Hello list. I have a problem-question regarding to the BYE message reaching SER. For the normail example i have a endpoint registered in SER and a PSTN-Gateway receiving the outgoing traffic. For this enviroment when i made a call from my endpoint to the PSTN gateway and the Gateway send the BYE (at the end of the call) this is what i see in the debug in SER.
U gw1.mydomain.com:52154 -> sersip.mydomain.com:5060 BYE sip:005622408196@sersip.mydomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP gw1.mydomain.com:5060 From: sip:005622408196@sersip.mydomain.com;tag=A0272B49-B2D To: sip:5555832351@sersip.mydomain.com;tag=ae4208f2a4 Date: Mon, 25 Oct 2004 21:34:02 GMT Call-ID: ae229c42-f2c3-08f3-81f2-0002a400f1e9@xx.xx.xx.42 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 6 Route: sip:5555832351@xx.xx.xx.42:5060 Timestamp: 1098740047 CSeq: 101 BYE Content-Length: 0
# U sersip.mydomain.com:5060 -> xx.xx.xx.42:5060 BYE sip:5555832351@xx.xx.xx.42:5060 SIP/2.0 Record-Route: sip:005622408196@sersip.mydomain.com;ftag=A0272B49-B2D;lr=on Via: SIP/2.0/UDP sersip.mydomain.com;branch=z9hG4bK4c65.37369754.0 Via: SIP/2.0/UDP gw1.mydomain.com:5060 From: sip:005622408196@64.76.148.231;tag=A0272B49-B2D To: sip:5555832351@sersip.mydomain.com;tag=ae4208f2a4 Date: Mon, 25 Oct 2004 21:34:02 GMT Call-ID: ae229c42-f2c3-08f3-81f2-0002a400f1e9@xx.xx.xx.42 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 5 Timestamp: 1098740047 CSeq:101 BYE Content-Length: 0
# U xx.xx.xx.42:5060 -> sersip.mydomain.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP sersip.mydomain.com;branch=z9hG4bK4c65.37369754.0 Via: SIP/2.0/UDP gw1.mydomain.com:5060 From: sip:005622408196@sersip.mydomain.com;tag=A0272B49-B2D To: sip:5555832351@sersip.mydomain.com;tag=ae4208f2a4 Call-ID: ae229c42-f2c3-08f3-81f2-0002a400f1e9@xx.xx.xx.42 CSeq: 101 BYE User-Agent: AddPac SIP Gateway Content-Length: 0 Record-Route: sip:005622408196@sersip.mydomain.com;ftag=A0272B49-B2D;lr=on
# U sersip.mydomain.com:5060 -> gw1.mydomain.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP gw1.mydomain.com:5060 From: sip:005622408196@sersip.mydomain.com;tag=A0272B49-B2D To: sip:5555832351@sersip.mydomain.com;tag=ae4208f2a4 Call-ID: ae229c42-f2c3-08f3-81f2-0002a400f1e9@xx.xx.xx.42 CSeq: 101 BYE User-Agent: AddPac SIP Gateway Content-Length: 0 Record-Route: sip:005622408196@sersip.mydomain.com;ftag=A0272B49-B2D;lr=on
As you can see the call is ended succesfully and there is no problem. Now, instead of having a PSTN-Gateway i use a SIP-H323 translator and all the calls are going out through this device. When i made a call from my endpoint and the SIP-H323 send the BYE message (at the end of the call) the debug shows this :
U sipquest.mydomain.com:5060 -> sersip.mydomain.com:5060 BYE sip:1112001003@sersip.mydomain.com;ftag=25421eefa4;lr=on SIP/2.0 Via: SIP/2.0/UDP sipquest.mydomain.com:5060; branch=z9hG4bKsKpiQfXRAwAYAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAB_ From: sip:1112001003@sersip.mydomain.com;tag=3896741524 To: sip:5555832351@sersip.mydomain.com;tag=25421eefa4 CSeq: 1 BYE Call-ID: 25229c42-16c9-1e95-81ef-0002a400f1e9@xx.xx.xx.42 Contact: sip:1112001003@sipquest.mydomain.com Route: sip:5555832351@sersip.mydomain.com Max-Forwards: 69 Content-Length: 0
# U sersip.mydomain.com:5060 -> sipquest.mydomain.com:5060 SIP/2.0 404 Not Found IT! Via: SIP/2.0/UDP sipquest.mydomain.com:5060; branch=z9hG4bKsKpiQfXRAwAYAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAB_;received=sipquest .mydomain.com From: sip:1112001003@sersip.mydomain.com;tag=3896741524 To: sip:5555832351@sersip.mydomain.com;tag=25421eefa4 CSeq: 1 BYE Call-ID: 25229c42-16c9-1e95-81ef-0002a400f1e9@xx.xx.xx.42 Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 sersip.mydomain.com:5060 "Noisy feedback tells: pid=15285 req_src_ip=sipquest.mydomain.com req_src_port=5060in_uri=sip:1112001003@sersip.mydomain.com;ftag=25421eefa4;l r=on out_uri=sip:1112001003@sersip.mydomain.com;ftag=25421eefa4;lr=on via_cnt==1"
My question is : Why the call is ended succesfully in the first case and in the second case SER answer with a "NOT FOUND"? I only have registered in my location table the endpoints. Here is my ser.cfg file.
