[Serusers] problem getting calls to my gateway
Rick Gocher
rgocher at coptalk.com
Thu Dec 4 18:31:54 CET 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 at localhost/ser")
modparam("usrloc", "db_url", "sql//secret at 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]+ at .*") {
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]+ 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=~"sip00[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=~"sip000[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("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 at 64.189.165.205;user=phone SIP/2.0..Via
SIP/2.0/UDP 64.189.165.2065060..From <sip6
044848235 at 64.189.165.205;user=phone>;tag=409936633..To
<sip776044445556 at 64.189.165.205;user=phone>..Call-ID
2945885252 at 64.189.165.206..CSeq 1 INVITE..Contact
<sip6044445555 at 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 at 64.189.165.205;user=phone>;tag=409936633..To
<sip776044445556 at 64.189.165.205;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.0ed0..Call-ID
2945885252 at 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 at 64.189.165.205;user=phone
out_uri=sip776044445556 at 64.189.165.205;user=phone via_cnt==1"....
#
U 64.189.165.2065060 -> 64.189.165.2055060
ACK sip776044445556 at 64.189.165.205;user=phone SIP/2.0..Via SIP/2.0/UDP
64.189.165.2065060..From <sip6044
445555 at 64.189.165.205;user=phone>;tag=409936633..To
<sip776044445556 at 64.189.165.205;user=phone>;tag=b27e1a1
d33761e85846fc98f5f3a7e58.0ed0..Call-ID 2945885252 at 64.189.165.206..CSeq
1 ACK..User-Agent Cisco ATA 186
v2.16.2 ata18x (030909a)..Content-Length 0....
More information about the sr-users
mailing list