More nf_conntrack_sip questions

[Philip Prindeville <philipp_subx@redfish-solutions.com>]

I did a little investigation into my one-way voice issue, and noticed 
that if I don't do voice-menus (i.e. where the Asterisk box itself 
generates the first outbound INVITE, then passes-through the 2nd INVITE 
once a handset picks up) then I get two-way voice (i.e. with sending the 
call directly to the phone).  (In this topology, my Asterisk box is also 
my firewall/NATting router...)

If I enable the voice menus in the inbound dialplan, however, it can 
hear the voice menus, but not the called-party when they pick up their 
phone (extension).

So someone (either the SIP conntrack module on the Asterisk border 
firewall or else the SBC at the ILEC) is failing to look into the 2nd 
INVITE (i.e. we're not rewriting it properly as it goes by, or the SBC 
is failing to see it).

I've put traces up on ftp://ftp.redfish-solutions.com/ as:

trace-20081128-230313.br0
trace-20081128-230313.br1
trace-20081128-230415.br0
trace-20081128-230415.br1

The traces on interface "br1" are the "internal" network, with private 
192.168.1.x addresses.  The "br0" traces are after outbound NATting (and 
conntrack rewriting) has been applied.

This was done on a Linux 2.6.25.19 box with iptables v1.4.2.

Thanks,

-Philip