Thanks a lot for your response!

I've been thinking about it and there's something abnormal. Firs, the RLS is suppose to send the initial SUBSCRIBE to the S-CSCF instead of sending it to the I-CSCF. Then, I don't know why the S-CSCF translates the SUBSCRIBE URI (sip:testuser02@open-ims.test) to "sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0". All the routing is the default configuration of both cscfs and rls.

How can I change the routing of the RLS in order it to send the SUBSCRIBE messages to the S-CSCF?

Here's the configuration file:

route {

    # per request initial checks
    route(REQINIT);

    # NAT detection
    route(NAT);

    # handle requests within SIP dialogs
    route(WITHINDLG);

    ### only initial requests (no To tag)

    # CANCEL processing
    if (is_method("CANCEL"))
    {
        if (t_check_trans())
            t_relay();
        exit;
    }

    t_check_trans();

    # authentication
    route(AUTH);

    # record routing for dialog forming requests (in case they are routed)
    # - remove preloaded route headers
    remove_hf("Route");
    if (is_method("INVITE|SUBSCRIBE"))
        record_route();

    # account only INVITEs
    if (is_method("INVITE"))
    {
        setflag(FLT_ACC); # do accounting
    }

    # dispatch requests to foreign domains
    route(SIPOUT);

    ### requests for my local domains

    # handle presence related requests
    route(PRESENCE);

    # handle registrations
    route(REGISTRAR);

    if ($rU==$null)
    {
        # request with no Username in RURI
        sl_send_reply("484","Address Incomplete");
        exit;
    }

    # dispatch destinations to PSTN
    route(PSTN);

    # user location service
    route(LOCATION);

    route(RELAY);
}

route[RELAY] {
#!ifdef WITH_NAT
    if (check_route_param("nat=yes")) {
        setbflag(FLB_NATB);
    }
    if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) {
        route(RTPPROXY);
    }
#!endif

    /* example how to enable some additional event routes */
    if (is_method("INVITE")) {
        #t_on_branch("BRANCH_ONE");
        t_on_reply("REPLY_ONE");
        t_on_failure("FAIL_ONE");
    }

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

# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
    if(src_ip!=myself)
    {
        if($sht(ipban=>$si)!=$null)
        {
            # ip is already blocked
            xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n");
            exit;
        }
        if (!pike_check_req())
        {
            xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n");
            $sht(ipban=>$si) = 1;
            exit;
        }
    }
#!endif

    if (!mf_process_maxfwd_header("10")) {
        sl_send_reply("483","Too Many Hops");
        exit;
    }

    if(!sanity_check("1511", "7"))
    {
        xlog("Malformed SIP message from $si:$sp\n");
        exit;
    }
}

# Handle requests within SIP dialogs
route[WITHINDLG] {
    if (has_totag()) {
        # sequential request withing a dialog should
        # take the path determined by record-routing
        if (loose_route()) {
            if (is_method("BYE")) {
                setflag(FLT_ACC); # do accounting ...
                setflag(FLT_ACCFAILED); # ... even if the transaction fails
            }
            route(RELAY);
        } else {
            if (is_method("SUBSCRIBE") && uri == myself) {
                # in-dialog subscribe requests
                route(PRESENCE);
                exit;
            }
            if ( is_method("ACK") ) {
                if ( t_check_trans() ) {
                    # no loose-route, but stateful ACK;
                    # must be an ACK after a 487
                    # or e.g. 404 from upstream server
                    t_relay();
                    exit;
                } else {
                    # ACK without matching transaction ... ignore and discard
                    exit;
                }
            }
            sl_send_reply("404","Not here");
        }
        exit;
    }
}

# Handle SIP registrations
route[REGISTRAR] {
    if (is_method("REGISTER"))
    {
        if(isflagset(FLT_NATS))
        {
            setbflag(FLB_NATB);
            # uncomment next line to do SIP NAT pinging
            ## setbflag(FLB_NATSIPPING);
        }
        if (!save("location"))
            sl_reply_error();

        exit;
    }
}

# USER location service
route[LOCATION] {

#!ifdef WITH_ALIASDB
    # search in DB-based aliases
    alias_db_lookup("dbaliases");
#!endif

    if (!lookup("location")) {
        switch ($rc) {
            case -1:
            case -3:
                t_newtran();
                t_reply("404", "Not Found");
                exit;
            case -2:
                sl_send_reply("405", "Method Not Allowed");
                exit;
        }
    }

    # when routing via usrloc, log the missed calls also
    if (is_method("INVITE"))
    {
        setflag(FLT_ACCMISSED);
    }
}

# Presence server route
route[PRESENCE] {

    if(!is_method("PUBLISH|SUBSCRIBE"))
        return;

#!ifdef WITH_PRESENCE

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

    if(is_method("PUBLISH"))
    {
                handle_publish();
                t_release();
    }
    else if( is_method("SUBSCRIBE"))
    {
        $var(ret_code)= rls_handle_subscribe();

        if($var(ret_code)== 10) {
            handle_subscribe();
        }
        t_release();

    }else if(method=="NOTIFY")
    {
        rls_handle_notify();
    }

    exit;
#!endif
    # if presence enabled, this part will not be executed
    if (is_method("PUBLISH") || $rU==$null)
    {
        sl_send_reply("404", "Not here");
        exit;
    }
    return;
}

