[Asterisk-Users] Help - is voip good for in-house calls?

[Francis Augusto Medeiros]

On Mon, 16 Aug 2004 11:44:53 +0200, Holger Schurig
<hs4233 at mail.mn-solutions.de> wrote:
> > My concern was if I'd have to teach folks how to dial, but I guess
> > that I can still have the option to assign a number that will give
> > immediate access to the PSTN,
> 
> In Germany, you usually use a 0 in hardware PXSes to get the PSTN dial
> tone. No problem with Asterisk to do the same.

Same here, that's what I want to do.

> > so no need to make a special dialplan to
> > acomodate the weird numbering system we have in Brazil (sometimes we
> > dial 7 numbers, sometimes 8, sometimes 12, sometimes 13, etc.)
> 
> Actually, we also have non-fixed phone numbers in Germany. I think this is
> not weird, I think this is very good. And again, Asterisk supports this.

Oh, so I how does Asterisk knows when to start dialing out the
numbers, if there are no rules?

> > This is really a great idea. See, my biggest concern is not the voice
> > quality in terms of audio, but if the conversation is allowed to flow
> > in the same way as with regular phones.
> 
> For me this is not a problem. I tested two different PA168 based phones, a
> Sipura SPA-2000 and two Grandstreams BT101s. Just the PA168 had noticable
> delay. Botht the Sipura and the Grandstreams had analog-phone-quality.
> I'm talking here from phoning inside my network, so there were no
> internet delay etc. The PA168 based ones where slower.

Hmmm, seems interesting. I'm growing to opt for this Sipura thing. To
avoid echoes, I'll opt for a BRI instead of any FXO available out
there.

> The used code has an implication to the delay. Because I use the phones
> inside my network only, I opted for alaw or ulaw, because they are
> faster. I don't care for good compression. Make all phones using the same
> codec, then Asterisk won't need to do any codec-conversion, this saves a
> millisecond (or so, see "show translations") as well.

That wil be my case as well.
 
> I also used both chan_capi and chan_zap with zaphfc to phone to and from
> EuroISDN lines as FXO. Again there was almost no delay and no echo. Never
> tried analog FXO.

Cool!
 
> So I think this setup would be ok for interrupting Brasilians. :-)

eheheh we never know, we can be VERY interrupting... ;)

> PS: search at www.voip-info.org if you don't know what I mean with PA168

I will! Thanks!

Yours,

Francis