier to ask forgiveness than permission". Fundamentally, I don't care how my voice gets from point A to point B, but I'm in favor of doing it as cheap as possible. I like the idea of a world where they run one cable (or no cables, woohoo) to my house and all the information flows over it. The tinfoil hat wearers can roll their own VOIP for talking to whomever they want to talk to and encrypt it out the wazoo. If they're paranoid enough, they can get multiple wired and wireless connections, split up the packets across them all, and have a grand time of it. As best I can tell, VOIP was never about avoiding the government, it was about talking on the cheap using resources already available.

Now, if they come for my encryption, they'll have to pry it from my cold, dead connection

- Tash
Vrrooommm...

Well the obvious answer (at least to me) is Asterisk. If you don't want a "computer running all the time" build a small box, well.... tuff. Think Mini-ITX. You can put a small HD in in, and put it in a small case. If its only "diverting" calls it doesn't need much power or storage space at all and wouldn't draw much power (also, if you do it right - it could be all passive cooled).

Also - I'm sure no one wants to spend _that much time_ setting up Asterisk, so use TrixBox (Formely Asterisk@Home) instead.

Asterisk

The bad part? The Moviebeam player also requires a connection to a phone jack -- every fortnight the box dials a toll-free number in the middle of the night to tally how much you've spent on movies so far, for the benefit of your monthly statement.

http://connect.voicepulse.com//
http://asterisk.org//
http://store.voxilla.com/customer/home.php/

I'd LOVE to build an asterisk PBX to run a hotel. That would rock so hard. I'd done the design for an asterisk installation in that type of environment, but have never had the opportunity to deploy it. I'd highly recommend it if you want absolute total control of your PBX.

Your girlfriend probably want to subscribe to something like Vonage, SkypeOut, or some other Internet telephone provider. Sure, she'll have to pay for her calls, but it's definitely a lot less hassle than having to set up a gateway.

Plus, she gets her own phone number people can call her on, if she gets a service that does that.

On the more geeky side, if you want to be your girlfriend's telephone operator so you can give her free calls (I don't know, maybe that kind of thing turns you or her on), you definitely want to play around with Asterisk, the free open source PBX, and get an account with a tinkerer-friendly SIP provider. Using that setup, and a SIP softphone program on the computer, or a hardware SIP telephone adapter or SIP telephone, you can do pretty much anything you can imagine.

Don't bother trying to do anything clever with Skype though, it's not an open system, and you're a lot better off with an account from some kind of tinkerer-friendly SIP provider. Not living in the US, I can't give you any specific recommendations.

Hope this helps.

I saw your post on Ekiga 2.0 when I googled for other stuff, but it was to late to replay and I can't find a function for sending private messages so I'll answer here instead. Sorry for beeing off topic.

Regarding SIP and Ekiga I'm not sure if there are any chances to use SIP behind a NAT without any forwarded ports (in that case using STUN but I doubt it.)
What ports are used are application dependant, but this is what I use and I have no troubles with any application or provider so far:

5060 TCP/UDP (SIP UAS, I think this should be enough.)
5000-5100 UDP (But with only the ports above I could only use my other account, not the Ekiga.net one, with these both works.)
3478-3479 UDP (STUN service)
49152-65535 UDP (RTP, RTCP multimedia streaming)

I _think_ that port 5060 (or 5000-5100) is used to say "hi, you got a phone call", but the actual voice data are sent over port 49152-65535 (valid for SJPhone, might be application dependant). I think you only need the STUN ports open if you use STUN, STUN seems to be a service which helps the application to figure out what kind of firewall it is behind.

I've covered this with screenshots in my blog but it's written in swedish so I don't know how much help that gives. Atleast you can look at the images?
I know for my "real provider" I had to tell them if I where behind NAT or not, doesn't seem like anything like that is needed for the ekiga.net account for whatever reason. Maybe with STUN + NAT enabled user you don't have to forward any ports at all?

Another alternative is to use Asterisk or Asterisk@home somewhere outside the firewall and have that handle the SIP account and use IAX for your clients instead. There is a nice (atleast screenshots says so) client available for UNIX called KIAX.