# Authentication route
route[AUTH] {
return;
#!ifdef WITH_AUTH
    if (is_method("REGISTER"))
    {
        # authenticate the REGISTER requests (uncomment to enable auth)
        if (!www_authorize("$td", "subscriber"))
        {
            www_challenge("$td", "0");
            exit;
        }

        if ($au!=$tU)
        {
            sl_send_reply("403","Forbidden auth ID");
            exit;
        }
    } else {

#!ifdef WITH_IPAUTH
        if(allow_source_address())
        {
            # source IP allowed
            return;
        }
#!endif
        # authenticate if from local subscriber
        if (from_uri==myself)
        {
            if (!proxy_authorize("$fd", "subscriber")) {
                proxy_challenge("$fd", "0");
                exit;
            }
            if (is_method("PUBLISH"))
            {
                if ($au!=$tU) {
                    sl_send_reply("403","Forbidden auth ID");
                    exit;
                }
            } else {
                if ($au!=$fU) {
                    sl_send_reply("403","Forbidden auth ID");
                    exit;
                }
            }

            consume_credentials();
            # caller authenticated
        } else {
            # caller is not local subscriber, then check if it calls
            # a local destination, otherwise deny, not an open relay here
            if (!uri==myself)
            {
                sl_send_reply("403","Not relaying");
                exit;
            }
        }
    }
#!endif
    return;
}

# Caller NAT detection route
route[NAT] {
#!ifdef WITH_NAT
    force_rport();
    if (nat_uac_test("19")) {
        if (method=="REGISTER") {
            fix_nated_register();
        } else {
            fix_nated_contact();
        }
        setflag(FLT_NATS);
    }
#!endif
    return;
}

# RTPProxy control
route[RTPPROXY] {
#!ifdef WITH_NAT
    if (is_method("BYE")) {
        unforce_rtp_proxy();
    } else if (is_method("INVITE")){
        force_rtp_proxy();
    }
    if (!has_totag()) add_rr_param(";nat=yes");
#!endif
    return;
}

# Routing to foreign domains
route[SIPOUT] {
    if (!uri==myself)
    {
        append_hf("P-hint: outbound\r\n");
        route(RELAY);
    }
}

# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
    # check if PSTN GW IP is defined
    if (strempty($sel(cfg_get.pstn.gw_ip))) {
        xlog("SCRIPT: PSTN rotuing 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(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
        return;

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

    $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);

    route(RELAY);
    exit;
#!endif

    return;
}

# XMLRPC routing
#!ifdef WITH_XMLRPC
route[XMLRPC]
{
    # allow XMLRPC from localhost
    if ((method=="POST" || method=="GET")
            && (src_ip==127.0.0.1)) {
        # close connection only for xmlrpclib user agents (there is a bug in
        # xmlrpclib: it waits for EOF before interpreting the response).
        if ($hdr(User-Agent) =~ "xmlrpclib")
            set_reply_close();
        set_reply_no_connect();
        dispatch_rpc();
        exit;
    }
    send_reply("403", "Forbidden");
    exit;
}
#!endif

# Sample branch router
branch_route[BRANCH_ONE] {
    xdbg("new branch at $ru\n");
}

# Sample onreply route
onreply_route[REPLY_ONE] {
    xdbg("incoming reply\n");
#!ifdef WITH_NAT
    if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB))
            && status=~"(183)|(2[0-9][0-9])") {
        force_rtp_proxy();
    }
    if (isbflagset("6")) {
        fix_nated_contact();
    }
#!endif
}

# Sample failure route
failure_route[FAIL_ONE] {
#!ifdef WITH_NAT
    if (is_method("INVITE")
            && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) {
        unforce_rtp_proxy();
    }
#!endif

    if (t_is_canceled()) {
        exit;
    }
}

Regards,
Andrés.

El 21/12/10 10:22, Klaus Darilion escribió:

Am 16.12.2010 13:36, schrieb "Andrés S. García Ruiz":

B.5) P-CSCF ---- SUBSCRIBE ----> Presentity???

SUBSCRIBE sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0
SIP/2.0
Record-Route: <sip:mt@pcscf.open-ims.test:4060;lr>
Record-Route: <sip:mt@scscf.open-ims.test:6060;lr>
Via: SIP/2.0/TCP 155.54.210.134:4060;branch=z9hG4bK07df.baa7cf24.0
Via: SIP/2.0/UDP
155.54.210.135:6060;received=155.54.210.135;rport=6060;branch=z9hG4bK07df.8a35e4f3.0

Via: SIP/2.0/UDP 155.54.210.136;branch=z9hG4bK07df.3688b985.0
Via: SIP/2.0/UDP 155.54.190.245;branch=z9hG4bK07df.964e0ba7.0
To: sip:testuser01@open-ims.test
From:
sip:restricted_areas@open-ims.test;tag=533cb9e91f4b999cf76861cbb9ed54ed-32d5

CSeq: 10 SUBSCRIBE
Call-ID: 7fd8dfdd-21694@127.0.0.1
Content-Length: 0
User-Agent: kamailio (3.2.0-dev1 (i386/linux))
Max-Forwards: 14
Event: presence
Contact: <sip:155.54.190.245:5060;transport=udp>
Expires: 10810
Max-Forwards: 70
Support: eventlist
P-Called-Party-ID: <sip:testuser01@open-ims.test>

The step B.5 is sent directly to the presentity testuser01. Instead of
that, I thought that message was suppose to be sent to the presence
server in the URI that is shown in Contact header.

If the P-CSCF is not forwarding the request as supposed, I guess you have to fix the routing on the P-CSCF.

klaus