[SR-Users] Re: Replace ip address of source

Can I ask here if this subst function reliable enough To use for replacing ip to domain And leave everything else as it is in Contact header ?

*Antony* satskiy.a@gmail.com

ср, 4 черв. 2025 р. о 11:18 Federico Cabiddu via sr-users < sr-users@lists.kamailio.org> пише:

Hi, have a look at the uac module and its uac_replace_to function ( https://kamailio.org/docs/modules/6.0.x/modules/uac.html#uac.f.uac_replace_t... ).

Regards,

Federico

On Wed, Jun 4, 2025 at 10:04 AM disa_369--- via sr-users < sr-users@lists.kamailio.org> wrote:

...

Hello to everyone. I deployed Kamailio on Debian system. Install was successfull. I have the task to use Kamailio as proxy between Avaya PBX and Videoconference System. Outgoing calls only from IPPhone ----- Avaya PBX ---sip----> Kamailio -----sip----> Video System-IVR IVR number is 3000. Avaya Prefix for routing to Kamailio is 1234. When we dial number 12343000 the call goes to Video system with Header "TO" 12343000@ip_of_PBXAvaya and call drops. How to replace ip-address of Avaya to ip-address of Video_System? What to add in kamailio.cfg?

ip video_system - *.*.60.3 ip avaya *.*.20.10

My route config

####### Routing Logic ######## /* Main SIP request routing logic

• • processing of any incoming SIP request starts with this route
• • note: this is the same as route { ... } */

request_route { # per request initial checks route(REQINIT);

    # NAT detection
    route(NATDETECT);

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

    # handle retransmissions
    if (!is_method("ACK")) {
            if(t_precheck_trans()) {
                    t_check_trans();
                    exit;
            }
            t_check_trans();
    }

    # handle requests within SIP dialogs
    route(WITHINDLG);


    ### only initial requests (no To tag)

    # 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
    }

if ($rU==$null) { sl_send_reply("484", "Address Incomplete"); exit;

}

$du = "sip:*.*.60.3:5060"; route(RELAY); exit;

}

# Wrapper for relaying requests route[RELAY] {

    # enable additional event routes for forwarded requests
    # - serial forking, RTP relaying handling, a.s.o.
    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] { # no connect for sending replies set_reply_no_connect(); # enforce symmetric signaling # - send back replies to the source address of request force_rport();

#!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|sipvicious|VaxSIPUserAgent|pplsip") { # 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("17895", "7")) {
            xlog("Malformed SIP request from $si:$sp\n");
            exit;
    }

}

# Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return;

    # sequential request withing a dialog should
    # take the path determined by record-routing
    if (loose_route()) {
            route(DLGURI);
            if (is_method("BYE")) {
                    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. 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;
            }
    }

    # when routing via usrloc, log the missed calls also
    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 #!ifdef WITH_MSGREBUILD # apply changes in case the request headers or body were modified msg_apply_changes(); #!endif 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;
    }

#!else

    # authentication not enabled - do not relay at all to foreign

networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; }

#!endif return; }

# Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT 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;

#!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("SIP-source-address replace-origin replace-session-connection"); } else { rtpengine_manage("replace-origin replace-session-connection"); } #!else if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } #!endif

    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();
            }
    }

    if(isbflagset(FLB_NATB)) {
            # no connect message in a dialog involving NAT traversal
            if (is_request()) {
                    if(has_totag()) {
                            set_forward_no_connect();
                    }
            }
    }

#!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;
    }

    # 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);
    } else {
            $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"
                                    + $sel(cfg_get.pstn.gw_port);
    }

    route(RELAY);
    exit;

#!endif

    return;

}

# JSONRPC over HTTP(S) routing #!ifdef WITH_JSONRPC event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if(src_ip!=127.0.0.1) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>"); exit; } if ($hu =~ "^/RPC") { jsonrpc_dispatch(); exit; }

    xhttp_reply("200", "OK", "text/html",
                            "<html><body>Wrong URL

$hu</body></html>"); 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;

}

# Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); return; }

# Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } return; }

# Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } return; }

# Manage failure routing cases failure_route[MANAGE_FAILURE] { 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 return; __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender!

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