[Asterisk-Dev] SIP REINVITE

[BJ Weschke]

 Server A (IP 192.168.1.1) 
 Server B (IP(s) 192.168.1.2 [actual] 192.168.1.3 [vip])
 Server C (IP(s) 192.168.1.4)

 All servers are Asterisk installs. All servers have SIP canreinvite=yes.

 Server A calls Server B on his VIP. The call sets up fine, but the
3rd of 4th step in the dial plan is to then transfer that call on to
Server C. Server B dials server C and then begins to attempt a native
bridge between Server A and C. Server A responds back with "SIP/2.0
482 Loop Detected" assumably because the man in the middle has
different terminating/originating IP addresses and has sent an
improper invite back to A to start the briding process.

 Does ANTHM's patch from a few weeks back to chan_sip fix this
problem, or is this still a "live" issue? If it is patched, who needs
the patch in the scenario above? Just server B? or Servers A and C
too?

 Thanks. 

 BJ