Re: [Users] Using # for Sip 2 Sip calls

So I took your advice and decided to use * to identify sip 2 sip calls. However, theres something wrong with my routing. I added route(6) to get authorize. Because when I try to dial sip to sip I get 407 proxy authentication required. Still after adding route(6), I still get the 407 proxy authentication required message. What is wrong? Route (1) is just the default message handler This is what I have: route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (uri=~"^sip:\*[0-9]*@"){ xlog("Sip 2 Sip\n"); strip(1); #strip the * because we dont need it route(4); route(6); route(1); return; }; if (!lookup("location")){ if (uri=~"^sip:[0-9]*@") { # International PSTN xlog("PSTN Gateway\n"); route(4); route(5); return; }; sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- xlog("Routed to route 5\n"); rewritehostport("pstn.gateway:5060"); avp_write("i:45", "inv_timeout"); route(1); } route[6] { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: Hi, that's right. For example SIPURA ATAs with two lines but online one terminal use # for line selection.... you better use a digit that does not overlap with the PSTN dialling plan. regards, bogdan Glenn Dalgliesh wrote: I came up with something like this: lookup("aliases"); if (uri=~"^sip:#[0-9]*@"){ xlog("Sip 2 SIP\n"); route(4); route(1); return; }; Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you. --------------------------------- Sneak preview the all-new Yahoo.com. It's not radically different. Just radically better.