[Serusers] call busy forward failed

[Charles Wang]

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 at 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;
       };
}