Hi
Call comes in from server A to server B. Call is answered by server B and immediately blind transferred to server C using Asterisk cmd Transfer. The RTP from server A is now going directly to server C - great! But RTP from server C is going to server B - not so great! RTP should be between server A and C only.
The result, one way audio.
If you are still with me, any help would be appreciated! 
Chris
The Asterisk Transfer application, when used for SIP, cannot do that, as it never touches server C. You would need a bug in server A for server C to become aware of the RTP path to server B.
This is particular true for an unanswered blind transfer (i.e. status 302), but is also true if you Answer first (REFER method) (although the only reason I can thing of for doing that is to work round some bug).
There isn’t a Transfer command, so unless you mean the Transfer application, I’m not sure what you mean.
You are probably going to need to provide a SIP trace.