# # $Id: ser.cfg,v 1.21.2.2 2003/10/13 22:53:06 jiri Exp $ # # simple quick-start config script #
# ----------- global configuration parameters ------------------------
debug=0 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E)
#Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes
listen=xx.xx.148.231
alias=sersip.mydomain.com
check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo" fifo_mode=0666
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database #loadmodule "/usr/local//lib/ser/modules/mysql.so"
loadmodule "/usr/local//lib/ser/modules/sl.so" loadmodule "/usr/local//lib/ser/modules/tm.so" loadmodule "/usr/local//lib/ser/modules/rr.so" loadmodule "/usr/local//lib/ser/modules/maxfwd.so" loadmodule "/usr/local//lib/ser/modules/usrloc.so" loadmodule "/usr/local//lib/ser/modules/registrar.so" loadmodule "/usr/local//lib/ser/modules/textops.so"
loadmodule "/usr/local//lib/ser/modules/exec.so"
# Para trabajar con NAT loadmodule "/usr/local//lib/ser/modules/mediaproxy.so" loadmodule "/usr/local//lib/ser/modules/domain.so" loadmodule "/usr/local//lib/ser/modules/dbtext.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" loadmodule "/usr/local/lib/ser/modules/acc.so"
# ----------------- setting module-specific parameters ---------------
# ----------------- setting module-specific parameters --------------- # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1)
# module authorization modparam("auth_radius","radius_config","/usr/local/etc/radiusclient/radiuscl ient.conf") modparam("auth_radius","service_type",15)
#module accounting modparam("acc","radius_config","/usr/local/etc/radiusclient/radiusclient.con f") modparam("acc","log_level",1) modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 3) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0)
#group radius modparam("group_radius","radius_config","/usr/local/etc/radiusclient/radiusc lient.conf") modparam("group_radius", "use_domain", 1)
#module mediaproxy modparam("mediaproxy", "natping_interval", 60) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") #modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/ser/sip-asymmetrics-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-asymmetrics-clients") modparam("registrar", "nat_flag", 5)
#module domain modparam("domain", "db_url", "/usr/local/etc/ser/domaintables") modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain")
# ------------------------- request routing logic -------------------
# main routing logic
route { # Llamada a Rutina Externa para cambio de numero # if (method=="INVITE") { # exec_dset("/usr/local/etc/ser/redirect.sh"); # };
# initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; };
# subsequent messages withing a dialog should take the # path determined by record-routing record_route();
if (loose_route()) { if (method=="BYE" || method=="CANCEL") { log(1, "NAT: BYE o CANCEL recibido --> terminando la sesion de media\n"); end_media_session(); setflag(1); }; t_relay(); break; };
# Maracamos las llamadas para accounting setflag(1);
if (client_nat_test("1")) { log(1, "NAT: Requerimiento de IP privada --> fixed contact (en rutina principal)\n"); setflag(5); force_rport(); fix_contact(); append_hf("P-hint: fixed NAT contact for request\r\n"); };
# (in case, it does not work, use the following command # with proper names and addresses in it)
if (uri==myself) { if (method=="REGISTER") { # Challenge/Response if ( !radius_www_authorize("")) { www_challenge("", "1"); break; };
