New subject: [Serusers] failure route and ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487

19 Dec 2003

I'm trying to do failure route to voicemail (which is working) but this 
error is logged:

ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487

I googled this and see that this came up last in October 
(http://lists.iptel.org/pipermail/serusers/2003-October/002921.html) but 
I don't see any evidence of a solution in the thread.

I suspect mine is the same problem as I have two UAs registered so the 
initial dset is two places and then the append_branch on timeout is just 
the voicemail uri.

BTW, I am doing this with asterisk for the usual DTMF access reasons 
(altough I haven't yet figured out how to map sip:alan@columbia.edu to 
DTMF.  Maybe I'll have to give up and do numeric mailboxes:-(
I am rewriting with a prefix of "vm*u" before punting over to asterisk 
which has
exten => _vm*u.,1,Wait,1
exten => _vm*u.,2,Voicemail(${EXTEN:3})	
exten => _vm*u.,3,Goto(#,1)

Any pointers would be appreciated.  My not-yet-complete ser.cfg is 
attached (not yet punting voicemail for unregistered subscribers; just 
registered subscribers who time out the invite).

/a

#
# $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $
#
# simple quick-start config script
#

# ----------- global configuration parameters ------------------------

debug=2         # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no	# (cmd line: -E)
#listen=128.59.39.127

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"
alias="columbia.edu"
# ------------------ module loading ----------------------------------

# Uncomment this if you want to use SQL database
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/acc.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"

# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"

loadmodule "/usr/lib/ser/modules/exec.so"

# ----------------- setting module-specific parameters ---------------
# -- transaction timers --
modparam("tm", "fr_inv_timer", 15 )
modparam("tm", "fr_timer", 10 )

# -- usrloc params --
#
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "timer_interval", 10)

# -- auth params --
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")

# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)

# -- acc params --
modparam("acc", "log_level", 1)
modparam("acc", "log_flag", 1 )
modparam("acc", "log_missed_flag", 2)
# -------------------------  request routing logic -------------------

# main routing logic

route{

        /* ********* ROUTINE CHECKS  ********************************** */
	# 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;
	};

	lookup("aliases");

	# 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
	record_route();	

	# loose-route processing
	if (loose_route()) {
		t_relay();
		break;
	};

	setflag(2);

	# 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=~"^sip:(.+@)?columbia.edu") {

		if (method=="REGISTER") {
			log(1, "REGISTER received\n");
			if (!www_authorize("columbia.edu", "subscriber")) {
				www_challenge("columbia.edu", "0");
				break;
			};

			save("location");
			break;
		}; 

/* ********** Dial out to PSTN logic ************* */ 

                # 5 Digit dialing, interior calls
                if (uri=~"^sip:[1347][0-9]{4}@columbia\.edu") {
                        rewritehostport("128.59.59.242:5060");
                        log(1,"5 digit expression match");
                        route(2);
                        break;
                };

		# 10 Digit dialing with outlide line
		if (uri=~"^sip:931[0-9]{10}@columbia\.edu") {
	        	if(!(src_ip=="128.59.59.242") 
			& !(proxy_authorize("columbia.edu","subscriber"))) {
			proxy_challenge("columbia.edu", "1");
			break;

		} else {
                        rewritehostport("128.59.59.242:5060");
                        log(1," 93 Outside line with 10 digit expression
match");
                        route(2);
                        break;
		};
                };

/* voicemail access */
		if (uri=~"^sip:\*86@columbia\.edu" 
		    |uri=~"^sip:vm@columbia\.edu" 
		    |uri=~"^sip:voicemail@columbia\.edu") {
			route(3);
			break;
		};

		# native SIP destinations are handled using our USRLOC DB
		if (!lookup("location")) {
			if (!exec_dset("/etc/ser/sipldap")) {
				sl_send_reply("404", "Not Found");
				break;
			} else {
	                        log(1," sipldap call");
			};
		}; #!lookup

	}; 

	if (method == "INVITE") {
	        t_on_failure("1");
	};
	# forward to current uri now; use stateful forwarding; that
	# works reliably even if we forward from TCP to UDP
	if (!t_relay()) {
		sl_reply_error();
	}; 

}

route[2] {       
        log(1,"route[2]:SIP-to-PSTN call routed");
        if (!t_relay()) {
                sl_reply_error();
        };
}
# ---- voicemail user access ----
route[3] {
	rewritehostport("127.0.0.1:5069");
	log(1,"voicemail access");
        if (!t_relay()) {
                sl_reply_error();
        };
}
# ------------- handling of unregistered user ------------------
route[4] {

	log(1,"route[4]: user not registered");
        # non-Voip -- just send "off-line"
        if (!(method == "INVITE" || method == "ACK" || method ==
"CANCEL")) {
                sl_send_reply("404", "Not Found");
                break;
        };

        # not voicemail subscriber
#        if (!isflagset(4)) { 
#                sl_send_reply("404", "Not Found and no voicemail turned
on");
#               break;
#       };

        # forward to voicemail now
	prefix("vm*u");
        rewritehostport("127.0.0.1:5069");
        t_relay_to_udp("127.0.0.1", "5069");
}
failure_route[1] {
	# transfer to asterisk voicemail with uMAILBOX for unavailable.
	# sip:USER@columbia.edu -> sip:vm*uUSER@127.0.0.1:5069
        t_on_failure("2");
	prefix("vm*u");
        rewritehostport("127.0.0.1:5069");
	append_branch();
	log(1,"redirection to voicemail\n");
	t_relay();
}

failure_route[2] {
	# forwarding failed (voicemail down?)
	log(1,"voicemail failed\n");
	t_reply("500","Weasels have eaten voicemail again");
}