Ekiga is a nice client, for whatever reason it crashes all the time now when I upgraded to Ubuntu Dapper, but I've got noone else to blame than myself for that one. SJPhone isn't open-source but it doesn't cost anything and is very competent aswell. For Linux linphone is less good and kphone is even less good ;), I would rate X-ten X-lite(is that the name?) as just beneath SJPhone.

...other than Asterisk, right? Or is this somehow much better?

Depends on what you consider an application. I'm pretty sure http://asterisk.org/ has a few months on you.

Yes, but does it run on Windows? :-/

Yes, it does. Kinda.

Depends on what you consider an application. I'm pretty sure http://asterisk.org/ has a few months on you.

Yes, but does it run on Windows? :-/

Ekiga is the first Open Source application to support both H.323 and SIP.

Depends on what you consider an application. I'm pretty sure http://asterisk.org has a few months on you.

I'm just curious... why not just replace it with an Asterisk PBX ?

You're right, I haven't seen any reference to Asterisk testing the processor type to make sure it can handle hundreds of calls at once.

If this was genuine Skype should test for Spyware / Virtual Machines and how many other processes are running before initiating a conference call. It's abuse of monopoly pure and simple.

Asterisk.

It does conference calls really well and is not just free as in beer.

Corporate stupidity isn't always a bad thing. It's just a matter of letting them shoot themselves in the foot and then reaping the benefits of their pain.

MTW

Stick to open source telephony. Asterisk makes an excellent enterprise grade open-source PBX for the back end. For the end user, Free World Dialup offers a SIP compatible service with a free downloadable client that does not limit you like this.

Firstly that's Asterisk not 'Asterick'.

Secondly if you have access to a telecomms switch that has ISDN BRI interfaces, then sure you can test to your hearts content.

Alternatively if you have links to any organisations that may have a few spare BRI interfaces you could always try to talk them into letting you borrow a few lines while you do your thing but with ISDN it's not like you can just use a 'crossover' cable or anything.

This is a wee bitty redundant, but the figures might be interesting:

One of my clients recently looked into a PABX/VoIP solution for their two very small offices. They required only 10 IP phones and two gatekeepers.

Samsung's quotation was ~AU$14,000; Nortel's was ~AU$18,000. [AU$1 ~= US$0.70]

These were proprietary systems with weak licensing (Nortel: 32 license minimum for voicemail, etc.), limitations (Samsung: only four calls simultaneously!)

Another mob wanted $8000 for just the IP phones necessary, with ongoing (extortionate) costs for using their ISP, their VoIP provider, and their gatekeeper.

My quoted Asterisk solution will be less than AU$6000 for 2 servers, ISDN/PSTN cards, quality IP phones, no licensing, et cetera. Plus the features on offer are more numerous and 100 times more customisable.

Why would you bother with anything else?

My AU$0.02

Asterisk -- 'nuff said.

I'm amazed asterisk@home wasn't the first thing posted here. Don't be fooled by the @Home part. This is a full fledged install of asterisk that is only limited by the hardware you install it on. You can have a working PBX in an hour. I'm planning to install this at all my remote sites (6 of them) with free extension call throughout and then plan to install it at my main location (150 phones) and have it all interconnected. A VERY powerful solution.

(Note: I just copied the rest of this from the handbook so I don't have to retype it all)

The Asterisk@Home project enables the home (or small office) user to quickly set up a full featured Asterisk PBX with a web based interface in about an hour on a dedicated PC. Even if you are new to Linux, Asterisk@home handles that by handling the complete Linux install for you. In order to get up and running all you need to do is download the Asterisk@Home .iso and burn it to a CD. Boot that CD and you will get a very complete Asterisk and Linux install.

Asterisk@Home provides a nicely integrated install of some of the best software from the Asterisk community, such as the Asterisk Management Portal, which provides an intuitive Web GUI for configuring asterisk, and the Flash Operators Panel, which lets you see and control your Asterisk PBX in realtime, and FAX support through span-dsp.

