[Serusers] Call transfer, ser not send BYE after REFER

Hello, i have problem with call transfer. I try A call to B, B receive call, B transfer A to B truncate ngrep port 5060: INVITE B From: A To: B REFER A From: B To: A Refer-To: C Referred-by: B INVITE C From: A To: C Problem: ser not send BYE after REFER My ser.cfg: debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (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 #mhomed=yes listen=195.137.182.11 # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/group.so" #loadmodule "SER_MODULES_DIR/acc.so" loadmodule "/usr/lib/ser/modules/acc_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix", "@GW_IP_3>;party=calling;id-type=subscriber;screen=yes;privacy=off") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # Acc modparam("acc", "report_ack", 1) modparam("acc", "log_level", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) modparam("acc", "log_fmt", "fisum") # Nemelo by se to objevit v logu #modparam("acc", "log_flag", FLAG_ACC) #modparam("acc", "log_missed_flag", FLAG_MISSED) # Nastaveni spojeni k Mysql pro moduly modparam("group|usrloc|uri_db|acc", "db_url", "mysql://ser:ser@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ # 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; }; # Do acc se zapisuji vsechny nasledujici pozadavky if (method == "INVITE" || method == "CANCEL" || method == "BYE" || method == "ACK") { setflag(2); }; # Uprava hlavicky Server remove_hf("Server"); append_hf("Server: EMEA Telecom IBM S2 SIP Server\r\n"); # !! Nathelper # Special handling for NATed clients; first, NAT test is executed: it looks for via!=received and RFC1918 addresses in Contact (may fail if line-folding is used); also, the received test should, if completed, should check all vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!(method=="REGISTER")) record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; # account all BYEs if (method=="BYE") setflag(2); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Kontrola prihlaseni podle tabulky subscribler if (!www_authorize("sip.ahoj.cz", "subscriber")) { www_challenge("sip.ahoj.cz", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if (does_uri_exist()) { # Existuje na serveru ale neni online sl_send_reply("404", "Not Found"); break; } else { # neni na tomto serveru, pujde do pstn, nebude se pouzivat rtpproxy resetflag(6); setflag(1); }; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper 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 client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); append_hf("X-rtpproxy: yes\r\n"); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP route(2); } route[2] { if ((method == "INVITE") && (uri =~ "sip:S302_")) { sl_send_reply("302", "Rewrited"); break; }; if (method == "INVITE") { if (src_ip == "xxx.xxx.xxx.xxx") { setflag(4); append_hf("X-s2router-source: gateway-pstn\r\n"); log("LOG: From PSTN\n"); }; if (!is_user_in("Request-URI", "free-pstn")) { if (!isflagset(4) && !proxy_authorize("DIGEST_REALM", "subscriber")) { proxy_challenge("DIGEST_REALM", "0"); break; }; # let's check from=id ... avoids accounting confusion if (!isflagset(4) && !check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (gw)"); break; }; }; }; # Pokud neni cislo registrovane na serveru zkus pstn if (isflagset(1)) { # Vola jen na cisla if (uri =~ "^[a-zA-Z]+:[0-9]+@") { route(4); } else { sl_send_reply("604", "Does Not Exist Anywhere"); }; break; }; route(3); } route[4] { log(1, "Jedu pres PSTN"); if (!(method == "INVITE" || method == "ACK" || method == "CANCEL" || method == "OPTIONS" || method == "BYE")) { sl_send_reply("500", "only VoIP methods accepted for GW"); break; }; rewritehostport("195.122.201.61:5060"); # Acc setflag(2); route(3); } route[3] { if (!t_relay()) { sl_reply_error(); }; break; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } Any idea? Thank -- Murdej Ukrutny