[asterisk-users] SIP - ooh323 Bridging

[Dovid B]

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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