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

[Holger Schurig]

> 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.

> 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.


> 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.

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.


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.

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



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

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M&N Solutions GmbH       http://www.mn-solutions.de
Holger Schurig
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