Jim,
I've been catching up to where you have already been with respect to failure route and the single global fr_inv_timer. It just doesn't work if the same SER is fielding invites for those having voicemail and those not. Before I go ahead and add a Wait 20 to my Asterisk voicemail, I had one other idea I wanted to ask about:
What if I run two ser instances? The main one checks is_user_in("Request-URI", "voicemail") and then punts any Invites for that user to a second ser which implements the fr_inv_timer, t_on_failure, etc. transfer to voicemail. Meanwhile all the non-voicemail subscribers (including stuff sent to our PSTN gateway, etc.) continues to chug along without the timer stuff happening.
Does this sound like a reasonable approach?
/a
Jim Burwell wrote:
I've found that the whole system works better when you don't use failure_route() to do "time out" style voicemail.
I have a test setup where I just add a call to Asterisk immediately if the user is registered (has location entry) with a special prefix that gets matched in the * extensions.conf file, and causes * to simply wait for X seconds before answering. If the calee phone(s) is answered, SER cancels the call to * and any other phones registered to the user. If no phones are answered, * answers and sends the call to voicemail.
For whatever reason, this approach seems more stable so far, and seems more compatible with more phones, etc, than the failure_route approach. For instance, if you have a locations entry that points a user to another user, or more than one user (e.g. mainline@domain.com -> receptionist@domain.com -> receptionist@phone-IP:port), SER seems to get confused and sends a CANCEL to the voicemail system you've just triggered the INVITE to in your failure_route. When doing it the other way, it doesn't seem to have this problem.
As for mapping user names to numeric extensions in Asterisk, I wrote an AGI script which does this by reverse-mapping the username to aliases in the SER script. It requires that your users have numerical extensions assigned in the SER aliases database (e.g. 1234 -> joe@domain.com). If there are multiple #s mapped to the user, it just returns the lowest one. Not sure if I can give this out though, since I wrote it for a client.
- Jim
Alan Crosswell wrote:
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=~"^si"^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=~"^si"^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 nowprefix("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"); }
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