this is it:

#
# OpenSIPS residential configuration script
#     by OpenSIPS Solutions <[email protected]>
#
# This script was generated via "make menuconfig", from
#   the "Residential" scenario.
# You can enable / disable more features / functionalities by
#   re-generating the scenario with different options.#
#
# Please refer to the Core CookBook at:
#      http://www.opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#


####### Global Parameters #########

log_level=3
log_stderror=no
log_facility=LOG_LOCAL0

children=4

/* uncomment the following lines to enable debugging */
#debug_mode=yes

/* uncomment the next line to enable the auto temporary blacklisting of
   not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns
   lookup failures (default disabled) */
#dns_try_ipv6=yes

/* comment the next line to enable the auto discovery of local aliases
   based on reverse DNS on IPs */
auto_aliases=no


listen=udp:192.168.16.6   # CUSTOMIZE ME
listen=udp:111.111.111.111   # CUSTOMIZE ME



####### Modules Section ########

#set module path
mpath="/usr/lib64/opensips/modules"

#### SIGNALING module
loadmodule "signaling.so"

#### StateLess module
loadmodule "sl.so"

#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)

#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)

#### MAX ForWarD module
loadmodule "maxfwd.so"

#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"

#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)

#### URI module
loadmodule "uri.so"
modparam("uri", "use_uri_table", 0)

#### MYSQL module
loadmodule "db_mysql.so"

#### HTTPD module
loadmodule "httpd.so"
modparam("httpd", "port", 8888)

#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "db_mode",   2)
modparam("usrloc", "db_url",
    "mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME


#### REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)

#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure the enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)
modparam("acc", "db_url",
    "mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("acc", "extra_fields", "db: src_ip; dst_ip")
modparam("acc", "leg_fields", "db: caller; callee")

#### AUTHentication modules
loadmodule "auth.so"
loadmodule "auth_db.so"
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db|uri", "db_url",
    "mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("auth_db", "load_credentials", "")

#### ALIAS module
loadmodule "alias_db.so"
modparam("alias_db", "db_url",
    "mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME

#### DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600)  # 6 hours timeout
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_url",
    "mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME

####  DIALPLAN module
loadmodule "dialplan.so"
modparam("dialplan", "db_url",
    "mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME

####  DYNAMIC ROUTING module
loadmodule "drouting.so"
modparam("drouting", "db_url",
    "mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("drouting", "use_domain", 0)


####  MI_HTTP module
loadmodule "mi_http.so"
loadmodule "mi_json.so"
loadmodule "proto_udp.so"

####  permissions
loadmodule "permissions.so"
modparam("permissions", "db_url",
    "mysql://opensips:opensipsrw@localhost/opensips")

#### call_center
loadmodule "b2b_entities.so"
loadmodule "b2b_logic.so"
modparam("b2b_logic", "script_scenario", "/etc/opensips/scenario_callcenter.xml")

loadmodule "call_center.so"
modparam("call_center", "db_url",
    "mysql://opensips:opensipsrw@localhost/opensips")
modparam("call_center", "acc_db_url",
    "mysql://opensips:opensipsrw@localhost/opensips")


####### Routing Logic ########

# main request routing logic

route{

    if (!mf_process_maxfwd_header("10")) {
        send_reply("483","Too Many Hops");
        exit;
    }

    if (has_totag()) {

        # handle hop-by-hop ACK (no routing required)
        if ( is_method("ACK") && t_check_trans() ) {
            t_relay();
            exit;
        }

        # sequential request within a dialog should
        # take the path determined by record-routing
        if ( !loose_route() ) {
            # we do record-routing for all our traffic, so we should not
            # receive any sequential requests without Route hdr.
            send_reply("404","Not here");
            exit;
        }

        # validate the sequential request against dialog
        if ( $DLG_status!=NULL && !validate_dialog() ) {
            xlog("In-Dialog $rm from $si (callid=$ci) is not valid according to dialog\n");
            ## exit;
        }

        if (is_method("BYE")) {
            # do accounting even if the transaction fails
            ##do_accounting("db","failed"); #no need for cdr

        }

        # route it out to whatever destination was set by loose_route()
        # in $du (destination URI).
        route(relay);
        exit;
    }

    # CANCEL processing
    if (is_method("CANCEL")) {
        if (t_check_trans())
            t_relay();
        exit;
    }

