Re: OGo and Asterisk

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?)

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

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? 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.
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?

Sorry for being an idiot here ;-)

>>    - that is: can I use my existing phone?
> Yes. It uses (open source hardware!) cards for both trunk lines and
> extensions

Hu? What is "open source hardware"? This just says that the 
specifications for building the hardware are open?
But in any case I probably need new phones (because my existing ones 
are proprietariely bound to the PBX I use) and I need new hardware to 
connect the telco line?

>>    - or do I need to use my laptop with microphone/headset?
> Not unless you want to.

I don't want to ;-)

>> - is there something like a phone number?
> There is exactly a phone number if you plug it into a POTS (Plain Old
> Telephone System) line.

Hm. Doesn't fit yet in my mind ;-) I have a
- Linux server which some (additional) hardware that plugs into the
   telco phone line
- the Asterisk PBX will split the line into several phone numbers for
   the various offices
- how does a call arrive at the actual phone if this isn't a VoIP phone?
   - does the Linux server need to have a (traditional phone) port for 
each
     phone in the office (how does that work with 50+ phones?)

> You can use it to dial either SIP or POTS numbers

OK. BTW: How does a SIP (VoIP?) number look like? Where/Who runs the 
registry?

> I use it so that, when I'm on the road, callers to the office can get 
> me on
> my laptop's SIP softphone....

Hm. Well, when I'm on the road, I just tell my PBX that phone calls are 
routed to my mobile. Don't see the point in that yet.
Or is the intended advantage that the calls are cheaper because they 
are run over Internet without additional mobile and flatrate payments?

Greets,
   Helge
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