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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