Dear all, I have number translation as below to manipulate the dialing pattern, it's works fine for it's purpose. One problem is all on net calls can connect but no voice. Any idea? route{ if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too many hops"); }; break; }; if (msg:len >= max_len) { if (method!="ACK") { sl_send_reply("513", "Message too big"); }; break; }; if (search("^From: sip:sipsak@") && (method=="OPTIONS") && (!uri=~"sip:.*[@]+.*")) { options_reply(); break; }; # !! Nathelper if (nat_uac_test("19")) { if (method == "REGISTER" || !search("^Record-Route:")) { fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via append_hf("P-hint: Proxied\r\n"); setflag(5); # Mark as NATed }; }; if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri=~"^sip:1[0-9]+@.*" && !uri=~"^sip:81[0-9]+@.*") { if (uri=~"sip:[1-9][0-9]+@.*") { prefix("81"); } else if (uri=~"sip:0[1-9][0-9]+@.*") { strip(1); prefix("81"); } else if (uri=~"sip:00[1-9][0-9]+@.*") { strip(2); }; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "1"); break; }; save("location"); break; }; setflag(1); lookup("aliases"); if (lookup("location")) { append_hf("P-hint: usrloc applied\r\n"); route(1); break; }; ....... Raymond