Dear Greger:
I have no voicemail service in my scenario. So I keep the voicemail flag and I setting it as "n" in usr_preferences table.
And where to add these code of your support? Please show me a example. Thank you very much.
Best Regard Charles.
On Thu, 24 Feb 2005 07:53:06 +0100, Greger V. Teigre greger@teigre.com wrote:
Without following all your logic below, a quick question: You wrap the setting of t_on_failure based on flag 31. Is that the only situation where you want failure route to be called?
if (isflagset(31)) { # is voice mail? t_on_failure("1"); };
g-)
Charles Wang wrote:
Dear ALL:
I use Paul's ser.cfg for 0.9.0 post last year and wanna implement a call busy forward. I think Paul is busy on testing 'Best Practice', so he doesn't answer too much mail.
I try to make a call from UA1 to UA2, and UA2 is busy. So the call will forward to another Trunk 3. Ex: UA1 ==> UA2(busy) ==(failure_route[1])==> Trunk 3
PS: UA1 and UA2 behind different NATs.
But when I watch the log, I find that miss the following log (diff with call from UA1 to UA2 and directly forward to Trunk 3 Ex: UA1 ==> UA2(callfwd) ==> Trunk 3
When I make a callfwd call, the call will run route[3] but not failure_route[1]. UA1==>UA2(callfwd)==>route[2]==>Trunk 3
route[2] ====> SIP to SIP call route[3] ====> SIP to PSTN call failure_route[1] ====> NoAnswer or Busy Forward logic
And I make a fwdbusy call, the call will run route[2] then go to failure_route[1], and return to route[3]. UA1==>UA2(fwdbusy)==>failure_route[1]==>route[3] =XXX=> Trunk 3 ????
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[3] { log(1, "SER: International Call Off-Net section route(3)\n");
# All Domestic Calls Go To CISCO if (method=="INVITE") { 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; }; # enable caller id blocking for PSTN calls if (isflagset(25)) { append_rpid_hf(); }; }; # SIP->PSTN calls get 45 seconds to timeout log(1, "SER: Connecting to PSTN.....\n"); avp_write("i:45", "inv_timeout");
rewritehost("xxx.xxx.190.243"); #### A Trunking Gateway
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(); }; if (isflagset(31)) { # is voice mail? 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 toroute(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 Failureand 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; }; }
In SysLog:
proxydispatcher[13265]: command request 06DC9503-1938-4D9D-8628-E648F112BEC5@192.168.11.7 192.168.11.7:8000:audio 61.217.225.225 ser.xxx.net.tw local xxx.xxx.190.243 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:9012@ser.xxx.net.tw,fromtag:2069977709,totag: Feb 24 00:59:21 ser proxydispatcher[13265]: domain ser.xxx.net.tw doesn't define any mediaproxy. Feb 24 00:59:21 ser proxydispatcher[13265]: will use default mediaproxy for this call. Feb 24 00:59:21 ser mediaproxy[1087]: command request 06DC9503-1938-4D9D-8628-E648F112BEC5@192.168.11.7 192.168.11.7:8000:audio 61.217.225.225 ser.xxx.net.tw local xxx.xxx.190.243 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:9012@ser.xxx.net.tw,fromtag:2069977709,totag:,dispatcher Feb 24 00:59:21 ser mediaproxy[1087]: command execution time: 0.35 ms Feb 24 00:59:21 ser proxydispatcher[13265]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: 'xxx.xxx.190.248 35030' Feb 24 00:59:21 ser proxydispatcher[13265]: command execution time: 311.69 ms
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