[asterisk-users] Answer() command?

[Yuan LIU]

>From: "Eric \"ManxPower\" Wieling" <eric at fnords.org>
>Date: Thu, 22 Feb 2007 10:09:07 -0600
>
>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?

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