[asterisk-users] directrtpsetup without reinvite

[regs at kinetix.gr]

Hi,

    I want to be able to bridge two sip channels using direct RTP 
between my endpoints (Audio IP : not local) but without
using reinvites. So I set up my asterisk sip endpoints as follows:

[test1]
type=friend
host=dynamic
username=test1
dtmfmode=info
context=test_rtp
allow=all
canreinvite=no
directrtpsetup=yes

[test2]
type=friend
host=dynamic
username=test2
dtmfmode=info
context=test_rtp
allow=all
canreinvite=no
directrtpsetup=yes

... but it doesn't work. How can I ensure that the RTP is not going 
through my asterisk box and that the re-invite method is not used?

P.S. Both endpoints are using the same codec, so no codec translation 
takes place.