What is included in Asterisk@Home 2.0:

Linux CentOS 4.2 - http://www.centos.org/ - CentOS is 100% compatible rebuild of the Red Hat Enterprise Linux (RHEL), in full compliance with Red Hat's redistribution requirements. CentOS 2, 3, and 4 are built from publically available open source SRPMS provided by Red Hat. CentOS conforms fully with the upstream vendor's redistribution policies and aims to be 100% binary compatible. CentOS mainly changes packages to remove upstream vendor branding and artwork. CentOS is for people who need an enterprise level operating system with stability to match without the associated cost and support.
Apache Web Server (2.0.52)
MySQL Database (4.1.12) - SQL database for Call Detail Reports and optional configuration information.
Php (4.3.9)
Asterisk 1.2 - http://www.asterisk.org/ An open source software implementation of a telephone private branch exchange (PBX). A PBX connects one or more telephones on one side to one or more telephone lines on the other side. A good example of this is a small company with 100 internal telephones sharing 20 outgoing/incoming telephone lines. A PBX can be more cost effective then having 100 direct telephone lines.
AMP 1.10.010 BETA - http://www.coalescentsystems.ca/ - Asterisk Management Panel is a web based GUI that allows you to easily manage Asterisk without having to edit sometimes complicated text configuration files. This package is can really make a difference in learning and configuring asterisk easily.
Flash Operator Panel 0.24 - http://www.asternic.org/ - Flash Operator Panel is a switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your PBX activity in real time. You can see what all of your extensions, trunks, and conferences are doing. You can also hang up, transfer, initate a call or create a conference call.
Festival Speech Engine version 1.96 - http://festvox.org/festival/ - Festival is a speech synthesis system. It allows you to enter text that the Asterisk@Home server "reads out loud" to anyone calling the server. Using this, you can be sure the same voice is used across the whole asterisk server.
SugarCRM with Cisco XML Services interface + Click to Dial - http://www.sugarcrm.com/crm/ - SugarCRM is designed to a be a complete customer/contact manager. Using SugarCRM we can manage all types of communications (faxes, te

I've switched to using http://asterisk.org/ along with http://www.broadvoice.com/rates_compare.html. I think you'll find this Wiki to be a very useful resource: http://voip-info.org/

The plan I'm using is BYOD-Lite which costs me only $6 a month and there was no activation fee, since I had my own VOIP equipment in the form of an Asterisk PBX installed on Linux. From what I can tell, they are one of the few providers who allow the use of customer supplied VoIP hardware/software, in my case Asterisk.

Something you'll have to research is what technology you want to use for hooking up individual phones to Asterisk. One possibility would be to use hardware from Digium: http://www.digium.com/index.php?menu=product_categ ory&category=hardware or any other Analog Telephone Adapter (ATA), or you could use Softphones installed on employee PCs such as X-Lite (free), or similar.

Good Luck!

http://www.gloryhoundz.com/

Try asterisk.

Just playing around I set up a 10 extension inter office VoIP system using this system in about 20 minutes on an old laptop. It's open source, free, and has a great a community behind it.

http://www.asterisk.org/ - Asterisk is a good place to start..

asterisk . try asterisk@home for a quick install/demo of asterisk's power

Asterisk.

Pure genius.

I'd love to know the technology behind this. Given Google's commitment to FOSS, I would venture to guess that they are using Asterisk somewhere in the mix since it's one of the most flexible and mature open source telephony projects. However, Asterisk isn't known for scaling very well when you start talking about enterprise level installs. In fact, one of the biggest complaints of the Asterisk community is that VoIP providers routinely hack the source to improve scalability and stability, but then never release those changes back to the project. If indeed Google is using Asterisk, it will be interesting to see how much they support the developer community.

Sony BMG - Well, they don't think much about ripping off artists, so why should they be concerned about ripping off consumers?

OpenDocument - Why would Microsoft support anything that threatens their monopoly? DUH!

