[Serusers] problem getting calls to my gateway - sr-users

4 Dec 2003


      Hi everyone,
I have been having lots of problems delivering calls to my companies 
gateway.  I have tried several different configs found online however 
nothing seems to work.  I am including the ser.cfg and a dump from ngrep in 
hopes a kind person will see what the problem is.  I do not know very much 
about sip or ser so any help is greatly needed.  Currently our company uses 
a cisco voip solution and I am setting up Ser as a test.  Unfortunately our 
admin does not seem very helpful, I'm  not sure if he has things setup 
correctly for my calls on the gateway or if it's my ser.cfg file.  I was 
hoping from the information I'm sending someone can tell me where the 
problem looks like it's coming from, wether it's his gateway or my config 
file.  I'm thinking it's me as I don't see any attempt of passing the call 
to the gateway in the ngrep output.
Also, I have had to alter my ip's listed in this email.  I have been warned 
under penalty of pain not to broadcast their ip addresses...  :p  I hope 
this does not cause a problem.
ATA 64.189.165.206
Ser Box  64.189.165.205
Cisco GW 65.189.155.101
Thank you,
# ----------- global configuration parameters ------------------------
debug=3        # debug level (cmd line -dddddddddd)
fork=yes
log_stderror=no # (cmd line -E)
#/* Uncomment these lines to enter debugging mode
#fork=no
#log_stderror=yes
#*/
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 031812 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//secret@localhost/ser")
modparam("usrloc", "db_url", "sql//secret@localhost/ser")
# ----------------- setting module-specific parameters ---------------
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
# -- 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 (msglen >=  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(urihost, uriport);
                 };
                 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[7][0-9][0-9][0-9]@.*"  /* local PBX */
                         | uri=~"sip98[0-9][0-9][0-9][0-9]") {
                 log("free call");
         } else if (src_ip==65.189.155.101) {
                 # 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=~"sip0[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=~"sip00[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=~"sip000[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("65.189.155.1015060");
# forward the request now
         if (!t_relay()) {
                 sl_reply_error();
                 break;
         };
}
################ ngrep output#######################
#
U 64.189.165.2065060 -> 64.189.165.2055060
   INVITE sip776044445556@64.189.165.205;user=phone SIP/2.0..Via 
SIP/2.0/UDP 64.189.165.2065060..From <sip6
   044848235@64.189.165.205;user=phone>;tag=409936633..To 
<sip776044445556@64.189.165.205;user=phone>..Call-ID
    2945885252@64.189.165.206..CSeq 1 INVITE..Contact 
<sip6044445555@64.189.165.2065060;user=phone;transpor
   t=udp>..User-Agent Cisco ATA 186  v2.16.2 ata18x (030909a)..Expires 
300..Content-Length 257..Content-Typ
   e application/sdp....v=0..o=6044445555 62848 62848 IN IP4 
64.189.165.206..s=ATA186 Call..c=IN IP4 64.189.165.206..t=0 0..m=audio 
16384 RTP/AVP 18 8 0 101..a=rtpmap18 G729/8000/1..a=rtpmap8 
PCMA/8000/1..a=rtpmap0PCMU/8000/1..a=rtpmap101 
telephone-event/8000..a=fmtp101 0-15..
#
U 64.189.165.2055060 -> 64.189.165.2065060
   SIP/2.0 407 Proxy Authentication Required..Via SIP/2.0/UDP 
64.189.165.2065060..From 
<sip6044445555@64.189.165.205;user=phone>;tag=409936633..To 
<sip776044445556@64.189.165.205;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.0ed0..Call-ID 
2945885252@64.189.165.206..CSeq 1 INVITE..Proxy-Authenticate Digest 
realm="gateway", nonce="3fcf790810cb0daaf030be719aa79e574b96b535"..Server 
Sip EXpress router (0.8.12 (i386/linux)).
   .Content-Length 0..Warning 392 64.189.165.2055060 "Noisy feedback 
tells  pid=32407 req_src_ip=64.189.165.206 req_src_port=5060 
in_uri=sip776044445556@64.189.165.205;user=phone 
out_uri=sip776044445556@64.189.165.205;user=phone via_cnt==1"....
#
U 64.189.165.2065060 -> 64.189.165.2055060
   ACK sip776044445556@64.189.165.205;user=phone SIP/2.0..Via SIP/2.0/UDP 
64.189.165.2065060..From <sip6044
  445555@64.189.165.205;user=phone>;tag=409936633..To 
<sip776044445556@64.189.165.205;user=phone>;tag=b27e1a1
   d33761e85846fc98f5f3a7e58.0ed0..Call-ID 2945885252@64.189.165.206..CSeq 
1 ACK..User-Agent Cisco ATA 186
   v2.16.2 ata18x (030909a)..Content-Length 0....