Dear ALL: I try to make a call from UA1 to UA2, and UA2 is busy. So the call will forward to another UA3. UA1 ==> UA2(busy) ==> UA3 But when I watch the log, I find that miss the following log (diff with call from UA1 to UA2 and directly forward to UA3) UA1 ==> UA2(callfwd) ==> UA3 Feb 22 12:15:07 ser mediaproxy[18476]: session 16AA8C75-5EAC-403C-85DA-C5D9BDFD15C7(a)10.18.1.70: started. listening on xxx.xxx.190.248:35026 When I make a callfwd call, the call will run route[2] but not failure_route[1]. UA1==>UA2(callfwd)==>route[2]==>UA3 And I make a fwdbusy call, the call will run route[2] then pass failure_route[1], and return to route[2]. UA1==>UA2(fwdbusy)==>failure_route[1]==>route[2] =XXX=> UA3 ???? Why does the method be failed? Do I must "end_media_session()" before start a busy call? How can I modify it? My snippet ser.cfg : -------------------------------------------------------------------------------------------------------------- route[2] { log(1, "SER: SIP Call On-Net section route(2)\n"); if ((method=="INVITE") && !allow_trusted()) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; } else if (!check_from()) { log(1, "Spoofed SIP call attempt"); sl_send_reply("403", "Use From=ID"); break; } else if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Please register to use our service"); break; }; }; if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; t_on_failure("1"); t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { log(1, "SER: Failure Route section failure_route(1)\n"); # if caller hung up then don't sent to voicemail if (t_check_status("487")) { break; }; if (isflagset(26) && t_check_status("486")) { # forward busy is flag 26 if (avp_pushto("$ruri", "s:fwdbusy")) { log(1, "SER: fork to fwdbusy\n"); avp_delete("s:fwdbusy"); append_branch(); resetflag(26); # test for domestic PSTN gateway if (uri=~"^sip:0[0-9]{9}@") { # if (avp_check("$fwd_busy_type", "eq/dom/i")) { # test for domestic PSTN gateway log(1, "SER: Busy Failure and Jump to route(3)\n"); route(3); } else if (uri=~"^sip:002[1-9][0-9]*@") { # } else if (avp_check("$fwd_busy_type", "eq/int/i")) { # test for international PSTN gateway log(1, "SER: Busy Failure and Jump to route(6)\n"); route(6); } else { # default to sip call log(1, "SER: Busy Failure and Jump to route(2)\n"); route(2); }; break; }; }; # here we can have either voicemail __OR__ forward no answer if (isflagset(27) && t_check_status("408")) { # forward no answer is flag 27 if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { avp_pushto("$ruri", "s:fwdnoanswer"); log(1, "SER: fork to fwdnoanswer\n"); avp_delete("s:fwdnoanswer"); append_branch(); resetflag(27); if (uri=~"^sip:0[0-9]{9}@") { # if (avp_check("$fwd_no_answer_type", "eq/dom/i")) { # test for domestic PSTN gateway log(1, "SER: No Answer Failure and Jump to route(3)\n"); route(3); } else if (uri=~"^sip:002[1-9][0-9]*@") { # } else if (avp_check("$fwd_no_answer_type", "eq/int/i")) { # test for international PSTN gateway log(1, "SER: No Answer Failure and Jump to route(6)\n"); route(6); } else { # default to sip call log(1, "SER: No Answer Failure and Jump to route(2)\n"); route(2); }; break; }; } else if (isflagset(31) && avp_pushto("$ruri", "$voicemail")) { avp_delete("$voicemail"); log(1, "SER: No Answer Failure and Jump to route(4)\n"); route(4); break; }; }