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Interview with Mark Spencer of Asterisk
comforteagle writes "OSDir has published an interview with Mark Spencer of Asterisk and Gaim about why and how he got started coding up the software platform PBX system and how it has become much more than -just- another phone system. He also shares his insights for the opportunities within the telecom industry for open source."
I have used and deployed * in a number of setups ( from large businesses to home ), and you folks should really understand something: This is the killer linux app.
Samba is great. qmail/sendmail/ect...is wonderful as well. But, as far as getting linux in the door, this is the application that will do it. For example, my first * implementation cost about 8grand ( parts and service ).
For a similar, but far less featured pbx from avaya, I was quoted 40grand. And that was a quote. Anybody here that has worked with phone venders should be chuckling right now at that number, as it amounts to a pie in the sky dream.
So, for my small business, I saved them 30 grand right up front ( likely more ). On top of that, as their needs change, so can the phone system. Just the other day they found out I was taking my desk phone home ( to play with, but also get my phone calls ). When I told them why, they were floored that the system could do that, no matter how many times I told them it could.
Larger businesses will see far more dramatic cost savings, and get more features to boot.
Mod me down with all of your hatred and your journey towards the dark side will be complete!
Private Branch eXchange.
A telephone system.
As of 10/06/03, I hate COBOL developers.
Officially it means: "Private Branch Exchange (private telephone switchboard)" In reality it is a switchboard placed inside your house or office commonly. You know, "press 1 for an outbound line" sort of thing!
It's like a router with a NAT... Only for telephones not the internet...
Generally, though, for services like Vonage or Phone Over Cable, you must use the supplied analog telephone adapter. So despite the fact that the VOIP is coming in over the internet, you actually can't just handle them with an Asterisk server. You would need to instead get a second analog-to-digital converter, and use your VOIP line as though it were an ordinary analog telephone line.
If you don't know where you are going, you will wind up somewhere else.
If you have iTunes, you can check out the latest systm video cast which features a demonstration by John Todd. Shows how to set up Asterisk. 47 minutes in length. Go to iTunes and search for "systm".
)9TSS
you might find this helpful
http://revision3.com/systm/asterisk
...when you have a termination provider capable of connecting with SIP phones.
Otherwise, when I go to a computer recycling depot, all I see is Asterisk boxes.
I have run 4 lines on my 450MHz box with no degradation at all.
You can buy cheap FXO cards for $10 and unlock Vonage Linksys PAP2s for $10 per FXS port.
Slap that together with a $25 PowerMAC 9600 and bam!
5 FXO + 10 FXS and witness the power of a fully operational PBX system for 175 bucks!
For those wishing to play with Asterisk, you can't beat Asterisk@Home. Nearly instant setup & web-based GUI config makes easy to administer too. I had it up and running in uner 10 min!
T.J. Schmitz - the man, the myth, the legend - o
Here's a link to a newer interview done with Mark Spencer last week, Jan. 19, 2006.
p lay&id=91&cast=585&castPage=
http://gabcast.com/index.php?a=episodes&query=&b=
While there certainly are the Vonages of the world, there are far more VoIP services that permit you to connect any phone you like.
It's not all that esoteric to set up, either. I didn't even bother with the various GUI configuration tools you can download. I did have better luck compiling it myself rather than using the one that Debian has packed for it, but that may have changed since I tried it.
If I were in the business of making commercial PBX systems, I'd be shaking in my boots right now. I think Asterisk will end up putting the lot of them out of business.
I'm trying to teach myself to set people on fire with my mind... Is it hot in here?
http://voip-info.org/wiki/index.php?page=Asterisk
'Go for the eyes, Boo, go for the eyes, aaarrrrrrrr!' -- Minsc
No, I don't think you do need one at all. All of the digital signal processing is handled in software. Digital/analog conversion is either done in the FXS/FXO cards, for traditional phones, or in the phone itself if you are using VOIP phones (that's why it matters what codecs the phone supports).
