On Wed, 2009-08-26 at 06:59 -0400, John A. Sullivan III wrote: > On Wed, 2009-08-26 at 09:15 +0200, Joerg Dorchain wrote: > > 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 > Yes, as I was thinking after I wrote this, it is probably ip_nat_sip > since it is doing packet rewriting. So it sounds like it is a problem > without sip_direct_media which sounds like it implies upgrading my > kernel :-( Thanks - John Hmm . . . on the other hand, these connections are NOT doing NAT. If they are separated by a VPN and using RFC 1918 addresses. Why would ip_nat_sip even come into play? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsullivan@xxxxxxxxxxxxxxxxxxx http://www.spiritualoutreach.com Making Christianity intelligible to secular society -- To unsubscribe from this list: send the line "unsubscribe netfilter" in the body of a message to majordomo@xxxxxxxxxxxxxxx More majordomo info at http://vger.kernel.org/majordomo-info.html
