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 <[EMAIL PROTECTED]> 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:[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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