From mboxrd@z Thu Jan  1 00:00:00 1970
From: Joerg Dorchain <joerg@dorchain.net>
Subject: Re: SIP conntrack defeating Asterisk canreinvite
Date: Wed, 26 Aug 2009 09:15:10 +0200
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On Tue, Aug 25, 2009 at 09:04:28PM -0400, John A. Sullivan III wrote:
> The reinvite works by the Asterisk server sending a SIP invite after the
> call has been set up. The new invite contains the address of the phone
> in the SDP portion of the packet rather than the address of the PBX.
> This should redirect the media stream to flow directly between the
> phones.  However, it appears conntrack is rewriting the SDP so that the
> address is reverted to the PBX address.

Rewriting sounds like nat. I am using conntrack_sip to be able
to have the rtp connections accepted as related to a sip
connection. Are you sure that you aren't using the sip nat helper
by change?

To have reinvites working, I needed sip_direct_media=0 as option
to nf_conntrack_cip

Bye,

Joerg

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