Hello,

could you send us the logs (send it to serhelp@lists.iptel.org if it is bigger than approximately 40kB) ? Also could you create SIP message dumps using ngrep or ethereal ?

We need to know how is the INVITE processed in the script (which branches of the script are executed). You can also try to use append_hf and add different header fields to SIP messages in different parts of the config file (it is similar to log(".."), but the text can be seen in SIP messages sent by SER).

Jan.

On 03-12 11:47, Rick Gocher wrote:

Hi All,

I am new to SER and have only recently installed and ran the application. I have a cisco gateway which I forward to for call termination and an ATA 186 to use as my sip client. I feel the problem is with my dial plan all I ever get is fast busy signals. When I set debug to 9 I get lots of stuff in /var/log/messages although nothing about the call. I had stripped everything out of the ser.cfg routing leaving just a simple dial plan for any digits, this did not work either.

In the beginning I was able to call out however the joy was short lived as I have mucked up the config so bad I can't seem to find my way back.

Any help would be most appreciated Thank you

Rick

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

debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E)

check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo"

# # $Id: pstn.cfg,v 1.2 2003/06/03 03:18:12 jiri Exp $ # #

# ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/group.so" modparam("auth_db", "db_url","sql://ser:passwd@localhost/ser") modparam("usrloc", "db_url", "sql://ser:passwd@localhost/ser")

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

modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) # -- acc params -- modparam("acc", "log_level", 1) # that is the flag for which we will account -- don't forget to # set the same one :-) # modparam("acc", "log_flag", 1 )

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

# main routing logic

route{

   /* ********* ROUTINE CHECKS  ********************************** */

   # filter too old messages
   if (!mf_process_maxfwd_header("10")) {
           log("LOG: Too many hops\n");
           sl_send_reply("483","Too Many Hops");
           break;
   };
            if (msg:len >=  max_len ) {
            sl_send_reply("513", "Message too big");
            break;
    };
   /* ********* RR ********************************** */

   /* grant Route routing if route headers present */
   if (loose_route()) { t_relay(); break; };

   /* record-route INVITEs -- all subsequent requests must visit us */
   if (method=="INVITE") {
           record_route();
   };

# now check if it really is a PSTN destination which should be handled # by our gateway; if not, and the request is an invitation, drop it -- # we cannot terminate it in PSTN; relay non-INVITE requests -- it may # be for example BYEs sent by gateway to call originator if (!uri=~"sip:+?[0-9]+@.*") { if (method=="INVITE") { sl_send_reply("403", "Call cannot be served here"); } else { # forward(uri:host, uri:port); forward(192.168.1.101, 5060); #ip of my cisco gateway }; break; };

   # account completed transactions via syslog
   setflag(1);

   # free call destinations ... no authentication needed
   if ( is_user_in("Request-URI", "free-pstn")  /* free destinations */
                   | uri=~"sip:[79][0-9][0-9][0-9]@.*" /* local PBX */
                   | uri=~"sip:98[0-9][0-9][0-9][0-9]") {
            log("free call");

   } else if (src_ip==192.168.1.101) {
           # our gateway doesn't support digest authentication;
           # verify that a request is coming from it by source
           # address
           log("gateway-originated request");
   } else {
           # in all other cases, we need to check the request against
           # access control lists; first of all, verify request
           # originator's identity

           if (!proxy_authorize(   "gateway" /* realm */,
                           "subscriber" /* table name */))  {
                   proxy_challenge( "gateway" /* realm */, "0" /* no 

qop */ ); break; };

           # authorize only for INVITEs -- RR/Contact may result in 
           weird
           # things showing up in d-uri that would break our logic; our
           # major concern is INVITE which causes PSTN costs

           if (method=="INVITE") {

                   # does the authenticated user have a permission for 

local # calls (destinations beginning with a single zero)? # (i.e., is he in the "local" group?) if (uri=~"sip:0[1-9][0-9]+@.*") { if (!is_user_in("credentials", "local")) { sl_send_reply("403", "No permission for local calls"); break; }; # the same for long-distance (destinations begin with two zeros") } else if (uri=~"sip:00[1-9][0-9]+@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", " no permission for LD "); break; }; # the same for international calls (three zeros) } else if (uri=~"sip:000[1-9][0-9]+@.*") { if (!is_user_in("credentials", "int")) { sl_send_reply("403", "International permissions needed"); break; }; # everything else (e.g., interplanetary calls) is denied } else { sl_send_reply("403", "Forbidden"); break; };

           }; # INVITE to authorized PSTN

   }; # authorized PSTN

   # if you have passed through all the checks, let your call go to GW!

   rewritehostport("192.168.1.101:5060");

   # forward the request now
   if (!t_relay()) {
           sl_reply_error();
           break;
   };

if (uri=~"^sip:[0-9]*@.*") { log("Forwarding to PSTN\n"); t_relay_to_udp ("192.168.1.101","5060"); # IP address of my cisco gateway break; }; }

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