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 = $null; route(TOVOICEMAIL); exit; } #!endif return;
I'm not sure if this will work, but I use it to change my Contact Header subst("/^Contact:.*<sip:(10.244.[0-9.]+):/Contact:<sip:62.28.235.114:/ig")
Hi, have a look at the uac module and its uac_replace_to function ( https://kamailio.org/docs/modules/6.0.x/modules/uac.html#uac.f.uac_replace_t... ).
Regards,
Federico
On Wed, Jun 4, 2025 at 10:04 AM disa_369--- via sr-users < sr-users@lists.kamailio.org> wrote:
Hello to everyone. I deployed Kamailio on Debian system. Install was successfull. I have the task to use Kamailio as proxy between Avaya PBX and Videoconference System. Outgoing calls only from IPPhone ----- Avaya PBX ---sip----> Kamailio -----sip----> Video System-IVR IVR number is 3000. Avaya Prefix for routing to Kamailio is 1234. When we dial number 12343000 the call goes to Video system with Header "TO" 12343000@ip_of_PBXAvaya and call drops. How to replace ip-address of Avaya to ip-address of Video_System? What to add in kamailio.cfg?
ip video_system - *.*.60.3 ip avaya *.*.20.10
My route config
####### Routing Logic ######## /* Main SIP request routing logic
- processing of any incoming SIP request starts with this route
- note: this is the same as route { ... } */
request_route { # per request initial checks route(REQINIT);
# NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they arerouted) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); }
# account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting }if ($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 thetransaction 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 anddiscard 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 foreignnetworks 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($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; }
# route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTNrouting if(!($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 notdefined\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!
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 arerouted) # - 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 thetransaction 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 anddiscard 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 foreignnetworks 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($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; }
# route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTNrouting if(!($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 notdefined\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!
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!
https://www.kamailio.org/wikidocs/cookbooks/5.8.x/transformations/#sreplacem...
Which can do the same as $td="whatever you wish to be here";
However replacing domains in from/to headers much more effective with uac_replace_* functions as it can convert data back.
On Wed, Jun 4, 2025, 19:51 Antonio via sr-users sr-users@lists.kamailio.org wrote:
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 arerouted) # - 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 thetransaction 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 ... ignoreand 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 foreignnetworks 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($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; }
# route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTNrouting if(!($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 notdefined\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!
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!
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!
Thank you. It was so hard to add uac.so as module. There were some errors with module RR. Well, yes, it works. uac_replace_to replaced header "TO".
-
Antonio -
disa_369＠msn.com -
Federico Cabiddu -
Fernando Lopes -
Yuriy G