Linux and Communication Devices - Astersik anyone? Your own PBX http://www.asterisk.org/

2 cents,

Queen B

1) Install a second NIC or connect the modem directly to the server. Connection to the Internet should be though the server and connection to the Office should be though a VPN on the server.
2) Install a new IDE Hard drive in a 3.5" removable rack and tray. The drive should be than big enough for the operating system (Linux of course) and copies of some of the local desktop partitions. A telecommuter can shut down the server and bring in the HD during the day to resync and repair.
3) Install DHCP demon to allocate local IP addresses, DNS and gateway settings. If the desktops are network boot capable then install TFTP to remotely boot KNOPPIX via PXE. IF the desktop OS is constantly crashing, the user can select PXE boot, network KNOPPIX. The user can then be instructed over the phone to enable ssh server to allow remote repair and reimaging of the desktop partitions from copies on the local server.
4) Partition the desktops with as small as required C: ( or in the case of Linux the root ) partition for software. When software is install, use dd and netcat via live KNOPPIX to copy a snapshot of the partition to the server. You can allocate the remaining free space as a persistant partition where documents are stored. ( Consider hireing someone who knows how to customise Knoppix for your setup.)
5) Install/Enable VNC on all the platforms, but only allow incoming connections from the local server ( which is redirected over a SSH tunnel ).
6) For local backup, create share directories on the desktop accessable by the server. On the local server create loopback encrypted file systems, unmount and copy the images to the desktops shares in chunks, using redundantcy if enough space is available on the desktops. Checksum ( MD5 is enough ) each piece.
7) If the network load to the Office is takeing up all the available internet bandwidth or the connection is just too slow then install proxy servers on the local server and consider using a distributed filesystem ( OpenAFS is still the best ) .
8) If phone charges are eating into the budget, and the internet connection is good enough, then install Asterisk on the local server ( upgrade the server to a Celron 800Mhz or better ) and a card with enough FXS ports for each local user. Don't bother with software based phones/headsets. The phone will work when the desktop does not.
9) Set up a Linux server at the Office that operates as a thin client application server. Allow remote access though both FreeNX and VNC. Create login accounts and logins that operate as virtual meeting rooms, with multiple users logging in via VNC. Use VNCserver with a screen size of around 1000x600, that will operate via a VNC viewer on any 1024x768 desktop. Use phone based conference calling for voice -- it's a lot less hassle for the users
10) Add the ususal list of cross platform applications: Firefox, Thunderbird, Gaim, OpenOffice etc.

Do the open ten step and save yourself and your santity from all those hours driving from site to site.

If you want to do VOIP using Linux or Mac (OS X) based PC's what are your options? As a follow up question, what works with Asterisk?

Yes, I agree that this is a very important subject. Actually, the topic of "OSS: Legal Issues" should be a subject that a lot of universities contemplate taking up. There's a lot of people who don't really understand concepts like GPL and how it affects OSS projects. One ongoing row: the Asterisk Open-source PBX project has just had some "competition", where a splinter group started a fork of the code and created the OpenPBX project. While it appears on the surface to be valid, in terms of the GPL, a lot of people who don't understand the rights etc. are blowing up and crying foul -- most of them commenting about "backstabbers" and "biters of the hand that feed"..

It's a no-brainer, in terms of market and community. And it's a classic open source project, in that it ties into everything, does everything, and is used by everyone.

I'll be installing next year. The more capital behind this project, the better.

And hopefully some of that capital will go to developing Mac drivers for the PCI cards. :)

Wow, I've been waiting for this topic to show up for, what, six years now? :)

What you want is a Davis Instruments station. These stations hook up, via serial cable, to any PC. If you're running some form of *nix, I highly recommend the Device::WxM2 Perl module. I've written various collection daemons that use WxM2 to pull weather data from the station and store it in RRD format or in a PostgreSQL database. I even wrote an AGI script that allows people calling my Asterisk PBX to hear the latest weather data. I also wrote a handy widget for Konfabulator that lets you watch the weather on your Mac/PC desktop in real-time.

Shameless plug: if you decide that the Davis station is right for you, stop by my employer's website, where we have a variety of Davis Instruments choices.

One word of advice: we sell cheaper stations than the Davis models but if you are planning on putting this up on a roof and leaving it for 5+ years, you really want to go with a quality peice of equipment, not a Radio Shack toy that will disintegrate after a year in the sun.

Questions? Ask and I'll be glad to answer.

Chris

What if some [terrorist, child porn, etc.] group decided to set up a network of Asterisk or Bayonne servers, virtually circumventing any established VoIP providers? I'm not sure about Bayonne, but Asterisk is extremely easy to throw together and set up. Will they make setting up such "unlicensed" servers illegal? I shudder to think what that would do to the community at large...

