[asterisk-users] help with Sipura SPA 3000

Jonson Player jonsonplayer at gmail.com
Thu Apr 12 10:24:23 MST 2007


Hello Francis,
I also hev asterisk and sipura. Can we chat online on gmail/yahoo. Let's
make some experiments... I hev the same problem like you.


On 4/12/07, Francis Augusto Medeiros <francis at mgate.com.br> wrote:
>
>
> On 10 de abr de 2007, at 23:05, James Harper wrote:
>
> >> 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:
>
> It worked perfectly! Thanks!
>
> > [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?
>
> Cheers,
>
> Francis
>
>
>
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