[asterisk-users] SIP - ooh323 Bridging
r.scobie at clear.net.nz
Mon Nov 19 20:59:06 CST 2007
I have the following setup, with asterisk on a dual homed box:
It is running a recent SVN version of Asterisk 1.2 and ooh323.
The problem I have, is that despite having "canreinvite=no" in the
sip.conf, asterisk still insists in attempting to native bridge the RTP
-- Executing Dial("OOH323/192.168.0.2-540d", "SIP/polywn1") in new
-- Called polywn1
-- SIP/polywn1-0817e828 is ringing
-- SIP/polywn1-0817e828 answered OOH323/192.168.0.2-540d
-- Attempting native bridge of OOH323/192.168.0.2-540d and
This results in audio only running in the H323 to SIP direction -
nothing the other way.
There is NO NAT involved in the asterisk box.
Investigation with Wireshark shows that as the call is setup, a couple
of packets worth of RTP audio does flow in the SIP-H323 direction, until
the native bridge occurs, at which point it fails.
I realise that this may be something that is fixed in 1.4, but that is
not an option at this stage.
Can anyone offer a way of forcing asterisk to stay in the path and not
be bridged out?
More information about the asterisk-users