[Serusers] Radius Accounting doesn't work with SER 0.9.0

Rafael J. Risco G.V. rafael.risco at gmail.com
Wed Mar 2 00:15:44 CET 2005


hello
I´ve just upgraded SER from version 0.8.99-dev1 to 0.9.0 and my radius
accounting system doesn't work properly, it only generates the start
log but I can´t see any "Stop record" in radius logs when call its
finished... see my configuration file below (same that I used with
0.8.99-dev1).

thanks 
RaFael


File: /usr/local/etc/ser/Ser_VM_RadAcc_NatHelp-Test1.cfg

# This default script includes nathelper support. To make it work
# you will also have to install Maxim's RTP proxy. The proxy is enforced
# if one of the parties is behind a NAT.
# 
# If you have an endpoing in the public internet which is known to
# support symmetric RTP (Cisco PSTN gateway or voicemail, for example),
# then you don't have to force RTP proxy. If you don't want to enforce
# RTP proxy for some destinations than simply use t_relay() instead of
# route(1)
#
# Sections marked with !! Nathelper contain modifications for nathelper
#
# Also Include Mysql support for digest authentication, Pstn forward, 
# voicemail and Radius Accounting module.

# ----------- global configuration parameters ------------------------

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

listen=****
port=5060

# hostname matching an alias will satisfy the condition uri==myself".
alias=domain.com

check_via=no    # (cmd. line: -v)
dns=no           # (cmd. line: -r)
rev_dns=no      # (cmd. line: -R)
children=4
fifo="/tmp/ser_fifo"

# sip_warning - Should replies include extensive warnings? 
# By default yes, it is good for trouble-shooting.
sip_warning=yes


# ------------------ module loading ----------------------------------
loadmodule "/usr/local/lib/ser/modules/mysql.so"
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/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/group.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so" 
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"

# digest authentication
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"

# !! Nathelper
loadmodule "/usr/local/lib/ser/modules/nathelper.so"

# ----------------- setting module-specific parameters ---------------

modparam("usrloc", "db_mode",   2)

# storing passwords in our database in plain text:
# modparam("auth_db", "calculate_ha1", yes)
# modparam("auth_db", "password_column", "password")

# For Rad Accounting
modparam("acc","radius_config","/usr/local/etc/radiusclient/radiusclient.conf")
modparam("acc", "service_type", 15)
modparam("acc", "radius_flag", 1)
modparam("acc", "radius_missed_flag", 3)
modparam("acc", "report_ack", 0) # 1 reporta dos starts en acc

modparam("tm", "fr_timer", 20 )
modparam("tm", "fr_inv_timer", 30 )
modparam("tm", "wt_timer", 20 )

# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)

modparam("group", "db_url", "mysql://ser:heslo@localhost/ser")  #
mysql in cvs head vs
# modparam("uri", "db_url", "sql://ser:heslo@localhost/ser") # in ser0814
modparam("uri_db", "db_url", "mysql://ser:heslo@localhost/ser") # in
cvs head version

# ------------- registration parameters
modparam("registrar", "nat_flag", 6)
modparam("registrar", "min_expires", 60)
modparam("registrar", "max_expires", 86400)
modparam("registrar", "default_expires", 3600)
modparam("registrar", "desc_time_order", 1)
modparam("registrar", "append_branches", 1)

# !! Nathelper
# modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1)   # Ping only clients behind NAT

# -------------------------- request routing logic --------------------------

route {

        log(1, "-------------------------------------------\n");
        log(1, "entering main loop\n");


        # 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;
        };

        # !! Nathelper
        # Special handling for NATed clients; first, NAT test is
        # executed: it looks for via!=received and RFC1918 addresses
        # in Contact (may fail if line-folding is used); also,
        # the received test should, if completed, should check all
        # vias for rpesence of received
        if (nat_uac_test("19")) {
                # Allow RR-ed requests, as these may indicate that
                # a NAT-enabled proxy takes care of it; unless it is
                # a REGISTER

                if (method == "REGISTER" || ! search("^Record-Route:")) {
                    log("LOG: Someone trying to register from private
IP, rewriting\n");

                    # This will work only for user agents that support symmetric
                    # communication. We tested quite many of them and
majority is
                    # smart enough to be symmetric. In some phones it
takes a configuration
                    # option. With Cisco 7960, it is called
NAT_Enable=Yes, with kphone it is
                    # called "symmetric media" and "symmetric signalling".

                    fix_nated_contact(); # Rewrite contact with source
IP of signalling
                    if (method == "INVITE") {
                        fix_nated_sdp("1"); # Add direction=active to SDP
                    };
                    force_rport(); # Add rport parameter to topmost Via
                    setflag(6);    # Mark as NATed
                };
        };


        # 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"); 
                # t_relay();
                route(1);  # Nathelper!!
                break;
        };


