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asterisk pbx


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#1 livinded

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Posted 05 January 2005 - 04:34 AM

well i heard this being talk about on binrev radio and so i go to the site and looks pretty cool though i have a few questions. 1) how does this work, do i plug my phone line into this? 2) whata re the system requirments? 3) doe sit effect the normal phone system in my house? 4) Do i need voip in order to use this or can i use it on copper?

#2 m2mike

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Posted 05 January 2005 - 05:53 AM

I just installed this myself on Redhat 9. I haven't gotten anywhere with it yet. I am definitely a n00b when it comes to the Asterisk pbx. I just want to spoof caller id with an SIP provider. That is why I installed it.

#3 Abstruse

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Posted 05 January 2005 - 06:09 AM

http://www.asterisk.org/ - Official Asterisk webpage
URL=http://www.digium.com/index.php?menu=documentation]http://www.digium.com/index.php?menu=documentation[/URL] - List of links to Asterisk documentation
http://www.digium.co...ex.php?menu=faq - Asterisk FAQ
http://www.automated...etoasterisk.htm - Getting Started with Asterisk
http://www.quescom.c...=5&reponse=what - Explains (kinda) what a PSTN gateway is.

There you go. A few minutes with Google gave me all this info. While it might not be what you're looking for exactly, it does serve to point you in the right direction. I personally don't have any knowledge of what Asterisk is other than it's the wonderful software that gives us the Bell's Mind and Telephreak PBXes, but hopefully someone else will be able to give you more detailed answers if you need them.

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#4 BlackRatchet

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Posted 05 January 2005 - 09:53 AM

1) how does this work, do i plug my phone line into this?

You can, but that requires additional hardware. Most likely to start off you would use computer soft phones over VoIP.

2) whata re the system requirments?

I've run it on a p3 450 w/128MB of RAM. I'm running it now on a dual celeron 433 server in my basement. I'm thinking about popping it on the phreaktop that is a P2-266 so I can run it as a portable server. It runs fine on my two systems, and I think it will run fine on my laptop, just slow.

3) doe sit effect the normal phone system in my house?

No. Only if you want it to.

4) Do i need voip in order to use this or can i use it on copper?

You can run it all on copper, but, like I said, it requires special hardware. VoIP is painless, I had IAXTel running within a half hour of setting up Asterisk my first time. Do you not have broadband?

(500th post! Wooooooooooooooooooooooo! :punk: )

#5 natas

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Posted 05 January 2005 - 05:21 PM

you cant use a sip provider to spoof tho. i heard that it may be possible with some, but using IAX providers (who usually offer SIP as well) is how its done properly.

#6 jedibebop

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Posted 05 January 2005 - 05:40 PM

Some guy in a channel on freenode told me he is roomates with the asterisk creator, I'll try to make him come here :P

#7 thought_poet

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Posted 05 January 2005 - 07:38 PM

Some guy in a channel on freenode told me he is roomates with the asterisk creator, I'll try to make him come here :P

There is no single 'asterisk creator'. Allot of people have worked hard in making it what it is today, and anyone stupid enough to say they did all the work deserves to be shot first in the nuts and then in the head.

With asterisk a good rule of thumb is to treat it like a gentoo box. The faster your hardware is the better it will run. Encodeing/transcodeing will always need allot of power.

BTW, allot of work on asterisk is done on gentoo and debian, so it runs best on these platforms.

#8 natas

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Posted 05 January 2005 - 10:21 PM

actually they use redhat/fedora for alot of asterisk stuff too. i think mark uses fedora.

#9 ennui

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Posted 06 January 2005 - 01:28 AM

Some guy in a channel on freenode told me he is roomates with the asterisk creator, I'll try to make him come here :P

There is no single 'asterisk creator'. Allot of people have worked hard in making it what it is today, and anyone stupid enough to say they did all the work deserves to be shot first in the nuts and then in the head.

With asterisk a good rule of thumb is to treat it like a gentoo box. The faster your hardware is the better it will run. Encodeing/transcodeing will always need allot of power.

BTW, allot of work on asterisk is done on gentoo and debian, so it runs best on these platforms.

The guy on freenode could be the creator. no one say his was the sole maker of it.

#10 SUB-S0NIX

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Posted 06 January 2005 - 01:59 AM

Well i know what asterisk is but have never expierenced it in any way shape or form. ( Yes i am ashamed ) But if one were to use asterisk with a copper line, would it only require 56k modem pci card? and a line with flash capabilities?

