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 ? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
