recieving "482 Loop Detected" using dispatcher mod

I'm really a fresh man on ser,please help me out. My goal is just simply distribute the in coming call to go through an asterisk, because I want to know asterisk's behaviour when dealing with the tasks that SER distrubuted. In anouther word, I just want to know how to write a config file, A (user) and B (user) registrate on opensips, and when A (user) dial to B (user), the SIP will go through opensips, but when B picks up the phone, the RTP will go through asterisk.
I have one Opensips1.4 and two asterisk1.6, I want SER to deal with SIP signals, and all the RTP flows goes through Asterisk. First, is this possible if both users are all in public network? Second, I actually haven't change anything in the original config file of opensips1.4.4, but only the route (1) [shown as below]

route[1] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("2");
t_on_reply("2");
t_on_failure("1");
}

if (!t_relay()) {
sl_reply_error();
};
exit;
}

Changed into this ----->

route[1] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
ds_select_dst("1","4");
t_on_failure("3");
t_relay();
exit();
}


if (!t_relay()) {
sl_reply_error();
};
exit;
}


And most of the others remains the same as the original.
But when I dial on the phone, the message shows on wireshark says 482 Loop Detected, and the phone just keep on ringing, but the phone which be called never ringings, WHY? where am I wrong?