On 6/12/06, Roger Schreiter <[EMAIL PROTECTED]> wrote:
Hi,
I put reinvite=yes in my sip.conf.
For testing, I restricted the codecs to alaw.
I have no modifiers in my dial command.
Thus, there should be no reason not to reinvite.
Call (sip, authenticated) comes in and is forward
via SIP (not authenticated) to another asterisk box.
Unfortunately, media path still passes through the asterisk
box in the middle.
Using sip debug I even can't find any attempt of a reinvite.
Now I would like to know, why the asterisk box in the middle
does not try to reinvite.
One reason might be is if you are passing parameters in app_dial (eg.
tT, etc) that require it to listen for DTMF that would cause it to
hang on to the RTP streams rather than reinvite them away.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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