[Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?

Wolfgang S. Rupprecht list+asterisk-users at lists.wsrcc.com
Sat Jul 31 09:44:26 MST 2004


radamson at routers.com (Rich Adamson) writes:
> Like *, it also has an internal dialplan, however understanding the
> various interactions requires some experimentation, as each of the
> interfaces seem to be considered a "gateway", and part of the dialplan
> directs calls to gw0, gw1, gw2 (etc) which correspond to physical
> interfaces in most cases.

I felt some pangs of guilt turning all that stuff off, but I couldn't
think of any time I'd want two dialplans in series.

> The box was truly targeted for the residential user where existing
> phones interface on one side, the pstn line on the other side, and
> the default call is sent to the voip interface. Disconnected (or
> failed) ethernet results in a relay flipping, tying the fxs directly
> to the fxo. Same with power failure. Nice.

I think the cut-through from the fxs to the fxo (and backwards) is via
a digital connection.  In normal use you appear to end up getting hit
by the digitization delays.  As far as I can tell the relay
cut-through is only used for power failure.

> Initial tests did not show any signs of echo, very good volume and 
> audio quality, and would probably be a good choice for small quantities
> of pstn lines (particularily soho and residential users).

I still notice some low-volume problems with
FXO->asterisk->grandstream-bt101 even though I bumped the FXO incoming (and
outgoing) gains to +12dB.  (To keep calls from the FXO->asterisk->FXS
a reasonable volume I needed to drop the gain of the fxs port to -15
(from the factory of -3).

Somebody with a real phone VU meter needs to have a look at the
Sipura-3000 FXO.  I can't believe it is off that much.  Might the
Grandstream BT-101 be really low in volume and I'm just mistakenly
blaming the volume problem on the Sipura?

> The only downside I've seen thus far (not much experience as yet) is
> that * calls to the pstn line are cut through immediately, so one 
> hears the initial dialtone from the pstn and the sending of the dtmf
> tones on all outgoing calls. Kind of annoying, but there might be 
> some config option to handle it; I've just not found it as yet. (If
> anyone knows how to handle that, sure would appreciate a suggestion.)

Given the choice between hearing dead air and hearing the tones, I
think I'd rather hear the tones.  At least I know something is
happening.

-wolfgang
-- 
Wolfgang S. Rupprecht                http://www.wsrcc.com/wolfgang/
openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch



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