nochnoi82 created an issue (kamailio/kamailio#4272)

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",
"Not allowed from $si");
exit;
}
if ($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;

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per request initial checks

NAT detection

CANCEL processing

handle retransmissions

handle requests within SIP dialogs

only initial requests (no To tag)

authentication

record routing for dialog forming requests (in case they are routed)

- remove preloaded route headers

account only INVITEs

Wrapper for relaying requests

Per SIP request initial checks

Handle requests within SIP dialogs

Handle SIP registrations

User location service

Presence server processing

IP authorization and user authentication

Caller NAT detection

RTPProxy control and signaling updates for NAT traversal

URI update for dialog requests

Routing to foreign domains

PSTN GW routing

JSONRPC over HTTP(S) routing

Routing to voicemail server

Manage outgoing branches

Manage incoming replies

Manage incoming replies in transaction context

Manage failure routing cases