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:[EMAIL PROTECTED],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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