Re: OGo and Asterisk

On August 9, 2004 6:07 pm, Helge Hess wrote:
> On Aug 9, 2004, at 23:05, Mark Harrison wrote:
> >> I still have not understood how this works or what it is supposed to
> >> do. Maybe someone could elaborate.
> >
> > Think of Asterisk as a OpenSource switchboard that happens to be
> > SIP-enabled
> > as well as supporting conventional phones.
>
> Talk to me like to someone who has no idea about hardware or telephony
> ;-)
> What is SIP? (the VoIP transport standard?)

http://www.cs.columbia.edu/sip/

> Where do I plugin my conventional phone? Do I need to buy additional
> hardware?

yes you need a piece of hardware for that and you need a peice of hardware to 
connect your server to a land line (ie: copper tel line) but you do not need 
to this as you can route all calls over the internet instead. As not everyone 
has such systems a functional private PBX will need to be integrated in part 
with the local telephone service providers line. (ie: copper line) 

> This link seems to suggest that:
>    http://www.azxws.com/asterisk/asterisk-debian-howto.html
> ---snip---
> If you want conferencing, you REALLY should have a Digium card, ztdummy
> or use zaprtc.
> ---snap---
>
> >> Some questions:
> >> - does it integrate somehow with my non-VoIP PBX?
> >
> > I guess it could. In practice, for my business, it replaces it.
>
> Hm. So what does a PBX do? 

It determines where to send the call and to whom in the office. Also it can 
reroute the call over the internet to anywhere you tell it to or to any other 
device. The server also acts as a host, so for true VOIP calls a remote user 
can connect to it and it automatically routes the call to either your 
computers softphone or a hardphone, what ever you have predetermined in 
advance either for that specific incoming call, or alternatively for all 
calls in general. 

> I get some line from the German Telekom 
> which is piped into the PBX. This in turn makes that available to the
> phones in the office which are connected to the PBX using some
> proprietary protocol.

No, consider the server a bridge between the internet and the land line but 
its all much more as its also a server for IP based calls. (VOIP systems) To 
do this yes you need hardware as well. Once the system is setup you can route 
incomming calls from Mageburg to your satelite office in Hamburg, at no cost 
as its all over the internet. Just one example Or alternatively, if you cant 
to call NA, and both ends have the appropriate software (ie: softphone) its 
all available at no cost if you are using asterisk or an alternative free 
provider.

This is growing so fast in the States that just last week it was decided that 
VOIP suppliers must make their system available for listening by government 
authorized security services. Encryption on these system will soon be removed 
as well as it will probably become illegal in the States. There is a lot of 
money invested in NA in these systems as it believed that VOIP are almost 
ready to supplant traditonal copper line based systems. It may or may not 
happen but the investment community seems to be selling it as such...:)

> This sounds like hardware stuff to me, so how can that be done in
> software? I somehow need to pipe in the Telekom line into my machine?

See the reference I send you earlier asterisk.org

Simple ascci art

user places call on normal handset
	|
	call is routed to hardware
		|
		call is then sent to server
		| 
		server determines if the call should routed over copper or over internet 
(sysadmin determined in advance)
		|
		if over copper (PBX) routes call to PCI card and out to PBX
		| 
		if over net routes call to eth(0) and out into the net 

Work also in reverse of course.

/ch


		
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