I'm trying to set up a direct SIP connection and have Asterisk stay
out of the media stream. When I look at the INVITE messages, I see
that Asterisk is changing the Session Description Protocol in the
INVITE message it receives, and send a INVITE message with a different
SDP to the receiver. This is not what I want. Is there any way to make
Asterisk leave the SDP exactly like it is sent from the sender?
I have set canreinvite=yes on both participants and my dialingplan is simply:
exten => _.,1, Dial(SIP/${EXTEN},20)
and NAT is not a problem
Thanks.
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