    # absorb retransmissions, but do not create transaction
    t_check_trans();

    if ( !(is_method("REGISTER")  ) ) {
        if (is_myself("$si") && is_myself("$rd")) {
            xlog("-- it looks like call from call_center module --");
            setflag(call_center_internal);
        } else {

            if (is_myself("$fd")) {

                # authenticate if from local subscriber
                # authenticate all initial non-REGISTER request that pretend to be                 # generated by local subscriber (domain from FROM URI is local)
                if (!proxy_authorize("", "subscriber")) {
                    proxy_challenge("", "0");
                    exit;
                }
                if (!db_check_from()) {
                    send_reply("403","Forbidden auth ID");
                    exit;
                }

                consume_credentials();
                # caller authenticated

            } else {
                # if caller is not local, then called number must be local

                if (!is_myself("$rd")) {
                    xlog("-- F");
                    send_reply("403","Relay Forbidden");
                    exit;
                }

                # if caller is not local, then check ip - address

                if(!check_source_address("0")){
                    send_reply("403","Forbidden for You");
                    exit;
                }


            }
        }

    }

    xlog("-- V1 --[$rm / $fu / $tu / $ru / $ci]");

    # preloaded route checking
    if (loose_route()) {
        xlog("L_ERR",
            "Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
        if (!is_method("ACK"))
            send_reply("403","Preload Route denied");
        exit;
    }

    # record routing
    if (!is_method("REGISTER|MESSAGE"))
        record_route();

    # account only INVITEs
    if (is_method("INVITE")) {

        # create dialog with timeout
        if ( !create_dialog("B") ) {
            send_reply("500","Internal Server Error");
            exit;
        }

        do_accounting("db", "cdr");

    }

    if (!is_myself("$rd")) {
        append_hf("P-hint: outbound\r\n");

        route(relay);
    }

    # requests for my domain

    if (is_method("PUBLISH|SUBSCRIBE")) {
        send_reply("503", "Service Unavailable");
        exit;
    }

    if (is_method("REGISTER")) {
        # authenticate the REGISTER requests
        if (!www_authorize("", "subscriber")) {
            www_challenge("", "0");
            exit;
        }

        if (!db_check_to()) {
            send_reply("403","Forbidden auth ID");
            exit;
        }
        if (!save("location"))
            sl_reply_error();

        exit;
    }

    if ($rU==NULL) {
        # request with no Username in RURI
        send_reply("484","Address Incomplete");
        exit;
    }

    $acc_extra(src_ip) = $si; # source IP of the request
    $acc_leg(caller) = $fu;
    $acc_leg(callee) = $ru;


    # apply DB based aliases
    alias_db_lookup("dbaliases");


    #Dial plan processing
    dp_translate("0","$rU/$rU", "$avp(attrs)");
        xlog("rU: $rU , avp(attrs): $avp(attrs)");

    if ($avp(attrs)=="pstn") {
        #route to pstn
        route(pstn);
    }

    if ($avp(attrs)=="media") {
        #route to media server
        route(media);
    }

    if ($avp(attrs)=="ivr") {
        #route to ivr
        route(ivr);
    }

    #all other calls are local ($avp(attrs)=="usrloc")
    #Route to usrloc
    route(lookup);

    send_reply("420", "Invalid Extension V2");
    exit;

    #######
}


route[relay] {
    # for INVITEs enable some additional helper routes
    if (is_method("INVITE")) {

        t_on_branch("per_branch_ops");
        t_on_reply("handle_nat");
        t_on_failure("missed_call");
    }


    if (!t_relay()) {
        send_reply("500","Internal Error");
    }
    exit;
}

route[lookup] {
    # do lookup with method filtering
    if (!lookup("location", "m")) {
        switch ($retcode) {
            case -1:
            case -3:
                t_newtran();
                t_reply("404", "Not Found");
                exit;
            case -2:
                sl_send_reply("405", "Method Not Allowed");
                exit;
        }
    }

    do_accounting("db","missed");

    serialize_branches(1);
    next_branches();

    route(relay);
    exit;
}

route[pstn] {
        #---- PSTN route ----#
    if(!do_routing("0","1")){
        send_reply("503", "No rules found matching the URI prefix");
            exit;
    }
    # flag 10 - route to pstn
    setflag(10);
    route(relay);
}

route[media] {
    #---- Route to media servers ----#
    xlog("route to media servers");
}

route[ivr] {
        #---- IVR route ----#
        xlog("route to IVR");

    if (!cc_handle_call("prq_test1")) {
        xlog("route to IVR failed, retcode: $retcode");
        send_reply("403","Cannot handle call");
        exit;

