[asterisk-users] Getting Asterisk out of the RTP media path

[SIP]

When calling from our SIP proxy through Asterisk to the PSTN provider, 
we support reINVITES which tend to work seamlessly.

However, when creating a call file which essentially connects a call 
from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP 
media path. I understand that this is sort of the idea behind a bridged 
channel, but is there any way to avoid it? Is there any way to say 
"Connect this number and this number and then get out of the way,"  or 
is this a design limitation?

N.