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