IF call is for ME (my SIP domain) THEN {
    IF callee is NOT online THEN {
        IF callee is in voicemail group THEN {
           call VM system immediately with URI modified signaling
              "answer immediately"
           break
        } ELSE (callee is NOT in voicemail group) {
           return 404 (user not found)
           break
        }
    } ELSE (callee is online) {
        IF callee is in voicemail group THEN {
           append destination set including VM system with URI
              modified signaling "wait before answering"
        }
    }
}

relay call to dest set

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"^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"^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 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");
}


 

------------------------------------------------------------------------

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-- 
+---------------------------------------------------------------------------+
|         Jim Burwell - Sr. Systems/Network/Security Engineer, JSBC         |
+---------------------------------------------------------------------------+
| "I never let my schooling get in the way of my education." - Mark Twain   |
| "UNIX was never designed to keep people from doing stupid things, because |
|  that policy would also keep them from doing clever things." - Doug Gwyn  |
| "Cool is only three letters away from Fool" - Mike Muir, Suicyco          |
| "..Government in its best state is but a necessary evil; in its worst     |
|  state an intolerable one.." - Thomas Paine, "Common Sense" (1776)        |
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+---------------------------------------------------------------------------+

-- 
+---------------------------------------------------------------------------+
|         Jim Burwell - Sr. Systems/Network/Security Engineer, JSBC         |
+---------------------------------------------------------------------------+
| "I never let my schooling get in the way of my education." - Mark Twain   |
| "UNIX was never designed to keep people from doing stupid things, because |
|  that policy would also keep them from doing clever things." - Doug Gwyn  |
| "Cool is only three letters away from Fool" - Mike Muir, Suicyco          |
| "..Government in its best state is but a necessary evil; in its worst     |
|  state an intolerable one.." - Thomas Paine, "Common Sense" (1776)        |
+---------------------------------------------------------------------------+
|   Email:  jimb@jsbc.cc                              ICQ UIN:  1695089     |
+---------------------------------------------------------------------------+
|  Reply problems ?  Turn off the "sign" function in email prog.  Blame MS. |
+---------------------------------------------------------------------------+