On 10 de abr de 2007, at 23:05, James Harper wrote:
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)
Thanks a lot for your answer!!
But, how do I found out what's the correct impedance of lines here?
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.
I see...
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.
I think if I choose the "*" to get a dialtone it won't work because
it seems that the SPA-3000 will pick up that character and use it as
if I was trying to access its own services...
By the way, for transfering calls, will asterisk or the SPA the one
that will actually do the transfer?
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!!!
Thanks... I don't mind if the echo is small, I actually prefer a
small echo than that cutting thing... :(
Cheers,
Francis
James
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