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

17 May 2006


      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