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Erm, either your site isnt really so critical you can drop everything and head from the woods to fix it.. or you can pay a PFY to do it for you? btw - i liked the combination of http://www.digitalcandle.com/asterisk.html and http://lists.digium.com/pipermail/asterisk-users/2 004-January/032521.html

You don't have to ask for donations. You can go ahead and ask them to pay for developer support accounts, subscriptions to value-added services, etc. in order to fund the further development. Or you could sell complimentary products.

For comparison, look at http://www.sql-ledger.org/ and http://www.digium.com/

Froogle search: SPA-3000.

Nothing in the article, and nothing in the comments above, fully explains the benefits of Asterisk for a small business or home. Transferring calls to a second line? Voicemail to email? What else?

Froogle search for the Digium card: Wildcard TE110P

T1 hardware: 24-Port FXS Analog Gateway (SIP).

Get a channel bank (I reccomend the Adit 600) and a TE110P T1 card.

Connect the channel bank to the T1 card via a crossover cable and you have a 24 (or 23 ISDN) port FXS interface.

Acronyms:
FXS: Foreinge Exchange Subscriber
NT: Network Trminator

Asterisk-Users mailing list

The other card is the one to drive POTS lines (so the Asterisk box is what rings my house phones). The current version can be found here. That's the one that requires PCI 2.2. One port in each direction will cost $195 new.

What I did was to get a new box, move all of the usual services (homedir, Samba PDC, etc.) to it, added Asterisk and a second mythbackend. Being a P4/2.4 with 1GB RAM, it handles all this stuff without breaking a sweat :-)

As far as how small a scale you can do this, it's about $1000 (Canadian)/month for a T1 in a data centre -- but I'm sure people still do fractional T1, don't they?

As long as nobody wants to call outside the local calling area of your server, you're set. Otherwise you'd need to set up servers elsewhere or make arrangements with someone else.

/msg nickserv register mypassword

/msg nickserv identify mypassword
then /join #asterisk

Come on in and say hi!
Some links

The Wiki [voip-info.org] bar none the best resource.

The Asterisk Documentation Project [asteriskdocs.org]

more links [digium.com] (look at the "Unnoficial Links")

Mod me up! :)...

The Digium cards are crap

Care to elaborate?


Sure.

For example, their TDM400 cards, they will randomly stop working and you'll just get static on the line when you pickup. The only way to fix it is to unload and reload the kernel modules.

This problem has been widely reported by numerous people on the mailing list. Digiums only action has been to offer to send a replacement card if you call em (which does nothing).

Don't believe me? Search the mailing list

It works pretty much exactly like a normal phone system, everyone has a cisco 7960 VoiP phone plugged into their ethernet port and their computer plugged into the phone, or a switch on their desk for both. People get extensions, and dial 9 to get out, voicemail system sends you an email with a .wav file (surprisingly small). My phone rings and it's homer simpson saying "I wasn't asleep"... errr or a non-copyright version of someone that sounds like homer simpson...

The system is easy to configure (pay someone to do it initially), easy to monitor, and very powerful and flexible.

We are in Utah, have a 1.5MB DSL and a VPN to our NJ office, and they're just more extensions on the phone system.

We have had a couple problems with Asterisk, our PCI cards are sharing IRQ's and I need to fix that to rid us of a weird beeping.
Also, someone from NJ calling out gets bad calls when we're downloading stuff, We've got a QoS router, but it needs more tweaking.. if only there were 2 of me...

I've used nortel and Intellisomethin pbx's and have always hated them. I love asterisk, and have no plans to return to $20,000 pile of crap windows NT floppy disk everything is $500 extra and technical help is $200/hr phone systems again!

IIRC, doesn't Asterisk let you use any ASDI phone? As such, there should be lots of nice cheap phones you can use (such as the ones specifically recommended by Digium). You need to make sure you can get the programming codes for the phones you buy, but that isn't as difficult as it once was.

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 ;)

Freeworlddialup can use IAX too. Register for a free account, then either get a soft client (tho' I've had trouble finding a decent stable one) or get a little box of tricks from http://www.digium.com/ called the IAXy which will convert a POTS phone to a IAX VoIP phone.

Of course, running an asterisk server gives you a lot more options and is definately the geek thing to do!

All the strings prove is that they're using something based on Astrisk. It could be that they've liscenced the entire software suite a different way from the copyright holder.