    }


    exit;

}

branch_route[per_branch_ops] {
    xlog("new branch at $ru\n");
}


onreply_route[handle_nat] {

    xlog("incoming reply\n");

    if ( $rs >= 200 )
        $acc_extra(dst_ip) = $si;
}


failure_route[missed_call] {
    do_accounting("db","missed");

    if(isflagset(10)){
        if (use_next_gw()) {
            xlog ("next gateway $ru \n");
                t_relay();
                   exit;
        } else {
            t_reply("503", "Service not available, no more gateways");
                   exit;
        }
    }

    if (t_was_cancelled()) {
        exit;
    }

    next_branches();
    # if we've got any more branches arm the failure route
    if ($rc != 2) {
        t_on_failure("missed_call");
    }

    if (!t_relay()) {
        send_reply("500","Internal Error");
    };

    if ( $avp(redirect_uri)!=NULL ) {
        # set the new destination
        $ru = $avp(redirect_uri);

        # create a new call leg
        acc_new_leg();
        # new caller is the callee of the previous leg
        $acc_leg(caller) = $(acc_leg(callee)[-2]);
        # new callee is the new destination
        $acc_leg(callee) = $ru;

        t_on_failure("missed_call");
        route(lookup);
        exit;
    }

    # uncomment the following lines if you want to block client
    # redirect based on 3xx replies.
    ##if (t_check_status("3[0-9][0-9]")) {
    ##t_reply("404","Not found");
    ##    exit;
    ##}


}



local_route {
    if (is_method("BYE") && $DLG_dir=="UPSTREAM") {

        acc_db_request("200 Dialog Timeout", "acc");

    }
}


Mikhail Laba

20.04.2019 19:47, David Villasmil пишет:
You’re not pasting the config scenario

On Sat, 20 Apr 2019 at 16:30, Mikhail <[email protected] <mailto:[email protected]>> wrote:

    Hi Alex,

    my config file is here: http://video2dv.com/download/opensips.cfg

    Mikhail Laba



    20.04.2019 15:30, Alexander Jankowsky пишет:
    > Hello Mikhail,
    >
    > Can you copy and show us your opensips config file.
    > Maybe it is something simple and very basic that someone will
    quickly recognise.
    >
    > Alex
    >
    > -----Original Message-----
    > From: Users [mailto:[email protected]
    <mailto:[email protected]>] On Behalf Of Mikhail
    > Sent: Saturday, 20 April 2019 8:17 PM
    > To: [email protected] <mailto:[email protected]>
    > Subject: [OpenSIPS-Users] call_center module issues
    >
    > Hello!
    >
    > I'm trying to setup call_center module for the first time and
    found the following problems:
    >
    >
    > #1. When a call is in the queue and waiting for the free agent,
    opensips
    > makes a call to a media server and caller hear music - that's is ok,
    >
    > but when a free agent found and opensips is calling to him, the
    call to
    > mediaserver is ended and caller hear nothing.
    >
    > If agent did not answer, opensips sets up a new call to mediaserver.
    >
    > This is very strange behavior, because normally the caller,
    while he is
    > in queue, should hear music without interrupt  until one of the
    agents
    > answer the call. Exactly answer, but not ringing.
    >
    >
    > #2. After the agent answer the call it's not possible to put call on
    > hold, both agent and caller can't do it. In sip dialog there is
    a "400
    > Not Acceptable"
    >
    >
    > Of course it's possible that I made a wrong logic in opensips
    config,
    > anyway Is this behavior was hardcoded in call_center module? How
    to fix it?
    >
    >
    > Mikhail Laba
    >
    >
    >
    > _______________________________________________
    > Users mailing list
    > [email protected] <mailto:[email protected]>
    > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
    >
    >
    > _______________________________________________
    > Users mailing list
    > [email protected] <mailto:[email protected]>
    > http://lists.opensips.org/cgi-bin/mailman/listinfo/users

    _______________________________________________
    Users mailing list
    [email protected] <mailto:[email protected]>
    http://lists.opensips.org/cgi-bin/mailman/listinfo/users

--
Regards,

David Villasmil
email: [email protected] <mailto:[email protected]>
phone: +34669448337

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