Slashdot Mirror
Asterisk Breeds A Cottage Industry
gardel writes "The open-source PBX is popular, powerful and affordable. But setting up and maintaining Asterisk in its distributed form is a technical challenge for even the most accomplished of geeks. Now, Voxilla reports, several new companies (more than 60, at last count), smelling a good business opportunity, offer simplified graphical front-ends for Asterisk. And more are on the way."
It is primarily used for voip, actually, though handles leased lines (T1, E1, etc) perfectly well with supported hardware. Everything form $20 pots cards so you can use it as an answering machine at home to multiple T1 cards are supported... and lots of voip.
You can do everything with it, but configuration is a lot of text files in true unix fashion.. it's more of a framework than a completed solution... which is what the article is about.. asterisk is really powerful, but setting up a complicated setup is sort of, well, complicated (though I find the complexity is about right for the level of flexibility)
WRT tapping lines and rerouting calls -- having just installed an Asterisk-based phone system at work, I find myself not informing the business-types of its full capabilities just for the sake of not making them nervous. Very, very cool stuff -- though we now have some extra dependencies involved in making the phones work, we also have a fully customizable (and largely customized), featureful phone system. As it is, we're tied into a T1 for the outside world and doing VoIP (IAX when we can and SIP when we can't) to talk to the phones themselves. Features on the TODO list include integration with the CRM system (to make a note whenever a customer calls one of us or visa-versa, for instance) -- nothing about it's hard, just time-consuming.
Unfortunately, IP phones with quality full-duplex speakerphone support (unlike the otherwise excellent Sipura SPA-841s we're using) are *expensive*. (Know of a sub-$200 SIP phone with good speakerphone support? Let me know!)
Asterisk is a FOSS PBX (private branch exchange) and Voice over IP gateway. The PBX part means that you have phones on your desks that don't connect to the real phone lines unless you want to dial out of the company. The VOIP gateway means that it can talk to SIP and H323 systems, as well as having its own protocol, IAX. Most of the useful features require extra hardware, called FXO and FXS cards. These cards allow it to talk to the phone company lines and to talk to the phones on the desk. Without the extra hardware, you just have a computer that can talk to software phones and take voice mail. You cannot just use regular modems. It is very flexible, and if you have two or three offices, it can save you long distance charges by routing those calls over the internet. This is just a basic idea of what it can do, it what Asterisk is used for. Check out "Asterisk at home" for a fairly simple installation that includes a good web interface.
If he could write it himself he wouldn't need it. What he needs is something from somebody who ~already~ understand this. So do I.
It's doable, and not that hard.
The only thing to remember is that the Meridian phones are proprietary crap. So you can't just plug them into asterisk, but rather you'll have to
plug your asterisk server between the phone lines that come from the phone company and your PBX.
Then, expand your system by either buying some Sipura 2000 boxes and regular telephones, or some IP phones.
I'm a Linux noob and even I setup Asterisk@Home successfully. I bought a $6 Digium FXO card signed up with FWDout and off we go for free worldwide phone service.
No matter where you go , there you are.
I currently use an asterisk system for my business from a company called switchvox. They just sent me a box I plug into my network and it works. It's simple for me and I'm willing to pay for that. Plus their support is really nice.
It also allows me to have extensions that route to my sales person's phones at THEIR home. Our clients don't know any different and people get to work from home. There are a lot of features I don't use, but it saves us about $400/month on long distance calls and adding additional lines can be done my IT staff rather than an Avaya tech.
The immature part of the asterisk technology is not asterisk itself, but the VOIP providers that work with asterisk. I have yet to find a reliable VOIP provider that can work with asterisk, I've tried LiveVOIP (horrible horrible service), Teliax, iax.cc, voicepulse, broadvoice, and SIPPhone. If someone can become a reliable VOIP provider that works ALL the time with asterisk, they can make a ton of money. We have to use analog lines for our incoming and outgoing lines because the VOIP providers are not caught up the reliability of asterisk.
I'm doing the Documentation for AMP which is probably (IMO) the best admin tool, and it's what is used for 99% of the administration of Asterisk@Home. AMP is rapidly becoming more than just a basic interface to Asterisk tho - the current CVS handles LCR, ZAP Trunks (eg, physical connections to the PSTN via ISDN or normal 2-wire FXO/FXS), Call Groups, Inbound call queues with everything you'd expect ("Your call is 4th in the queue. Your expected wait time is 3 minutes"). The current CVS of Asterisk, when used with AMP, gives you attended transfers, call (audio) recording, and a whole pile of other stuff.