If anyone's to blame, it might be Digium for not building and enforcing a branding on their non-GPL'd customers. If a company is using Digium software, it's in Digium's best interests to require some form of advertisement on the product reguarding that, similar to Intel Inside. It protects their clients, builds brand, and if the brand is strong enough, gives a statement of quality.

in case you wish to read the source

Your point (1) is faulty. Linux may be Asterisk's primary platform but since it's open source you are just as free to go ahead and run it on, say, BSD or Solaris or Mac OS X. Meanwhile ports to more esoteric platforms are certainly an option, and it's already possible to run it on Windows if you have compatibility layer software. Linux will probably be the most likely platform to benefit from Asterisk being popular, but Asterisk definitely has potential outside of Linux.

Specifically, Marc Spencer of Digium is aware of the issue and has commented on it.

It looks like the Asterisk mailing list is already aware of this.

Or giving away the software and selling the hardware to go with it like Digium does with Asterisk. There are several reasonable ways to make money from Open Source software.

But the problem is that all of them basically devalue the software and the work put into developing it in the first place. And it basically makes it impossible to make money as a small software company - you are making money as a support company, or a hardware company, and just using the software as a hook to get people interested in buying. This is a problem because these small software companies have long been where the best jobs for real software developers have been. If everybody is using Open Source software, then the jobs move to being basically plumbing/IT jobs at larger companies, where you are treated like a cog, a commodity.

I do worry sometimes that the overzealousness to make everything Open Source hurts the very programmers who generously contribute their time.

I'm a big fan of Open Source software, and I think there are a lot of exceedingly common problems that ought to have solutions provided by the Open Source community for the benefit of all, and I'm glad they are there. But there is no reason to think that every niche in the software world should or will be filled by Open Source.

The Digium IAXy is what you're talking about. They go for $100 or so.

http://www.digium.com/index.php?menu=iaxy

I've started to use Asterisk for various applications, including as a

- PSTN to VOIP gateway: combine a cheap server, asterisk, and a few $50 voicemodem cards and you've got a VOIP gateway that can connect your outside phone lines to any VOIP phone.

- VOIP to PSTN gateway: cheap server, asterisk, open VOIP provider like VoicePulse Connect, and some Digium FXS cards and you can connect every phone in your house to a VOIP network.

- PSTN/VOIP front-end to IVR gateway: cheap server, Asterisk, IVR provider like Voxeo and you can connect all of the above to custom voice recognition applications. (Asterisk has some built in IVR but its limited today.)

Several companies are starting to offer commercial PBX products based on Asterisk, including http://www.signate.com/ and http://www.fonality.com/.

In summary, Asterisk is becoming an amazing "telephony widget" - it can address a variety of telephony solution requirements, depending on how you configure it.

Gentoo has the ebuild information here

And if your a hardcore BSD person... check out this page about Asterisk on BSD

Hum... much like the senior citizens... Gentoo and BSD may serve a purpose.

The Asterisk Wiki
Note: the wiki search is useless. Search with google instead, use "searchterm site:voip-info.org" (without quotes).

The Asterisk Documentation Project

The Asterisk Mailing Lists
Note: to search the lists use google again. "searchterm site:lists.digium.com" (without quotes)" in google.

the #asterisk chat room on irc.freenode.org. Drop by and say hello.
Note that due to problems with massive spambot attacks regisitration is required to join the channel. Simply type
/msg nickserv register mypassword
/join #asterisk

The next time you join you will need to type
/msg nickserv identify mypassword

With call screening, you can set it up to ignore the CID and they are asked, every time, for their name. This is recorded and your extension is dialed. You answer and it tells you that someone introducing themselves as: wants to talk to you, and you can either talk to them immediately, send them to voice mail, or give them one of two sendoffs before it hangs up on them. (One slow and tortuous, the other quick and polite).

While they wait, they are serenaded with whatever Music on Hold you want to subject them to. If you want to use CID, you can database your decision, and it will be used in the future to decide how to handle the call. You can even store the recorded introductions they provide, and use them on a PA if you so desire.

CID can be fun to play with, but if its non-reliability goes over some threshold of pain, you can drop it and still avoid picking up the phone for callers whose voice you don't recognize.

These fixes have been submitted to the bugzilla database, and will most likely be included in Asterisk when the voice prompts are done in the same voice as all the others.

SO, I guess you could say that if Asterisk is being used to provide CID spoofing, it can also be used to thwart the anonymous caller!

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....

That's why the best approach from a business perspective seems to be bundling or packaging fancy software with hardware, services or both. The software may be the hook to get people in, and you might even give it away (and while you're at it, make it Open Source, it makes your customers happy). But tie it to your expensive hardware. Or just convince companies that it works best with your expensive hardware. Or that your expensive services personnel are best equipped to customize or build value-added functions on top of it.