        # Set flag for Radius Accounting:
                
                if (method=="INVITE") {
                log(1, "INVITE MESSAGE RECEIVED - START ACC\n");
                setflag(1); /* set for accounting (the same value as
in log_flag!) */
                };

                if (method=="BYE") {
                log (1, "BYE  - STOP ACCOUNTING\n");
                setflag(1);
                };

                if (method=="CANCEL") {
                log (1, "CANCEL - STOP ACCOUNTING\n");
                setflag(1);
                };

        setflag(3); # Set Radius Missed Flag            

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

        if (uri==myself) {

                if (method == "REGISTER") {
                        log(1, "ANALYZING REGISTER REQUEST\n");
                        # to use digest authentication
                        if (!www_authorize("domain.com", "subscriber")) {
                                www_challenge("domain.com", "0");
                                break;
                        };
                        if (!save("location")) {
                                sl_reply_error();
                        };
                        break;
                };

                /* ***************** Dial out to PSTN logic
****************** */
                ### Pendiente agregar seguridad a esta etapa, usar
Digest-Auth o "credentials"

                # forward n digit requests to gateway AS5350 (Celulares Lima)
                if(uri=~"^sip:9"){
                        log(1,"n digit expression match - Celulares");
                        rewritehostport("X.X.X.X:5060");
                        route(2);
                        break;    
                };

                # forward international calls to Asterisk (a FWD, H323gws)
                if(uri=~"^sip:00"){
                        rewritehostport("Y.Y.Y.Y:5060");
                        log(1,"n digit expression match - LDI");
                        route(2);
                        break;
                };

 
                /*
********************************************************************
*/
        
                lookup("aliases");
                if (!uri==myself) {
                        append_hf("P-hint: outbound alias\r\n"); 
                        # t_relay();
                        route(1);
                        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")) {
                        log(1, "requested user is in voicemail group");
                        setflag(4);
                };

                # native SIP destinations are handled using our USRLOC DB
                if (!lookup("location")) {
                        log(1,"unable to locate user");
                        # handle user which was not found
                        route(4);
                        break;
                };

        }; # End of "if(uri==myself)"

        append_hf("P-hint: usrloc applied\r\n"); 
        route(1);

        # if user is on-line and is in Voicemail group, enable redirection
        if (method == "INVITE" && isflagset(4)) {
                log(1, "invite for voicemail user->initiate failureroute[1]\n");
                t_on_failure("1");
        };
        
        # t_relay();
}

route[1] 
{
        # !! Nathelper
        if (uri=~"[@:](192\.168\.)" && !search("^Route:")){
            sl_send_reply("479", "We don't forward to private IP addresses");
            break;
        };

        # if client or server know to be behind a NAT, enable relay
        if (isflagset(6)) {
            force_rtp_proxy();
        };

        # NAT processing of replies; apply to all transactions (for example,
        # re-INVITEs from public to private UA are hard to identify as
        # NATed at the moment of request processing); look at replies
        t_on_reply("1");

        # send it out now; use stateful forwarding as it works reliably
        # even for UDP2TCP
        if (!t_relay()) {
                sl_reply_error();
        };
        break;
}

# !! Nathelper
onreply_route[1] {
    # NATed transaction ?
    if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
        fix_nated_contact();
        force_rtp_proxy();
    # otherwise, is it a transaction behind a NAT and we did not
    # know at time of request processing ? (RFC1918 contacts)
    } else if (nat_uac_test("1")) {
        fix_nated_contact();
    };
}



# ----------------- SIP-to-PSTN call routed -------------------

route[2]{       
        log(1,"route[2]:SIP-to-GW call routed");
        if(!t_relay()){
                sl_reply_error();
        };
}

# --------------- Handling of Unavailable user ----------------
route[4] {
 
        # non-Voip -- just send "off-line"
        if (!(method=="INVITE" || method=="ACK" || method=="CANCEL" ||
method=="BYE")) {
                sl_send_reply("404", "Not Found");
                acc_rad_request("404 Not Found");
                break;
        };

        # not voicemail subscriber
        if (!isflagset(4)) { 
                sl_send_reply("404", "Not Found and no voicemail turned on");
                acc_rad_request("404 Not Found");
                break;
        };

        # forward to voicemail adding prefix to simplify * "extension.conf"
        prefix("vm");  
        rewritehostport("Y.Y.Y.Y:5060");
        t_relay_to_udp("Y.Y.Y.Y", "5060");
}

# if forwarding downstream did not succeed, try voicemail running at Asterisk 

failure_route[1]{
        if (t_check_status("485")){
                revert_uri ();
                prefix("vm");
                rewritehostport ("Y.Y.Y.Y:5060");
                append_branch();
                t_relay();
                break;
        }
}




-- 

rrgv




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