Re: [SR-Users] Manipulating SDP IP for Inbound Calls

I tried using nathelper in the following way -  fix_nated_sdp("2","XX.XX.XX.XX"); - it still shows my internal IP. Attaching my request routes, can you kindly check and see if I am using it correctly? *SDP* . v=0. o=FreeSWITCH 1596486133 1596486134 IN IP4 172.18.0.40. s=FreeSWITCH. c=IN IP4 172.18.0.40. t=0 0. m=audio 43954 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. *REQUEST ROUTES* request_route {     setflag(22);     route(REQINIT);     if (is_method("CANCEL")) {         if (t_check_trans()) {             route(RELAY);         }         exit;     }     route(WITHINDLG);     if(t_precheck_trans()) {         t_check_trans();         exit;     }     t_check_trans();     remove_hf("Route");     if (is_method("INVITE|SUBSCRIBE")){// && is_present_hf("X-SESSION-ID")){       record_route();     }     if (is_method("INVITE")) {         setflag(FLT_ACC);     }     if ($rU==$null) {         sl_send_reply("484","Address Incomplete");         exit;     }     route(OUTGOING);     route(PSTN);     route(INCOMING);     route(RELAY); } route[REMOVE_X_HEADERS] { if(is_present_hf("X-SESSION-ID")) {         remove_hf("X-FS-Support");    remove_hf("X-Src");    remove_hf("X-DESTINATIONS");    remove_hf("X-SESSION-ID");     }         xinfo("Remove X Headers; Contact Header is $ct"); } route[INCOMING] {     if(is_present_hf("X-SESSION-ID")) {       return;     } if(ds_is_from_list("4")) { route(TRANSLATE_SRC_IN); } route(REQUEST_PERMISSIONS); fix_nated_sdp("2","XX.XX.XX.XX");     exit; } route[REQUEST_PERMISSIONS] {     $var(body) = 0;     $var(from) = $fU;     if($(var(from){s.substr,1,4})=="0972") {         $var(from)=$(var(from){s.substr,2,0});         $fU = $var(from);     }     jansson_set("string", "from", "$var(from)", "$var(body)");     if(is_present_hf("Diversion")) {         xlog("L_INFO", "Call has been forwarded.");         jansson_set("string","to","$oU","$var(body)");     } else {    jansson_set("string","to","$tU","$var(body)");     }     jansson_set("string","forcepstn","false","$var(body)");     jansson_set("string", "source", "EGRESS", "$var(body)");     $http_req(all) = $null;     $http_req(method) = "POST";     $http_req(hdr) = "Content-Type: application/json";     $http_req(hdr) = "Accept: application/json";     $http_req(hdr) = "Connection: keep-alive";     $http_req(body) = $var(body);     $var(re_url)= "https://VNVHOST/voiceandvideo/makeCall";     t_newtran();     if (http_async_query("$var(re_url)", "REQUEST_PERMISSIONS_REPLY") < 0) {         t_reply("500", "Server Internal Error");         exit;     } } route[REQUEST_PERMISSIONS_REPLY] {     if ($(http_err{s.len})) {         xlog("L_ERR","Got error from server 1");    t_reply("500", "Server Internal Error");         exit;     } else if ($http_rs != 200) {         xlog("L_ERR","Got error from server 2");         t_reply("500", "Server Error");         exit;     }     # Populate dialog variables for CDR Creation     $var(count) = 0;     jansson_get("list",$http_rb,"$dlg_var(destinations_array)");     jansson_get("msgID", $http_rb, "$dlg_var(session_id)");     jansson_get("resultCode",$http_rb,"$dlg_var(resultCode)");     if($dlg_var(resultCode)!=0) {         t_reply("500","Server Internal Error");         exit;     }     ## EGRESS Server Information     route(ADD_TELEMESSAGE_HDRS);     $var(setid) = "1";     if(!ds_select_dst("1", "4")) {         send_reply("404", "No destination");         exit;     }     route(RELAY);     exit; } route[RELAY] {     if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {         if(!t_is_set("branch_route")) {             t_on_branch("MANAGE_BRANCH");         }     }     if (is_method("INVITE|SUBSCRIBE|UPDATE")) {         if(!t_is_set("onreply_route")) {             t_on_reply("MANAGE_REPLY");         }     }     if (is_method("INVITE")) {         if(!t_is_set("failure_route")) {             t_on_failure("MANAGE_FAILURE");         }     }     if (!t_relay()) {         sl_reply_error();     }     exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD     # flood detection from same IP and traffic ban for a while     # be sure you exclude checking trusted peers, such as pstn gateways     # - local host excluded (e.g., loop to self)     if(src_ip!=myself) {         if($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($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") {         # silent drop for scanners - uncomment next line if want to reply         # sl_send_reply("200", "OK");         exit;     }     if (!mf_process_maxfwd_header("10")) {         sl_send_reply("483","Too Many Hops");         exit;     }     if(is_method("OPTIONS") && uri==myself && $rU==$null) {         sl_send_reply("200","Keepalive");         exit;     }     if(!sanity_check("1511", "7")) {         xlog("Malformed SIP message from $si:$sp\n");         exit;     } } # Handle requests within SIP dialogs route[WITHINDLG] {     xlog("L_ERR","Entering withindlgs");     if (!has_totag()) return;     # sequential request withing a dialog should     # take the path determined by record-routing     if (loose_route()) {         xlog("L_ERR","loose route");         route(DLGURI);         if (is_method("BYE")) {             xlog("L_ERR","BYE recevied in loose route");             setflag(FLT_ACC); # do accounting ...             