This is the whole reason that quite a few tech businesses have embraced Open Source. It's not a function of their love of the community.

You can use any decent recycled PC for the * server. I'm running a Duron 850 with 256MB RAM, but I understand even slower will work.

There are actually a couple of ways of doing it. You can use something like Voicepulse Connect and for $8 per month for an incoming number (48 states) and/or about $.03 a minute get calling anywhere in the USA or Canada. Even outbound local numbers cost this way, though.

The other option is to buy a digium fxs/fxo card and plug into the regular PSTN. The card can run as little as $150, I believe. Check out Digium for more info there.

A good way to get cheap long distance (at less than $.03 a minute) is to go with * and use the PSTN and Voicepulse connect (it is only a setting or two to make * do the differentiation). You route the local calls through the PSTN, and the long distance ones through Voicepulse connect. It depends, of course, on how much calling you do.

One caveat - VOIP and 911 emergency do not go well together.

Cheers,

s.

You need a PC with some recent flavor of Linux. Then you buy a card from Digium to interface with T1, ISDN, VOIP, or POTS lines. For example, this card will support a mix of standard analog phones and POTS lines up to 4 devices/lines.

You need a PC with some recent flavor of Linux. Then you buy a card from Digium to interface with T1, ISDN, VOIP, or POTS lines. For example, this card will support a mix of standard analog phones and POTS lines up to 4 devices/lines.

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

/msg nickserv register mypassword

/msg nickserv identify mypassword
then /join #asterisk

Come on in and say hi!
Some links

The Wiki
The Asterisk Documentation Project
Andy's Getting Started With Asterisk Guide (it's written for a old version of asterisk, but still useful)
ManxPower's site
For some advanced examples see John Todd's site
Also read all files in ./asterisk/doc after you download Asterisk.

more links (look at the "Unnoficial Links")

Mod me up! :)...

I've just barely started playing with it, but it's pretty easy to use once you get the hang of it. It even comes with prerecorded messages such as "all members of our household are currently dealing with telemarketers", "somethings *terribly* wrong", and one that's just angry monkeys screaming for 20 seconds.

Here are some great resources for getting started:
http://www.digium.com/handbook-draft.pdf

and a good soft phone (x-lite) at http://www.xten.com/

(Mark offered to give a free IAXy to the person who got this slashdot story posted) :-)

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.

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/

If you're a geek willing to put some time into learning VOIP and Asterisk, the options are endless.

How about this? Her work would give her a "desk" with an analog phone. You put an old Linux PC at the "desk" with a Digium FX0 card. You then have another PC at her home with with a VOIP phone jack or a headset with SIP software (like this Windows or this Linux) or run Asterisk on her home Linux box and run IAX between the two.

Reliability would depend on the reliability of the IP connection between home/work. Because of Internet delay (and possibly delay from your VPN encryption), there may be a noticable delay on the connections, so it may feel more like a cell phone conversation than a land line.

If you don't have time to tinker and really care about reliability, just get a $30 nationwide unlimited plan from your local phone company or long distance provider (BellSouth/MCI/AT&T), expense it to work, and be done with it.

NuFone is good for outgoing long distance calls. They charge in 15 second increments to many numbers (others are 30 or 60 seconds) and are pretty darned cheap compared to other providers.

I have great luck with Vonage for my local calling (North America, flat rate is like, $45 p/m and gets you all the dandy doodads). I also have Asterisk setup to receive faxes and Email them to me, so far no corrupted pages at all and the bandwidth usage is pretty reasonable.

I have this setup on my Asterisk box (Vonage attaches using an X100P card ($100 from Digium for the real-thing, clones have been spotted for cheap including $0.99 but YMMV), NuFone is native IAX).

Cordless phone is attached using a Grandstream Ata-286, so I can wonder around the house with a cordless headset whilst talking to who-ever using VoIP.

and don't forget to register your number on e164.org, for native voip ;)

This is an incumbment free zone

Also, the Wildcard X100P is totally unrelated to the Tormenta cards you mention.

Finally, without the Asterisk software, most definitely written in vast majority by Digium, those Wildcards don't exactly do much...

But that's just not "the right thing to do". Asterisk development is paid for in part by sales of this hardware. Buy it from Digium, and you get support as well. I had a problem compiling the zap drivers when I got mine. When I called, the phone was picked up immediately, by a real person who knew exactly what they were talking about. Digium support actually SSHed into my box and fixed it/showed me what I was doing wrong. The support is well worth the price, especially if you are building a production server. Or if your time is worth anything at all for that matter.

1) The AT standard for voice is half-duplex: it can't listen and play at the same time. That won't work for a PBX.

