[OpenSIPS-Users] OpenSIPS is not running, Erorr

[Khan Friend]

Hi guys,

I am trying to troubleshoot errors in my OpenSIPS config file but unable to
understand what am i doing wrong.

The log file shows as follows:
Dec 26 21:38:02 [22302] INFO:usrloc:ul_init_locks: locks array size 512
Dec 26 21:38:02 [22302] INFO:registrar:mod_init: initializing...
Dec 26 21:38:02 [22302] INFO:textops:mod_init: initializing...
Dec 26 21:38:02 [22302] INFO:avpops:avpops_init: initializing...
Dec 26 21:38:02 [22302] INFO:auth:mod_init: initializing...
Dec 26 21:38:02 [22302] INFO:auth_db:mod_init: initializing...
Dec 26 21:38:02 [22302] INFO:core:probe_max_receive_buffer: using a UDP
receive buffer of 214 kb
Dec 26 21:38:56 [22303] ERROR:core:forward_reply: no 2nd via found in reply
Dec 26 21:38:57 [22308] ERROR:core:forward_reply: no 2nd via found in reply
Dec 26 21:38:58 [22306] ERROR:core:forward_reply: no 2nd via found in reply
Dec 26 21:38:59 [22304] ERROR:core:forward_reply: no 2nd via found in reply
Dec 26 21:39:00 [22303] ERROR:core:forward_reply: no 2nd via found in reply
Dec 26 21:39:10 [22308] ERROR:core:forward_reply: no 2nd via found in reply
Dec 26 21:39:11 [22306] ERROR:core:forward_reply: no 2nd via found in reply
D

-- 


My opensips.cfg is as follows:

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");
        exit;
    };

    if (msg:len >=  2048 ) {
        sl_send_reply("513", "Message too big");
        exit;
    };

    # 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);
    };

    if (!uri==myself) {
        # mark routing logic in request
        append_hf("P-hint: outbound\r\n");
        route(1);
    };

    # 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") {
            if (!www_authorize("", "subscriber")) {
                www_challenge("", "0");
                exit;
            };

            save("location");
            exit;
        };

        # requests for Media server
        if(is_method("INVITE") && !has_totag() && uri=~"sip:\*9") {
            route(3);
            exit;
        }

        # mark transaction if user is in voicemail group
        if(is_method("INVITE") && !has_totag()
            && is_user_in("Request-URI","voicemail"))
        {
            xdbg("user [$ru] has voicemail redirection enabled\n");
            # backup R-URI
            avp_pushto("$ru","$avp(i:10)");
            #avp_write("$ruri","$avp(i:10)");
            setflag(2);
        };
        # native SIP destinations are handled using our USRLOC DB
        if (!lookup("location")) {
            if(isflagset(2)) {
                # route to Asterisk Media Server
                prefix("1");
                rewritehostport("192.168.1.11:5060");
                route(1);
            } else {
                sl_send_reply("404", "Not Found");
                exit;
            }
        };
        append_hf("P-hint: usrloc applied\r\n");
    };

    route(1);
}


route[1] {

    if(isflagset(2))
        t_on_failure("1");

    if (!t_relay()) {
        sl_reply_error();
    };
    exit;
}


# voicemail access
# - *98 - listen caller's voice messages, being prompted for pin
# - *981 - listen voice messages, being promted for mailbox and pin
# - *98XXXX - leave voice message to XXXX
#
route[3] {
      # direct voicemail
    if (uri =~ "sip:\*98@" ) {
            rewriteuser("1");
        xdbg("voicemail access\n");
    } else if (uri =~ "sip:\*981@" ) {
         strip(4);
        rewriteuser("11");
    } else if (uri =~ "sip:\*98.+@" ) {
         strip(3);
        prefix("1");
    } else {
        xlog("unknown media extension $rU\n");
        sl_send_reply("404", "Unknown media service");
        exit;
    }

    # route to Asterisk Media Server
    rewritehostport("192.168.1.11:5060");
    route(1);
}

failure_route[1] {
    if (t_was_cancelled()) {
        xdbg("transaction was cancelled by UAC\n");
        return;
    }
    # restore initial uri
    avp_pushto("$ru","$avp(i:10)");
    #avp_pushto("$ru", "i:10");
    prefix("1");
    # route to Asterisk Media Server
    rewritehostport("192.168.1.11:5060");
    resetflag(2);
    route(1);
}


Thank you,


Khan
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