setflag(FLT_ACCFAILED); # ... even if the transaction fails         } else if ( is_method("ACK") ) {             # ACK is forwarded statelessly             route(NATMANAGE);         } else if ( is_method("NOTIFY") ) {             # Add Record-Route for in-dialog NOTIFY as per RFC 6665.             xlog("L_ERR","*****************1 Adding rr");           record_route();         }         route(RELAY);         exit;     }     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             route(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")) return;     if(isflagset(FLT_NATS)) {         setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING         # do SIP NAT pinging         setbflag(FLB_NATSIPPING); #!endif     }     if (!save("location")) {         sl_reply_error();     }     exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL     # search for short dialing - 2-digit extension     if($rU=~"^[0-9][0-9]$") {         if(sd_lookup("speed_dial")) {             route(SIPOUT);         }     } #!endif #!ifdef WITH_ALIASDB     # search in DB-based aliases     if(alias_db_lookup("dbaliases")) {         route(SIPOUT);     } #!endif     $avp(oexten) = $rU;     if (!lookup("location")) {         $var(rc) = $rc;         route(TOVOICEMAIL);         t_newtran();         switch ($var(rc)) {             case -1:             case -3:                 send_reply("404", "Not Found");                 exit;             case -2:                 send_reply("405", "Method Not Allowed");                 exit;         }     }     if (is_method("INVITE")) {         setflag(FLT_ACCMISSED);     }     route(RELAY);     exit; } # Presence server processing route[PRESENCE] {     if(!is_method("PUBLISH|SUBSCRIBE")) return;     if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") {         route(TOVOICEMAIL);         # returns here if no voicemail server is configured         sl_send_reply("404", "No voicemail service");         exit;     } #!ifdef WITH_PRESENCE     if (!t_newtran()) {         sl_reply_error();         exit;     }     if(is_method("PUBLISH")) {         handle_publish();         t_release();     } else if(is_method("SUBSCRIBE")) {         handle_subscribe();         t_release();     }     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; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH     if((!is_method("REGISTER")) && allow_source_address()) {         # source IP allowed         return;     } #!endif     if (is_method("REGISTER") || from_uri==myself) {         # authenticate requests         if (!auth_check("$fd", "subscriber", "1")) {             auth_challenge("$fd", "0");             exit;         }         # user authenticated - remove auth header         if(!is_method("REGISTER|PUBLISH"))             consume_credentials();     }     # if caller is not local subscriber, then check if it calls     # a local destination, otherwise deny, not an open relay here     if (from_uri!=myself && uri!=myself) {         sl_send_reply("403","Not relaying");         exit;     } #!endif     return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT     force_rport();     if (nat_uac_test("19")) {         if (is_method("REGISTER")) {             fix_nated_register();         } else {             if(is_first_hop()) {                 set_contact_alias();             }         }         setflag(FLT_NATS);     } #!endif     return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT     if (is_request()) {         if(has_totag()) {             if(check_route_param("nat=yes")) {                 setbflag(FLB_NATB);             }         }     }     if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return;     if(nat_uac_test("8")) { if(ds_is_from_list()){ xdbg("__META rtpengine priv->pub"); rtpengine_manage("replace-session-connection replace-origin direction=priv direction=pub"); } else { xdbg("__META rtpengine pub->priv"); rtpengine_manage("replace-session-connection replace-origin direction=pub direction=priv"); }     } else { if(ds_is_from_list()) { xdbg("__META rtpengine priv->pub"); rtpengine_manage("replace-session-connection replace-origin trust-address direction=priv direction=pub"); } else { xdbg("__META rtpengine pub->priv"); rtpengine_manage("replace-session-connection replace-origin trust-address direction=pub direction=priv"); }     }     if (is_request()) {         if (!has_totag()) {             if(t_is_branch_route()) {                 add_rr_param(";nat=yes");             }         }     }     if (is_reply()) {         if(isbflagset(FLB_NATB)) {             if(is_first_hop())                 set_contact_alias();         } xlog("L_ERR","20202020 in is_reply");     } #!endif     return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT     if(!isdsturiset()) {         handle_ruri_alias();     } #!endif     return; } # Routing to foreign domains route[SIPOUT] {     if (uri==myself) return;     append_hf("P-hint: outbound\r\n");     route(RELAY);     exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN     # check if PSTN GW IP is defined     if (strempty($sel(cfg_get.pstn.gw_ip))) {         xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n");         return;     }     # route to PSTN dialed numbers starting with '+' or '00'     #     (international format)     # - update the condition to match your dialing rules for PSTN routing     if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return;     # only local users allowed to call     if(from_uri!