[asterisk-users] help with Sipura SPA 3000

[James Harper]

> > 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:
> 
> It worked perfectly! Thanks!

Just remember that having Asterisk supply the dialtone does add (a
slight) additional load, rather than it just routing calls between
endpoints. Not an issue with one or two ATA's though.

> > [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.
> 
> On the "sip_extension_in", I entered the following
> 
> exten => 0,1,NoOp(outgoing call via POTS line to No. ${EXTEN:1})
> exten => 0,2,Dial(SIP/${EXTEN:1}@LinkSysOut,120)
> exten => 0,3,Congestion()
> exten => 0,4,Hangup
> 
> However, when I press the "0", it does gives me a dialtone, but it
> doesn't seem to be delivering the tones imediately. I even suspect it
> isn't my PSTN tone after the 0. Is there something else?

A few things to check:

. ${EXTEN:1} will be empty because the extension can only be '0'. Change
it to 'SIP/LinkSysOut' instead

. I'm not sure but I think that the SPA3000 can either present a 'false'
dialtone to the SIP call on the PSTN line, take the digits, then send
them to the PSTN then connect the SIP call to it, or it can give the
real PSTN dialtone and connect the call immediately. I think the latter
is what you want but I can't remember the name of the setting. Maybe
'one stage dialling'?

. Related to the above, I think you might need to set the dialplan on
the VoIP to PSTN settings to 'none'.

HTH

James