Girish I have just followed your advise and it works perfect now, ser accounting module reports "Sip-Response-Code=487" and no "invite" goes to asterisk again, ...thank you very much for your invaluable help .

Rafael

On Wed, 10 Nov 2004 04:58:15 -0800 (PST), Girish gr_sh2003@yahoo.com wrote:

Rafael,

As i said in my previous mail, it just was a guess. Can you try modifying your route[1] something like this. I think this will send only the unanswered calls to voicemail. All other 4** status will be ignored. I havent tested it though.

Also, you were right. It is route[1], not route[4].

route[1] {

if (t_check_status("408")){ revert_uri(); rewritehostport("200.110.2.132:5060"); append_branch(); t_relay (); break; }

}

Good Luck,

--- "Rafael J. Risco G.V." rafael.risco@gmail.com wrote:

...

hello Girish I did some test using t_check_status ("487") in failure_route[1] like this:

failure_route[1] { if (t_check_status("487")){ revert_uri (); rewritehostport ("200.110.2.132:5060"); append_branch(); t_relay_to_udp("200.110.2.132", "5060"); }

...but did not work, can you send us more details about how to use t_check_status("487") in route[4]?

thanks rafael

On Tue, 9 Nov 2004 02:25:26 -0800 (PST), Girish gr_sh2003@yahoo.com wrote:

...

Hello,

Just a guess.. Can you add t_check_status (487) in your route[4] and see if the INVITE goes to Asterisk after cancelling?

--- "Rafael J. Risco G.V." rafael.risco@gmail.com wrote:

...

Hello I have to insist in this issue since I´ve done several test using Sems and asterisk with very simple configuration files including the original example from ser-cvs... in brief: if I call to a user who belongs to voicemail group and I cancel the call before VM forward routine begin then an "invite" is sent to a voicemail server generating and sending a file with No audio, and I cant account this call with "Sip-Response-Code=487" (just an start record without stop)...

does someone know how to solve this problem????

thanks in advance

Rafael

PS: ser.cfg and asterisk debug for this test:

# # SER SIMPLE CFG for VM without acc... # ----------- global configuration parameters ------------------------

#debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E)

#/* Uncomment these lines to enter debugging mode debug=9 fork=yes log_stderror=yes #*/

listen=127.0.0.1 port=5060

# simple proxy script for forwarding to voicemail server # for unavailable users #

loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so"

# time to give up on ringing -- global timer, applies to # all transactions modparam("tm", "fr_inv_timer", 90)

# database with user group membership modparam("group", "db_url", "mysql://ser:heslo@localhost/ser")

# --------------------- request routing logic ------------------- route {

    if (!mf_process_maxfwd_header("10")) {
            log("LOG: Too many hops\n");
            sl_send_reply("483", "Alas Too Many Hops");
            break;
    };

    if (!(method=="REGISTER")) record_route();
    if (loose_route()) {
            t_relay();
            break;
    };

    if (!uri==myself) {
            t_relay();
            break;
    };

    if (method == "REGISTER") {
            if (!save("location")) {
                    sl_reply_error();
            };
            break;
    };

    # does the user wish redirection on no availability? (i.e., is he
    # in the voicemail group?) -- determine it now and store it in
    # flag 4, before we rewrite the flag using UsrLoc
    if (is_user_in("Request-URI", "voicemail")) {
            setflag(4);
    };

    # native SIP destinations are handled using our USRLOC DB
    if (!lookup("location")) {
            # handle user which was not found
            route(4);
            break;
    };

    # if user is on-line and is in voicemail group, enable redirection
    if (method == "INVITE" && isflagset(4)) {
            t_on_failure("1");
    };
    t_relay();

}

# ------------- handling of unavailable user ------------------ route[4] {

    # 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
    rewritehostport("200.110.2.132:5060");
    t_relay_to_udp("200.110.2.132", "5060");

}

# if forwarding downstream did not succeed, try voicemail running # at 200.110.2.132:5060

failure_route[1] { revert_uri(); rewritehostport("200.110.2.132:5060"); append_branch(); t_relay_to_udp("200.110.2.132", "5060"); }

===== Girish Gopinath gr_sh2003@yahoo.com

---

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