Im only interested because if one could get asterisk to run on say a PDA with some flavor of linux and a modem for that PDA with two RJ jacks. Then it could be possible to simulate the old Gold Box? ( Im assuming that asterisk has calling out capabilites sinces its labeled a PBX? )

#11 doug

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Posted 06 January 2005 - 02:11 AM

 
Well i know what asterisk is but have never expierenced it in any way shape or form. ( Yes i am ashamed ) But if one were to use asterisk with a copper line, would it only require 56k modem pci card? and a line with flash capabilities?


lol....naw a 56k pci card wont do with asterisk copper line.
You're gonna need an fxo card.

#12 Athnex

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Posted 06 January 2005 - 02:46 AM

I have 2 IAX providers on my Asterisk box, I have never been able to get a SIP to work, No idea why, I may just be unlucky. But I only plan on using it to dial in. Digum cards seemt to go for about 300 bucks for 4 FXO lines. However, I did see a knockoff clone card for 55 bucks with 2 FXO outs. Sounded like a good deal if it worked good.

#13 jedibebop

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Posted 06 January 2005 - 05:17 AM

Some guy in a channel on freenode told me he is roomates with the asterisk creator, I'll try to make him come here :P

There is no single 'asterisk creator'. Allot of people have worked hard in making it what it is today, and anyone stupid enough to say they did all the work deserves to be shot first in the nuts and then in the head.

With asterisk a good rule of thumb is to treat it like a gentoo box. The faster your hardware is the better it will run. Encodeing/transcodeing will always need allot of power.

BTW, allot of work on asterisk is done on gentoo and debian, so it runs best on these platforms.

Who Made This?

Asterisk was originally written by Mark Spencer of Digium, Inc. Code has been contributed from open source coders around the world, and testing and bug-patches from the community have provided invaluable aid to the development of this software.



#14 Strom Carlson

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Posted 06 January 2005 - 12:00 PM

Here's my asterisk setup:

Asterisk itself runs on a 1ghz PIII with 384 MB of RAM running Debian with a custom 2.6.8 kernel. The box has a Digium TDM400P with 4 FXS ports and an X100P FXO interface card, although I rarely use the FXO for anything (i can count the times I've used it on one hand). I have three telephone sets plugged into the four FXS ports (a Nortel 9417CW eats up two of the dial tones, then there's a Western Electric 2500 for dial tone number three and a cordless phone for dial tone number four). At the moment, they're all on my desk, but I may put one in the kitchen, one in the living room, etc.

I have four IAX providers for PSTN termination (Voicepulse Connect, Nufone, Voipjet, and Asterlink) as well as IAX connectivity with both IAXTEL and Free World Dialup. I also have peering arrangements with several friends so that I can dial numbers on their Asterisk boxes. Everything speaks only ulaw, and the sound quality is unbelievable. (My DSL connection, for reference, is 6 megabit down, 608kbps up)

At my office, I have a Cisco 7960 VoIP phone that I've reflashed with the SIP firmware; each of its six call appearance buttons eats up one SIP extension on my Asterisk box. The other 7960 on my desk speaks SCCP and talks to my company's CallManager, but that is Not Asterisk™ so I'll ignore that phone :)

I have several friends who have extensions off my box; they either have their own boxes to which my extension routes, or they have a SIP phone which just hangs off my box. A number of us with our own asterisk boxes are talking of unifying our dialplans and implementing a DUNDi network.

I've still got work to do on my setup (for example, I want to make the SIP extensions on my office phone act like a hunt group) but I'm mostly happy with it.

#15 natas

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Posted 06 January 2005 - 04:53 PM

I'm working on doing an installation of Asterisk with the 4 port Digium FXO card, in conjunction with SER the SIP Express Router, and SIP Phones/ATA's. SER handles SIP to SIP calls and forwards PSTN requests to the Asterisk box. We'll see how this all goes....

#16 thought_poet

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Posted 06 January 2005 - 09:04 PM

Who Made This?

Asterisk was originally written by Mark Spencer of Digium, Inc. Code has been contributed from open source coders around the world, and testing and bug-patches from the community have provided invaluable aid to the development of this software.

It's a well known fact that he likes to take credit for other peoples work. Several people have complained about not getting credit for there work on asterisk becuase he simply took out there names in the code used or changed the files data to say copyright by him only.

I know several people who have done allot of work on asterisk. Don't go telling me they should not get credit for there work.




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