Why I moved off of Exchanve Server -- I wanted my data in open formats and out of the "black box" that Exchange Server is. We moved to Exchange4Linux, which stores everything (and I mean everything) in a PostgreSQL database (18G and growing). SMTP is whatever you want, but Postfix is what they recommend. I've tried practically every Exchange replacement out there (SLES/SLOX, OpenExchange, a plethora of web-based crap, Bynari, Steltor (now Oracle's) CorporateTime, Hitachi's solution, etc., etc.) and this one is the (clear) winner in my eyes. The entire thing is written in Python, including the Outlook connector, and everything but the connector is open-source. (Outlook connectors are EUR$50/seat with discounts for volume). We still run Outlook on the desktops since that is the user interface and many here still want it, but as far as the backend is concerned, I couldn't be happier now. There is something just plain cool about being able to run arbitrary SQL queries over all of the company's emails, contacts, todos, journals, you name it... We have it tying in to our Asterisk PBX as well so, for example, the service guy who's on call gets the emergency page. The service department just maintains their Pager Calendar and I do a lookup to see who's on duty.

E4L isn't without its warts (the IMAP server is still in early development, no POP or LDAP yet), but being Open Source and also being in active development, these get polished or cut out (as necessary) in time. And I can add/change the system and get my changes contributed back. I don't have to worry about where my data went to or if the system ever crashes how to recover the data. If some weird-ass situation comes up and I need to correlate my data in some unforseen way... well now I can, and I don't need some kind of screwed-up and possibly commercial API to get it done. And most importantly for me, I don't have to worry about the system changing or being eliminated due to some other company's paradigm shift.

Seriously, what good is a closed software-only "telephony" application when VoIP is the latest craze? With SIP, I can use an adapter to connect any common phone. With Asterisk I have a complete PBX that I can program to do anything, from call routing to voicemail to menus to different behaviour based on callerid and whatnot.

Sitting in front of a PC and being able to talk to someone is so 1999.

Cheese and rice... EBAY paid this much money for simple software VoIP. Imagine if they had simply thrown $100 million at the Asterisk project with the following criteria:

1) NAT-friendly Windows and MAC VoIP client
2) NOTHING!

Seriously... If you provide clients, Asterisk already has the capability to do EVERYTHING that Skype has. They even have a NAT-friendly IAX protocol. They'd just need to develop the clients and then build a large PSTN interface.

The inefficiency in business is astounding. EBAY had money burning a hole in their pocket so their remedy was to remove the money and throw it into the fire.

I'm breathless.

I would think that this type of a system not only thwarts your average pickpockets and mail thieves, but also more ambitious criminals who are willing to go a step further. You'd have to 1) either fake the originating phone #, 2) break into the owner's home and get the actual PIN using their own phone, or 3) have personal details like last four of a SSN-type number, address, birthdate, etc., and by that time the problem is bigger than a stolen PIN.

google could save 4b on the purchase price of skype and buy digium to get asterisk instead - which interoperates with all the major voice protocols (sip, iax, h323, mgcp but not skype at present). Asterisk is approaching version 1.2 at a breakneck clip.

Off-topic, and very slightly tongue-in-cheek:

Although telcos, in the US and elsewhere, suck royal dick 24/7, having the recipient pay for incoming calls turns out to have advantages, rooted in the fact that it liberates you from the real telco {mono|oligo}poly: The PSTN numbering cartel.

Consider something like UMA, only turned inside-out: Instead of making your mobile number your "public" number, you use your SIP/IAX endpoint for that instead. Since a lot of your customers/friends/colleagues etc still use the PSTN, you give it a nice PSTN proxy through a VoIP operator that supports this.

When you're out of the house/office your SIP endpoint is forwarded to your mobile number. You can do that for zero incremental cost - to the original caller - in the US because you'll be paying the termination fees through your incoming airtime.

Add the advent of WiFi-capable mobiles and the proliferation of free hotspots in the places you're likely to use your mobile (home-work-starbucks-mall) and you have a very juicy, extremely low barrier-to-entry lever over the mobile operators.

