[Serusers] Re: Radius Accounting doesn't work with SER 0.9.0 ---SOLVED

Hello I solved this problem moving "set flag for Radius Accounting" section before "record-route all messages", hope it helps...

regards Rafael Risco

On Tue, 1 Mar 2005 18:15:44 -0500, Rafael J. Risco G.V. rafael.risco@gmail.com wrote:

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