[SR-Users] Route to PSTN no works

[sip user]

Hi, I have kamailio connect to Teams, and works form Asterisk -> Teams
calls.
For Teams -> Asterisk calls I'd worked using extension and register
Asterisk with that extension.

But I'd like to use direct routing with IP.

In kamailio.cfg I activate define WITH_PSTN.
I configured the IP and PORT for my PSTN.

I'm using the default route[PSTN]:

route[PSTN] {
#!ifdef WITH_PSTN
        # check if PSTN GW IP is defined
        xlog("L_INFO","PSTN ACTIVADO");
        if (strempty($sel(cfg_get.pstn.gw_ip))) {
                xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not
defined\n");
                return;
        }

        # route to PSTN dialed numbers starting with '+' or '00'
        #     (international format)
        # - update the condition to match your dialing rules for PSTN
routing
        if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")){
                xlog("L_INFO", "Error en el formato numerico!!");
                return;
        }

        # only local users allowed to call
        if(from_uri!=myself) {
                sl_send_reply("403", "Not Allowed");
                exit;
        }

        # normalize target number for pstn gateway
        # - convert leading 00 to +
        #if (starts_with("$rU", "00")) {
        #       strip(2);
        #       prefix("+");
        #}

        if (strempty($sel(cfg_get.pstn.gw_port))) {
                #$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
                xlog("L_INFO","SELECCION CON PUERTO");
                $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"
                                        + $sel(cfg_get.pstn.gw_port);
        } else {
                xlog("L_INFO","SELECCION CON PUERTO");
                $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"
                                        + $sel(cfg_get.pstn.gw_port);
        }

        route(RELAY);
        exit;
#!endif

        return;
}

And in my request_route:

 remove_hf("Route");
        if (is_method("INVITE|SUBSCRIBE")) {
                if($src_ip != "IP ASTERISK"){
                        xlog("L_INFO", "***********ROUTE PSTN***********");
                        route(PSTN);
                } else {
                        xlog("L_INFO","LLamada desde $si con puerto $sp");
                        record_route_preset("FQND:5061;transport=tls", "IP
KAMAILIO:5060");
                        add_rr_param(";r2=on");
                        route(DISPATCH);
                        route(RELAY);
                }
        }

But never see that the call go to PSTN route..

I'd made any wrong??

Thanks
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