In the rest of the world this would not be possible because the initiator of the mobile leg of the call (your asterisk box/subscription service) will have to pay the high (captive market, fun fun fun!) termination fees the mobile operator charges. This is where not paying for incoming airtime works against you: Cost is not the only problem here; the major problem is that this makes you have to buy [or subscribe to someone who buys] into the PSTN numbering cartel in order to get a number with a high enough termination "cushion" to cover the cost. The barrier to entry has been somewhat safely defended...

Of course, in the US and elsewhere, the operators are going to do everything they can to stop all this from happening. Some of it may even be meaningful, like offering decent UMA rates (don't hold your breath, this sounds too much like competition). When WiFi mobiles get introduced in the US I expect to get a laugh from their T&Cs and general brain-dead-ness when they try to both sell and cripple WiFi at the same time :-)

Having run my own OSS PBX, Asterisk for over a year now I was able to use its Call Detail Records (CDR) database to figure out that my best bet was to use Broadvoice and its unlimited in-state plan at $9.99/mo and Voicepulse with its DirectConnect! service to pay 2.4 cents/min for all other calls. I estimate my phone bills will be around $15/mo instead of paying SBC $38 just for local service!

Not to mention that I can now take and make multiple calls simultaneously.

It's all in knowing your usage when designing a provider solution but having your own PBX also gives you the flexibility to actually mix and match.

As usualy YMMV.

More people should spend some time nerding out with Asterisk and other free VOIP technology. It's way cool.

It could be just what you need. Check out Asterisk for more information. At the bottom of the article is a "interesting links". You'll see something similar to what you might be talking about doing. For software client (SJPhone, IAXComm, X-lite) - Google around for "soft phones".

I never signed up for the DO NOT CALL list because I thought it would be a political football and I could not be certain of the out come. (like if they change the law and now all these telemarketing firms would then use the DO NOT CALL list) To that end, I set up an asterisk box.

http://www.asterisk.org/

And if there is no caller id, straight to voicemail without ever ringing the phone. Call me with some telemarketing BS with a caller id (or any other BS for that matter), well then you get the blacklist and it plays an obnoxious recording with no person having to answer the phone or listen to a telemarketing voicemail. Pretty sweet.

Asterisk is an open sourced Linux-based Soft-PBX system. It will interface with just about any type of telephone or telephone network, including POTS, cell phones, VoIP phones, etc. Dump your answering machine for something REALLY cool!

set up a small business that specializes in the installation and maintenence of your software, as it would seem that it requires a somewhat specialized hardware/software configuration.

target businesses that could save money by using the open source software, and sell it just like you would anything else.

chances are that most theatres have maintenence contracts with the software vendors they're currently using. They're not going to want to lose that, so you should probably offer that as well.

This has worked very well for projects such as Asterix, a linux-based telephony system.

Your best bet would be to target small independent venues. Chances are that the larger chains run a package they developed in-house.

Asterisk. Say it with me. Asterisk.

http://www.asterisk.org/

It's http://www.asterisk.org/ and the name is Asterisk, not asterix.

Use wireless and switch all your telephones to VoIP (perhaps using the excellent Asterisk).

Lets see them audit that.

My company is considering a purchase which includes a PBX based on Asterisk.

With this, our office will have "extensions" in various cities and states, all with the same incoming and outgoing, local CA phone numbers. The PBX will be connected to standard, hard-wired telephone lines, and will operate and act as normal telephone extensions.

If you picked up one of these phones and dialed "9", you'd get a dial tone in CA, wherever you happened to be. This is a big feature, since our business does business remotely all the time, and having people in different offices in different cities on the same "internal" PBX is a BIG DEAL.

The phone may be in Phx, AZ. The dialtone comes from a switch box in Northern CA. So, what should happen when I dial 911? What happens when I dial 911 from San Diego, and get an operator near Truckee, 600 miles away? My own home/office is not much closer, at 591 miles.

This 911 locality problem is one that's only really beginning to surface, and will get alot worse before it gets better...

Sounds like an excellent opportunity to use the features of http://www.asterisk.org/Asterisk to send them all into the bit bucket. Better yet, drive 'em into psychotherapy with http://www.muzak.com/Muzak ;-)