[asterisk-users] Answer() command?

[Paradise Dove]

On 2/23/07, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
>
> On Thu, Feb 22, 2007 at 09:40:54PM +0330, Paradise Dove wrote:
> > On 2/22/07, Eric ManxPower Wieling <eric at fnords.org> wrote:
> > >
> > >Paradise Dove wrote:
> > >> On 2/22/07, Yuan LIU <yliu11 at hotmail.com> wrote:
> > >>>
> > >>> >From: Pavel Jezek <pavel.jezek at i.cz>
> > >>> >Date: Thu, 22 Feb 2007 09:39:22 +0100
> > >>> >
> > >>> >I think, this can be solved using phone autoanswer feature, look at
> > >>> wiki...
> > >>> >
> > >>> >  exten => s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer)
> > >>> >  exten => s,2,Dial(SIP/myphone)
> > >>>
> > >>> Or without.  One of my contexts is set up exactly like the original
> > >>> sample.
> > >>> Just Dial(), no Answer(). (I think I've seen textbook samples like
> > >that,
> > >>> too.)  Asterisk bridges the call when the callee picks up. (That's
> the
> > >>> main
> > >>> work Asterisk does: bridging calls.)
> > >>
> > >>
> > >>
> > >> BUT, when callprogress=yes, asterisk doesn't bridge the call and just
> > >ring
> > >> for the caller and noise for called!!
> > >> is it a bug or it's normal?
> > >
> > >Don't use callprogress.  It doesn't work.
> >
> >
> > GOOD NEWS:
> > Problem  Fixed!
> > i wrote a patch for dsp.c and chan_zap.c now both callprogress and
> answer
> > problem work fine together.
> > i also add a config option in zapata.conf to tune callprogress now it
> works
> > with over 95 percent accuracy.
>
> Great!
>
> Mind posting your patch on http://bugs.digium.com ?


my patch needs some tailoring its not very clean now!
i will post it soon. ;-)

--
>                Tzafrir Cohen
> icq#16849755                    jabber:tzafrir at jabber.org
> +972-50-7952406           mailto:tzafrir.cohen at xorcom.com
> http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir
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