Hello Sammy, The default kamailio.cfg was quite intimidating for me while trying to read it, so I got a simpler one that just proxies all requests to a backend PBX and started from there. Here's my current configuration file (Public IP masked): #!KAMAILIO

#!define IPADDRESS "X.X.X.X"

#!define FLAG_FROM_SWITCH 1 #!define FLAG_FROM_USER 2

#!define NODE_FAILURE 100 mpath = "/usr/lib/x86_64-linux-gnu/kamailio/modules" listen = "X.X.X.X"

# ------------------ module loading ---------------------------------- loadmodule "mi_fifo.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "sl.so" loadmodule "maxfwd.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "sdpops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "path.so" loadmodule "dispatcher.so" loadmodule "usrloc.so"

# ----------------- setting module-specific parameters --------------- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") modparam("nathelper|registrar", "received_avp", "$avp(s:rcv)") modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(GRP_DST)") modparam("dispatcher", "cnt_avp", "$avp(CNT_DST)") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_interval", 3) modparam("dispatcher", "ds_probing_threshhold", 1) modparam("dispatcher", "ds_probing_mode", 1) # ------------------------- request routing logic ------------------- # main routing logic

route { # Handle insane requests route(SANITY_CHECK);

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

# Determine where the requests are coming from route(CHECK_SOURCE_IP);

# Handle NAT route(NAT_DETECT);

# Handle existing SIP dialogs route(WITHINDLG);

# Process new SIP dialogs below if (is_method("INVITE|REFER")) { record_route(); }

if (is_method("REGISTER")) { add_path(); }

if (isflagset(FLAG_FROM_SWITCH)) { # don't send INVITE from SWITCH back to SWITCH, set reply route to handle NAT and forward them along t_on_reply("EXTERNAL_REPLY"); } else { # Route to one of the switches #route(TO_SWITCH_NODE); if(!ds_select_dst("1", "1")) { sl_send_reply("503", "No route available"); exit; } }

route(RELAY); }

# Failover gracefully route[TO_SWITCH_NODE] { ds_select_dst("1", "1"); t_on_failure(NODE_FAILURE); t_relay(); exit(); }

failure_route[NODE_FAILURE] { ds_mark_dst("p"); ds_next_dst(); t_on_failure(NODE_FAILURE); t_relay(); }

route[SANITY_CHECK] { if (!sanity_check()) { exit; }

if (!mf_process_maxfwd_header("10")) { send_reply("483", "Too Many Hops"); exit; }

if ($ua == "friendly-scanner" || $ua == "sundayddr" || $ua =~ "sipcli" ) { exit; }

if ($si == IPADDRESS) { exit; }

}

route[CHECK_SOURCE_IP] { if (ds_is_from_list("1")) { setflag(FLAG_FROM_SWITCH); } else { setflag(FLAG_FROM_USER); } }

# Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing #lookup("location"); if (loose_route()) { route(RELAY); } else { if (is_method("NOTIFY")) { route(RELAY); } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } }

onreply_route[EXTERNAL_REPLY] { route(NAT_TEST_AND_CORRECT); }

route[NAT_TEST_AND_CORRECT] { if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } force_rport(); } if (has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); } }

route[RELAY] { if (!t_relay()) { xlog("L_WARN", "RELAY failed!\n"); sl_reply_error(); } exit; }

route[NAT_DETECT] { force_rport(); if(nat_uac_test("19")) { if(is_method("REGISTER")) { fix_nated_register(); add_path_received(); #save("location"); } else { # Disabled, maybe unnecessary # if(is_first_hop()) # set_contact_alias(); fix_nated_contact(); }

if (sdp_content()) { fix_nated_sdp("10"); } } return; }

Regards, Iskren

On 05/05/17 16:36, SamyGo wrote:

Hi, Can you share the config file. Seems you've made changes to configuration file. Using save("location") w/o any authentication above it will result in anyone getting 200OK for a REGISTER.

Looking at the config will help point you to the right way.

Regards, Sammy

On Fri, May 5, 2017 at 10:56 AM, Iskren Hadzhinedev <iskren.hadzhinedev@ikiji.com mailto:iskren.hadzhinedev@ikiji.com> wrote:

Hi list!

I'm using kamailio 4.2 for load-balancing and failover via
dispatcher, but I'm having some NAT related issues and I was
hoping that someone might point me in the right direction.
My setup is the following:

PSTN - PBX - kamailio - NAT - client

Calls from the NATed client to PSTN and/or PBX features (e.g.
voicemail) work just fine.
However, when a call comes from PSTN to the client, kamailio sends
the INVITE to the client's RFC1918 IP and I can't figure out how
to send it to the correct destination. I tried loading the
registrar and usrloc modules and used save("location") during
REGISTER and lookup("location") just before loose_route(), but the
end result was that kamailio replied to REGISTERs with any
username/password with a 200 OK and the endpoints never registered
with the PBX.
I can attach the config file if that will help.

Thanks for all input in advance!

Regards,
Iskren

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Hi Daniel, I'm already adding a Path header and my PBX sends correctly to Kamailio. However, Kamailio tries to send to the RFC1918 IP address. I tried replacing `add_path()` with `add_path_received()` but that had no effect. I'll try with `add_contact_alias` and `handle_ruri_alias`. I'll see how it will go.

Thanks, Iskren

On 05/05/17 16:42, Daniel Tryba wrote:

On Fri, May 05, 2017 at 03:56:10PM +0100, Iskren Hadzhinedev wrote:

...

However, when a call comes from PSTN to the client, kamailio sends the INVITE to the client's RFC1918 IP and I can't figure out how to send it to the correct destination. I tried loading the registrar and usrloc modules and used save("location") during REGISTER and lookup("location") just before loose_route(), but the end result was that kamailio replied to REGISTERs with any username/password with a 200 OK and the endpoints never registered with the PBX.

If the PBX support Path, add the headers to the REGISTER in kamailio. Else you have to mangle the contact header before relaying (add_contact_alias) and use that in INVITEs to the client (handle_ruri_alias)

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