Probably the best thing for someone new to VoIP is to get the latest version of Asterisk@Home (which is 0.9 at the time of this post) and an old machine, a couple of soft-phones (VoIP software that lets you make calls from your PC using your sound card) and a FWD number and start playing.
Feel free to leave me voicemail on my FWD number - 47876 - if you have any questions or comments!
--RobSchlock Mercenary.
I'm doing the Documentation for AMP which is probably (IMO) the best admin tool, and it's what is used for 99% of the administration of Asterisk@Home. AMP is rapidly becoming more than just a basic interface to Asterisk tho - the current CVS handles LCR, ZAP Trunks (eg, physical connections to the PSTN via ISDN or normal 2-wire FXO/FXS), Call Groups, Inbound call queues with everything you'd expect ("Your call is 4th in the queue. Your expected wait time is 3 minutes"). The current CVS of Asterisk, when used with AMP, gives you attended transfers, call (audio) recording, and a whole pile of other stuff.
Probably the best thing for someone new to VoIP is to get the latest version of Asterisk@Home (which is 0.9 at the time of this post) and an old machine, a couple of soft-phones (VoIP software that lets you make calls from your PC using your sound card) and a FWD number and start playing.
Feel free to leave me voicemail on my FWD number - 47876 - if you have any questions or comments!
--RobSchlock Mercenary.
That should be voip-info.org. I'm so used to mozilla just auto-completing, I type 'voip' and push enter in my address bar - I don't think about the top domain. (Annoyed Grunt).
However, voip-info has been having significant performance issues, so I think that *not* linking to it was a good idea. It looks like it's been slashdotted just by having the VoIP meme high in the geek global awareness.
--Rob
Schlock Mercenary.
And yes, major PBX systems like Meridian are all CLI.
I used to work on a Aspect phonesystem that has the complete callflow in a GUI kind of way. Just drag and drop the different steps and you were done.
Although not completely easy, it is a lot easier to do on more complicated callflows. A lot easier then working on a sort of basic where you needed much more knowledge on another system.
Another advatages was that you could inform both management and people what happend in a phonecall by just doing a printout and follow the system. Also very easy to addept if waiting times are too long, when there are hollidays or to insert emergency messages.
Perhaps not needed if all you need is a message when you are closed and an aswering service for those that are not in. It will become handy if you have several numbers recieving larger amounts of numbers from different sources fr different reasons with different priorities.
Or even first start with one number and then want to insert extra possibilaties as your company grows, without having the need for a programmer.
I am in Belgium so what we had was naturlay first language choice, then department choice, then depending on the department another extra choice, then connection to the different people if they were in, otherwise to others. All depending on the language skills of the people as well.
e.g. see that if the person had a question about his bill that he would not be connected to the reception.
A lot more choices and options were involved and we were working on even more.
Don't fight for your country, if your country does not fight for you.
No, to connect to regular phones ("stations") you use FXS ("foreign exchange station") ports. FXO ("foreign exchange office") ports are for connecting to the phone company CO ("central office").
If you get your outgoing line from a VoIP provider such as Vonage, Packet8 or Broadvoice, you don't need any hardware for the outgoing side of Asterisk. If you don't, you only need a card that costs $6.85 + shipping on ebay.
For the stations, you either need an FXS card (about $100 per extension) or an IP phone (about $70 per phone) or a headset and software phone (about $10 per extension). Since most people aren't satisfied with the pure software phones, it's the hardware cost per extension that matters.
The Asterisk computer itself usually costs from $100 to $200; for "real" use you want a battery backup, and that's included in that estimate, as well as one FXO (outgoing) card. Then the best solution is IP phones for the stations, at whatever the cheapest you can get on ebay. You can get them for $40 sometimes, but usually it will be more.
Purely because the Telephone System is the communications hub of most businesses. It's the one thing you don't expect to go down - so reliability is critical.
Do you have some inside knowledge that indicates that Asterisk is unreliable? I hadn't heard that.
There's no vendor backup, etc - same with most Open Source software, and while that wouldn't be an issue with most other applications - PBX's are a different kettle of fish.
I don't know what you mean by "vendor backup". If you buy a Asterisk-based solution then it is backed by your solution provider. They have access to the source code in the same way that a proprietary software vendor has access to the source code. On the other hand, unlike the situation with a proprietary software vendor, there is competition between solution providers with equal access to the source code.
It's the one thing you don't expect to go down - so reliability is critical.
