[Asterisk-Dev] monitoring a call and media path

[Wolfgang S. Rupprecht]

On a purely SIP call between two sip phones with canreinvite=yes and
the dialing on both phones being done via SIP info commands, is there
any need to keep asterisk in the RTP media path if no recording is
being made but one wants to allow a later recording via *1?  Currently
asterisk seems to stay in the loop so it can catch inband dialing, but
that isn't always needed if the phones cooperate and put the dialing
into sip msgs.

I was thinking of hacking things a bit to allow my asterisk to stay
out of the media path in the above case, but figured it couldn't hurt
to post a quick sanity check here.  Anyone see any problems?

-wolfgang
-- 
Wolfgang S. Rupprecht                http://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html