[asterisk-users] help with Sipura SPA 3000

[Francis Augusto Medeiros]

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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