# Mandamos un mensaje de Log si es que un cliente NATEADO se registro. if (isflagset(5)) { log(1, "NAT: Cliente NAT'eado registrado (dentro del chequeo de REGISTER)\n"); };
save("location"); break; }; # FIN DE CHEQUEO DE REGISTRO
# lookup("aliases");
# CHEQUEO EL INVITE DE LA LLAMADA if (method=="INVITE" && search("^From: sip:.*@xx.xx.148.231")) { if ( !radius_proxy_authorize("")) { proxy_challenge("", "1"); log (1, "LOG: Llamada no AUTORIZADA\n"); break; }; };
if ( (uri=~"^sip:0.*@.*") ){ if (method=="INVITE") { log(1,"FORWARDING : Llamada hacia plataforma H.323\n"); route(3); break; }; }; # Llamadas hacia dominio H.323 if ( (uri=~"^sip:111.*@.*") | (uri=~"^sip:777.*@.*") | (uri=~"^sip:333.*@.*") ){ if (method=="INVITE") { log(1,"FORWARDING : Llamada hacia plataforma H.323\n"); route(1); break; }; };
# Llamadas SIP locales if (!lookup("location")) { #Usuarios que no estan en "location" database. route(4); break; };
setflag(3);
# Manejo para llamadas NAT'das if (method=="INVITE") { log(1, "ROUTING: Reply processing (online user) enabled to handle NAT'd resonses\n"); t_on_reply("1"); if ( (isflagset(5)) && (method=="INVITE" || method=="ACK") ) { log(1, "NAT: Invite received --> enabling media proxy (else del lookup 2do if)\n"); use_media_proxy(); append_hf("P-hint: request forced to media proxy\r\n"); }; };
# Do it ? (Ricardo Martinez) }; append_hf("P-hint: USRLOC\r\n");
if (!t_relay()) { sl_reply_error(); break; };
} /* end of initial routing logic */
#--------- Route 1 hacia SIPQUEST ------------------------------- route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP
# Esta NATEADO? if (isflagset(5)) { log(1, "NAT: At least one participant NAT'd --> enable reply processing (hacia GW)\n"); t_on_reply("1"); if (method=="INVITE") { log(1, "NAT: Invite received --> enabling proxied media session (hacia GW)\n"); setflag(1); use_media_proxy(); }; };
rewritehostport("sipquest.mydomain.com:5060"); append_hf("P-hint: GATEWAY\r\n");
if (!t_relay()) { sl_reply_error(); break; };
}
#------- Route 3 : Trafico de Internet hacia la PSTN -------------------
route[3] { setflag(1);
#Esta NETEADO?
if (isflagset(5)) { log(1, "NAT: At least one participant NAT'd --> enable reply processing (hacia GW)\n"); t_on_reply("1"); if (method=="INVITE") { log(1, "NAT: Invite received --> enabling proxied media session (hacia GW)\n"); #setflag(1); use_media_proxy(); }; };
rewritehostport("gw1.mydomain.com:5060"); append_hf("P-hint: GATEWAY\r\n");
t_relay(); break;
}
#------ Route 4 : Llamadas a usuarios Off-line ------------------------
route[4] {
if (!t_newtran()) { sl_reply_error(); };
if (!t_reply("404", "Not Found IT!")) { sl_reply_error(); }; break; }
# ---------------------------- Begin On-Reply Routes --------------------------
onreply_route[1] { # If we've got here, it's because we've previously passed through a block # which handles NAT requests and has set a t_on_reply condition. DB 03-08-2004 if (status =~ "(183)|(2[0-9][0-9])") { if (client_nat_test("1") || isflagset(5)) { log(1, "NAT: Reply from NAT'd client --> fixing contact (onreply_route)\n"); fix_contact(); use_media_proxy(); setflag(5); }; log(1, "NAT: NAT'd transaction answered --> enabling media proxy (onreply_route)\n"); # use_media_proxy(); };
}
# ------------------------------ End On-Reply Routes --------------------------
Can someone please help me?.
Ricardo Martinez