Google.com and Amazon.com are both based in large part on open source software. Would you say that reliability is not "critical" for their websites?
I'm by no means an open source zealot (I write proprietary software) but I can't let illogic just pass by. There is some highly reliable open source software and some highly reliable proprietary software. And there is some crappy open source and proprietary software out there.
Since the API is really open and can call your own little procedures in just about whatever script language you want makes for some really wild features being added to the Asterisk world that mystify traditional PBX people. Things like quick routing to voicemail or somewhere else based upon your AIM logged-in status.
The possibilities are huge.
I've just started cataloguing some of the more creative ones.
If you don't want to repeat the past, stop living in it.
You might try the Polycom IP-500 SIP phones. They are supposed to have great speakerphones, just barely under $200 at many places.
c om%20SoundPoint%20IP%20500
We're about to upgrade at my work. Its between the SPA-841's and the IP-500's. Both look pretty nice!
More info from the Asterisk wiki
http://www.voip-info.org/tiki-index.php?page=Poly
Background: You can't connect two ISDN devices or two modems with some kind of cross cable witout some additional tricks. To drive analog phones, you need a modem card with FXS support, for ISDN telephones, the card must support the NT-mode. E.g. the Junghanns QuadBRI card support NT and can drive up to 4 ISDN lines. The Wildcard TDM400P supports FXS can drive four analog devices. Both run fine with Asterisk.
Acronyms:
FXS: Foreinge Exchange Subscriber
NT: Network Trminator
Unfortunately, this article lends yet more support to those who like to dismiss Asterisk based on the cliche that it can only be handled by hard core Linux geeks.
Sure, if you want to use Asterisk to its full potential, then you have to learn a thing or two. But that isn't any different from any other tool, be it Apache, IIS, Oracle, PeopleSoft, Siebel, InDesign, Photoshop, Bryce, Final Cut, etc etc etc.
The important thing however is that you can get started with Asterisk very easily and without any special skillset.
The article doesn't mention anything about the fact that you can download an Asterisk installer for MacOS X along with a few configuration wizards and have a running PBX within a few minutes. It also doesn't mention that there is a similar Asterisk installer for Windows. At present, the Mac is the easiest platform to set up a basic PBX with Asterisk, but it shouldn't be too long before there will be configuration wizards for Asterisk on Windows, too.
Asterisk for MacOS X: http://www.sunrise-tel.com/
Asterisk for Windows: http://www.asteriskwin32.com/
How can we expect decision makers in companies to consider Asterisk if it is always presented as a Linux toy which requires Linux gurus to set up and run. That's precisely the kind of perception the incumbent proprietary system vendors love to promote when they pinch their overpriced stuff.
Let those people know that Asterisk is multi-platform and have them play with it on their platform of choice and there will soon be more mainstream deployments and more ease of use front ends.
Other than for Linux, Asterisk is so far available for FreeBSD, NetBSD, OpenBSD, DragonflyBSD and Irix (both through the NetBSD package manager), MacOSX/Darwin, Windows and Solaris. Zaptel drivers (to use telephony interface cards) are available or in the works for FreeBSD, NetBSD, MacOSX and Solaris. If that doesn't deserve mentioning in an article about an Asterisk cottage industry, then I don't know what does.
the macintosh asterisk mailing list http://www.astm
1. POTS lines will work. You will need an FX0 card per line. Not practical if you need a lot of lines. There are some multi Line FXO cards available. FX0=Hook up to telephone lines. There is a flavor of Intel Modem that will work as a single Line FXO card. They are pretty cheap and would be a good way to build a cheap test or home system.
2. To hook up just plain old phones to Asterisk you need FXS cards. FXS= hook phones up to Asterisk.
Or you can get VoIP phones and hook them up to a 100BaseT or 1000BaseT network. I will probably also want to use a power inserter so you can have power over ethernet or PoE. That way the phones will get their power over the network connection and will not have to have a wall wart.
Or you can use a softphone. A softphone is a program that runs under Windows, Linux, BSD, PalmOS, WinCE, or the Mac that uses your computers soundcard as a telephone.
Your best place to look is the VoIP Wiki http://www.voip-info.org/tiki-index.php.
Another good site is the Asterisk@Home project http://asteriskathome.sourceforge.net/. It is a Linux/Asterisk distro. Pop it in and you get an Asterisk box. Warning! This is NOT a live CD. It will reformat your hard drive and install Linux and Asterisk on it.
See my blog http://ilovecookes.blogspot.com/ for light hearted technical information.