asterisk users - Apr 2007 - help with Sipura SPA 3000

Hi there everyone!

I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where
I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured).

However, living in Brazil, I'd like to know if there are optimal settings to
my PSTN that I should enter into the config of the device. I experience a
little bit of echo on the FXO probably because I raised the gain of that
port because I wasn't sounding loud enough.

But there are two things I would like to do with the device, and I'd
appreciate if anyone could help me out:

1 - Is there a way to stop "cutting" other people when I speak through
the
PSTN? What I mean is that, when sound is captured by my telephone, it
dimishes the other peer's voice, and sometimes it makes communication
harder, as if the line weren't full duplex.

2 - How can I gain full control to the FXS? I mean, a simple * dialed is not
sent for asterisk (the server) interpretation, probably because it's used by
Sipura's suplementary services, I don't know. Also, is it possible to
get a
dial tone from ASterisk, instead of Sipura's? My goal with this is to
provide users with direct access to the PSTN line pressing 0, instead of
collecting calls and making the call themselves, or at least making
ignorepat to work!

Cheers,

Francis

-- 
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Francis Augusto Medeiros
ICQ:7825595
Skype: francisaugusto
AIM/iChat: francisaugusto
Vit?ria da Conquista - Bahia - Brasil
"L?mpada para os meus p?s ? a Tua palavra, e luz para o meu caminho"
         --- Salmo 119:105
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> I've bought a Sipura SPA 3000, and succesfully connected it to my Mac,
> where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well
> configured).
> 
> However, living in Brazil, I'd like to know if there are optimal
settings
> to my PSTN that I should enter into the config of the device. I
experience
> a little bit of echo on the FXO probably because I raised the gain of
that
> port because I wasn't sounding loud enough.
Get the impedance settings right. An impedance mismatch will cause echo
(but may not be the only cause)
> But there are two things I would like to do with the device, and I'd
> appreciate if anyone could help me out:
> 
> 1 - Is there a way to stop "cutting" other people when I speak
through
the
> PSTN? What I mean is that, when sound is captured by my telephone, it
> dimishes the other peer's voice, and sometimes it makes communication
> harder, as if the line weren't full duplex.
I think the 'echo suppression' setting causes this. It is meant to
reduce the incoming audio (and hence the echo) while you are talking,
which can be annoying but is supposed to be less annoying than the echo
itself.
> 2 - How can I gain full control to the FXS? I mean, a simple * dialed
is
> not sent for asterisk (the server) interpretation, probably because
it's
> used by Sipura's suplementary services, I don't know. Also, is it
possible
> to get a dial tone from ASterisk, instead of Sipura's? My goal with
this
> is to provide users with direct access to the PSTN line pressing 0,
> instead of collecting calls and making the call themselves, or at
least
> making ignorepat to work!
A dialplan of '(S0<:s>)' will get your phone to jump straight into
the
's' extension in asterisk as soon as someone picks it up. From there you
can do something like:

[sip_ata_incoming]
exten => s,1,Answer
exten => s,n,DISA(no-password|sip_extension_in)

so Asterisk will give you dialtone and do the dialplan stuff for
you.
>From the 'sip_extension_in' context you can make a single
'0' or '*'
call the PSTN line.

Good luck with the echo situation. I have an spa3000 and no matter what
I do I get echo coming back to me with almost no reduction in volume!!!

James