Hello everyone,I'm doing some custom Kamailio configuration to achieve few things but seem to be stuck with SIP trunks. I'm currently using Kamailio 4.4.5 and here is what I have at the moment:1. Asterisk 11.25.1 servers behing Kamailio using RealTime for SIP peers.2. Route that checks if packet was sent from one of configured SIP trunks. This part works correctly:route[FROMPSTN] {xlog("== FROMPSTN ==\n");$var(socket) = $si + ":" + $sp;xlog("**** $var(socket)\n");xlog("**** $(sht(trunks_kamailio=>$var(socket)))\n"); if ($(sht(trunks_kamailio=>$var(socket))) != $null) { xlog("**** From $(sht(trunks_kamailio=>$var(socket))) PSTN\n"); return 1;}xlog("**** Not from PSTN\n");return -1;}3. Here is a fragment that sends INVITE packet to TOASTERISK route if it was received from one of trunks. I have 4444444444@XXX.XXX.XXX.XXX hardcoded for now, and in my configuration 4444444444 and XXX.XXX.XXX.XXX represent my real caller id number and carrier's IP address: if(route(FROMPSTN)) {uac_replace_from("sip:4444444444@XXX.XXX.XXX.XXX"); route(TOASTERISK);}4. TOASTERISK basically has folloving fragment that load balances traffic between Asterisk nodes:if(!ds_select_dst("0", "4")) {xlog("**** Call was not sent to none of Asterisk hosts\n");if(!is_method("REGISTER")) {xlog("**** Method is not REGISTER\n");send_reply("404", "No destination");}xlog("**** Dispatcher fail\n");exit;}So when I'm making a call to my test number, I can see following incomming packet in Asterisk from Kamailio. From header has correct caller id number and carrier's IP address (192.168.88.5 is Kamailio's internal IP address):And when Asterisk tries to find a maching peer for this incoming call, it is using Kamailio's instead of Carrier's IP address. So it is actually finds one of peers (101-XXXXXXX) that is being authenticated on Kamailio as many others:I'm not sure if Asterisk is just using packet's source IP to lookup for corresponding peer or something else from it's content. I will really appreciate any help on this.Thanks a lot!