The image you provided didn’t open so I’m not sure about the design. If you
can send some SIP flow diagram and Asterisk CLI logs maybe it’ll help
understand the problem.
On Fri, Sep 16, 2011 at 1:28 AM, Gilles wrote:
> My ISP provides an FXS port to plug a handset, which can be used to
> make free calls to (GSM) cellphones, similar to the Billion ADSL
> My plan is to install an SIP client on a smartphone, so that when I’m
> travelling, I can connect to a good wifi hotspot, register with an
> Asterisk server at home which has an FXO card, tell Asterisk the
> number I wish to dial, and have it dial out through the FXO card and
> the FXS port on the ADSL modem.
> Here’s the diagram:
> Problem is, Dahdi/Zaptel doesn’t provide call progression, so that 1)
> when Asterisk passes the call to Dahdi/Zaptel, Asterisk considers the
> call “answered” although there’s no actual phone connection yet, and
> 2) Dahdi/Zaptel doesn’t trigger an event so we know if the call was
> answered (and if yes, by a live human being rather than an answering
> machine) or if the line is still ringing.
> A so-so solution is to simply tell Asterisk to loop through a voice
> message (“This is a call from Joe Allen. Please hit any key and you
> will be connected”), so we know that a human being has answered the
> call, but I was wondering if there were a better solution.
> Is it possible for Asterisk to somehow play on channel #1 what’s
> happening on channel #2 while Dahdi/Zaptel is actually still dialing,
> so that I handle call progression manually from my cellphone and the
> callee doesn’t end up hearing that odd recorded message?
> Thank you.
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