If you don't know where you are going, you will wind up somewhere else.
I'm in charge of replacing our to-old NEC PBX for a brand new Asterisk in our ISP.
As a Unix sysop for long time, with some knowledge in general VoIP/H.323/SIP, I would say that the jump into Asterisk is not too dificult. We use SSH/vi/etc. in our day-to-day task, so one more system is not hard to swallow.
However I would like to point out that unless you are a really small user, with standard needs, for example in a situation where Asterisk@Home resolves all your needs, or you can live using only SIP or IAX, you will have some problems.
Asterisk will be a killer-app, but it is not there yet. Each new version tends to break something, configuration switches are added or removed, new features, are added changing the way things should be done, behavior of old functionality changes, etc. Its great, but its still evolving. Just check the mailing list and you will see the kind of problems that arise, and are resolved by the community.
Evolving is a Good Thing, but you have to take that into account before jumping in.
Pablo.
Mark Spencer will be speaking on Sunday at the Southern California Linux Expo. In addition Digium will be exhibiting Saturday and Sunday.
Let us not forget some of my personal favorites.
Mod me down with all of your hatred and your journey towards the dark side will be complete!
FYI - Mark Spencer will be talking at our local Linux group tomorrow. Check www.flux.org for details.
from: http://www.freeswitch.org/docs/
l
"Licensing
Freeswitch is licensed under the terms of the MPL 1.1"
this license is *not* compatible with the gpl. even mozilla.org has stopped using this license:
Mozilla Relicensing FAQ
http://www.mozilla.org/MPL/relicensing-faq.html
mozilla is relicensing all of their code under a triple mpl/lgpl/gpl license in order to make their products compatible with the gpl. please consider doing the same with freeswitch.
read this if you need some more convincing as to why to relicense:
Make Your Open Source Software GPL-Compatible. Or Else.
http://www.dwheeler.com/essays/gpl-compatible.htm
bottom line, if freeswitch isn't gpl-compatible it's much less likely to be successful.
And if you're using IAX2, you don't need to use Zaptel normally. The only reason you'd need to use Zaptel with IAX2 is if you were doing IAX trunking, something which most people do not do.
A little bit of background for those readers unfamiliar with the issues. Telephone systems use a very strict timer of 8000 Hz. Given that this is far too heavy of an interrupt load for the PCI bus, Asterisk compromises and uses a 1000Hz interrupt, moving 8 samples for each interrupt. The RTC is close to 1000, but really isn't: it generates a 1024Hz interrupt. We compensate for the RTC clock in systems which don't have Zaptel hardware by ignoring 24 interrupts per second, thus letting us move the same 8,000 samples per second. Unfortunately, because we have to ignore some interrupts, the samples spacing is not quite equidistant, so there's some audio jitter caused by using RTC. It's not terrible, but it's not optimal.
Zaptel, in additional to the telephony interface needed, whether it be T1, E1, or analog, also generates that nice consistent 1000Hz interrupt. Note that in systems that especially interface to the PSTN digitally (i.e. T1 or E1), we absolutely MUST match the timing of the PSTN exactly. 1024 minus 24 is not "good enough". This is the primary reason why we must have an accurate timer and why Asterisk relies intrinsically upon having such a timer available. Given that you have an accurate timer available, why would you code to anything else?
The zaptel driver is a crude piece of crap that was rejected by the kernel developers. It is unfit for serious use.
I'm unaware that Zaptel has ever been submitted to the kernel developers, let alone rejected. Frankly, the code moves too quickly for us to consider letting it be maintained by the kernel developers. Even if it was in the kernel, the stock kernel driver would be unlikely to be ever used, given that it would almost always be out of date. That's quite simply just wasted effort on maintenance by kernel developers. I'm sure we can all agree that the kernel developers have more important issues to deal with. As to the zaptel driver being "unfit", well, it's used in thousands of Asterisk systems around the world with no problem at all. This declaration that it's unfit sounds suspiciously like FUD.