I doubt this will help but try also nat=yes.
----- Original Message -----
From: "Richard Scobie" <r.scobie at clear.net.nz>
To: "asterisk-users" <asterisk-users at lists.digium.com>
Sent: Tuesday, November 20, 2007 4:59 AM
Subject: [asterisk-users] SIP - ooh323 Bridging
> Hi,
>
> I have the following setup, with asterisk on a dual homed box:
>
>
>
PolyIP500(SIP)--<192.168.4.0>--Asterisk--<192.168.0.0>--Panasonic(H323)
>
> 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
> streams:
>
> -- Executing Dial("OOH323/192.168.0.2-540d",
"SIP/polywn1") in new
> stack
> -- 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
> SIP/polywn1-0817e828
>
> 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?
>
> Regards,
>
> Richard
>
>
>
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