2) There are plenty of softmodems that can do full-duplex voice. However, their API's are not documented, and therefore are not supported.

An X100P or an FXO module for the TDM400P, in fact, are little different than a normal softmodem: they just have documented API's for dealing with voice. And, of course, buying the products from Digium helps support the people who make all of this possible...

1) The AT standard for voice is half-duplex: it can't listen and play at the same time. That won't work for a PBX.

2) There are plenty of softmodems that can do full-duplex voice. However, their API's are not documented, and therefore are not supported.

An X100P or an FXO module for the TDM400P, in fact, are little different than a normal softmodem: they just have documented API's for dealing with voice. And, of course, buying the products from Digium helps support the people who make all of this possible...

1) The AT standard for voice is half-duplex: it can't listen and play at the same time. That won't work for a PBX.

2) There are plenty of softmodems that can do full-duplex voice. However, their API's are not documented, and therefore are not supported.

An X100P or an FXO module for the TDM400P, in fact, are little different than a normal softmodem: they just have documented API's for dealing with voice. And, of course, buying the products from Digium helps support the people who make all of this possible...

1) The AT standard for voice is half-duplex: it can't listen and play at the same time. That won't work for a PBX.

2) There are plenty of softmodems that can do full-duplex voice. However, their API's are not documented, and therefore are not supported.

An X100P or an FXO module for the TDM400P, in fact, are little different than a normal softmodem: they just have documented API's for dealing with voice. And, of course, buying the products from Digium helps support the people who make all of this possible...

1) Buying the Asterisk-compatible card does not help out the company who has done 95% of the development, both hardware and software, that makes that board do something: Digium.

2) Digium has announced an FXO module for the TDM400 board that replaces the X100P. In other words, you can add up to 12 FXO (talks to telco) or FXS (talks to telephones) interfaces in the same computer, instead of just a couple.

There's nothing immoral about buying the off-brand X100P's, but it doesn't help you to get the next version of the X100P developed, or the software to make it work...

Now, if Skype made some users share their modems (yes, I hear some people still use those things), then it would be a true P2P interface to the local phone system, which I believe is the only thing that could give Skype a real chance. Any thoughts on this?

Would a typical modem chipset be adequate for this? Assuming even a traditional Rockwell voicemodem.. I think you'd need to be running a FXO board or something

The Digium cards seem a mght expensive, but there are definately cheaper then channel banks. But don't worry the Asterisk software can handle H.323, SIP and IAX (asterisk's own VoIP protocol). So you can use hard phone, soft phones and hard soft phones?!? (e.g. Cisco VoIP phone)

I've installed two of the PSTN (FXO) cards, and phone (TDM) card in a spare server with Asterisk. The cards sound and work great. No hint that the call is travelling via my computer. I'm going to be spending this week-end configuring asterisk as my Dual Line/3 Extension Home PBX.

Look at the IAX protocol that Asterisk PBX uses. Several devices are coming out now that talk this protocol.

Main selling points are:
#1 It works VERY well
#2 Only 1 port is ever used so NAT fowarding fixes all NAT issues
#3 Is a full PBX level intercommunication protocol so you can have any device using it do very advanced things that SIP and H323 only wish they could do well. (example... line in use indication for secretaries phones)

Virbiage is preparing to sell there FT201 based on IAX protocol and Digium (makers of Asterisk) are beginning mass production on their "IAXy" which is an ATA brick for analog phones.

Setup an asterisk pbx server, and signup with any number of VoIP providers who support G.711 codecs (like Voicepulse or their no bells service, Voicepulse Connect service). Plug your fax machine into a TDM400p card from digium.

Another option, pickup a Grandstream HandyTone 286 (from here for instance) or a Sipura SPA-2000 (from here for instance) (SIP devices, plug a regular phone, or fax, into it) instead of the asterisk box, but it gives you less flexibility. Both devices would work with the Voicepulse services, or most any other true SIP based VoIP service.

This works, been able to fax to people over Pulver's Free World Dialup service without any problems using both types of setup.

Setup an asterisk pbx server, and signup with any number of VoIP providers who support G.711 codecs (like Voicepulse or their no bells service, Voicepulse Connect service). Plug your fax machine into a TDM400p card from digium.

Another option, pickup a Grandstream HandyTone 286 (from here for instance) or a Sipura SPA-2000 (from here for instance) (SIP devices, plug a regular phone, or fax, into it) instead of the asterisk box, but it gives you less flexibility. Both devices would work with the Voicepulse services, or most any other true SIP based VoIP service.

This works, been able to fax to people over Pulver's Free World Dialup service without any problems using both types of setup.