[asterisk-users] directrtpsetup without reinvite

[Kristian Kielhofner]

On Mon, Nov 10, 2008 at 8:13 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:
> 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.
>

What version of Asterisk is this?  Last I heard (from Olle) this
option was very experimental and should not be used on production
systems.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com