asterisk users - Apr 2009 - opensips and asterisk canreinvite

Hi,

I'm using opensips as the registrar server for my users.
I am redirecting calls going out to pstn to my asterisk server.
call flow is basically:
ua --> opensips server --> * server --> sip gateway provider

if (uri=~"sip:00[0-9]*@sip\.myserver\.com") {
    xlog("L_INFO", "Call to PSTN\n");
    #strip(2);
    #prefix("011");
    rewritehostport("20.21.22.23:6050"); <--- IP and Port of *
Server
    route(1);
    exit;
}

call routing works properly, but i would like for the rtp not to go thru 
asterisk, i'm using the canreinvite option, but when i try to make a 
call, rtp debug still sees rtp passing thru the asterisk.

Sent RTP packet to 12.34.56.78:16410 (type 18, seq 063687, ts 035408, 
len 000020)
Got  RTP packet from 87.65.43.21:21376 (type 18, seq 000310, ts 074400, 
len 000030)
Sent RTP packet to 12.34.56.78:16410 (type 18, seq 063688, ts 035568, 
len 000020)
Got  RTP packet from 87.65.43.21:21376 (type 18, seq 000311, ts 074640, 
len 000030)

12.34.56.78 public IP of the UA, 87.65.43.21 IP of the SIP gw provider.

note: opensips and asterisk are on the same box.

i apologize in advance as i'm not sure if i'm sending it on the correct 
list.

regards,
nhadie