From: "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
Date: Thu, 22 Feb 2007 10:09:07 -0600
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?
That zap channel happens to use usecallprogress=yes, and it did not have
this problem. I'm very confused about all these feature names like why
usecallprogress and callprogress (some examples use one, others use the
other), version compatibility, etc. But this particular setting does not
affect SIP/RTP connection. Come to think about it, callprogress only
affects Zap channel and should not affect RTP. There must be other things
that prevent RTP from streaming.
Don't use callprogress. It doesn't work.
Until you are desperate and callprogress is the last straw in sight:-)
Yuan Liu
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