[Serusers] Fast Busy Signals

[Rick Gocher]

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]+ at .*") {
                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]+ at .*") {
                                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]+ at .*") {
                                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]+ at .*") {
                                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;
                 };
  }