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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.
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.
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.
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
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.