On 2/23/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
On Thu, Feb 22, 2007 at 09:40:54PM +0330, Paradise Dove wrote:
> On 2/22/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
> >
> >Paradise Dove wrote:
> >> On 2/22/07, Yuan LIU <[EMAIL PROTECTED]> wrote:
> >>>
> >>> >From: Pavel Jezek <[EMAIL PROTECTED]>
> >>> >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:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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