=myself) {         sl_send_reply("403", "Not Allowed");         exit;     }     if (strempty($sel(cfg_get.pstn.gw_port))) {         $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);     } else {         $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"                     + $sel(cfg_get.pstn.gw_port);     }     route(RELAY);     exit; #!endif     return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] {     # allow XMLRPC from localhost     if ((method=="POST" || method=="GET")             && (src_ip==127.0.0.1)) {         # close connection only for xmlrpclib user agents (there is a bug in         # xmlrpclib: it waits for EOF before interpreting the response).         if ($hdr(User-Agent) =~ "xmlrpclib")             set_reply_close();         set_reply_no_connect();         dispatch_rpc();         exit;     }     send_reply("403", "Forbidden");     exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL     if(!is_method("INVITE|SUBSCRIBE")) return;     # check if VoiceMail server IP is defined     if (strempty($sel(cfg_get.voicemail.srv_ip))) {         xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n");         return;     }     if(is_method("INVITE")) {         if($avp(oexten)==$null) return;         $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)                 + ":" + $sel(cfg_get.voicemail.srv_port);     } else {         if($rU==$null) return;         $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip)                 + ":" + $sel(cfg_get.voicemail.srv_port);     }     route(RELAY);     exit; #!endif     return; } //convert phone number from international to internal //+972112223344 <-> 011222334 route[TRANSLATE_DST_OUT] {         xinfo("__META TRANSLATE_DST_OUT");         xdbg("__META To: $hdr(To)");         xdbg("__META Regexp: NUM_TRANSLATE_OUT_RE");         if(subst_uri("/NUM_TRANSLATE_OUT_RE/0\2/"))                 xdbg("__META URI translated");         else                 xdbg("__META Not translating number in URI");         if(subst_hf("To", "/NUM_TRANSLATE_OUT_RE/0\2/", "a"))                 xdbg("__META To header translated");         else                 xdbg("__META Not translating number in to header"); } //convert phone number from internal format to international //011222334 <-> +<972>112223344 route[TRANSLATE_SRC_IN] { $var(number)=$rU;         if($(var(number){s.substr,1,4})=="+972") {  $rU="0"+$(var(number){s.substr,5,0}); } } ///////////////////////////////////////////////////////////////////////////// # Manage outgoing branches branch_route[MANAGE_BRANCH] {     xdbg("new branch [$T_branch_idx] to $ru\n");     route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] {     xdbg("incoming reply\n");     // fix_nated_contact();     xlog("L_ERR","2020202020202 Got reply $ct");     if(status=~"[12][0-9][0-9]") {         route(NATMANAGE);     } if(is_method("INVITE") && is_present_hf("P-Asserted-Identity")) { remove_hf("P-Asserted-Identity"); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] {     xlog("Failure! Going to failure route.");     route(NATMANAGE);     if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX     # block call redirect based on 3xx replies.     if (t_check_status("3[0-9][0-9]")) {         t_reply("404","Not found");         exit;     } #!endif #!ifdef WITH_BLOCK401407     # block call redirect based on 401, 407 replies.     if (t_check_status("401|407")) {         t_reply("404","Not found");         exit;     } #!endif #!ifdef WITH_VOICEMAIL     # serial forking     # - route to voicemail on busy or no answer (timeout)     if (t_check_status("486|408")) {         $du = $null;         route(TOVOICEMAIL);         exit;     } #!endif } event_route[topoh:msg-outgoing] {   if($sndto(ip)=="freeswitch") {     drop;   }   if($sndto(ip)=="81.24.193.248") {     drop;   } } Edward ------------------------------------------------------------------------ *From:* Daniel-Constantin Mierla &lt;miconda(a)gmail.com&gt; *Sent:* Tuesday, August 4, 2020 11:19 AM *To:* Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.kamailio.org&gt;rg>; Edward Romanenco &lt;edward(a)telemessage.com&gt; *Subject:* Re: [SR-Users] Manipulating SDP IP for Inbound Calls   Hello, the mangler module does not have any idea of inbound/outbound directions, so you can use it for any of them. Also, the nathelper module should have a function allowing to change the ip in the sdp, iirc. On the other hand, if you use rtppengine for the calls, then the ips should be replaced by it. Do not forget to use msg_apply_changes() in case you want those changes to be visible immediately in the configuration file. Cheers, Daniel On 29.07.20 13:18, Edward Romanenco wrote: -- Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com> www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla