[Serusers] error 404 user not found when call to PSTN - sr-users

6 Jul 2005

      --- gunawan gun_cool182@yahoo.com wrote:
Hi, All..
I used ser.cfg from getting started document 4a from
www.onsip.org
but I get error 404 user not found when I tried to
call to PSTN number ( I tried 0811xxxxxx)
is something I missed in my ser.cfg????/
the following is my ser.cfg :
debug=3
fork=yes
log_stderror=yes
listen=202.xx.xxx.xxx           # put your server IP
address here
port=5060
children=4
dns=no
rev_dns=no
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:heslo@localhost/ser"
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/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
loadmodule "/usr/local/lib/ser/modules/mediaproxy.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/avpops.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/permissions.so"
modparam("auth_db|permissions|uri_db|usrloc","db_url",
"mysql://ser:heslo@localhost/ser")
modparam("auth_db|uri_db|usrloc", "db_url",
"mysql://ser:heslo@localhost/ser")
modparam("auth_db", "calculate_ha1", 1)
modparam("auth_db", "password_column", "password")
modparam("nathelper", "rtpproxy_disable", 1)
modparam("nathelper", "natping_interval", 0)
modparam("mediaproxy","natping_interval", 30)
modparam("mediaproxy","mediaproxy_socket",
"/var/run/mediaproxy.sock")
modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-
clients")
modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-
clients")
modparam("usrloc", "db_mode", 2)
modparam("registrar", "nat_flag", 6)
modparam("rr", "enable_full_lr", 1)
modparam("tm", "fr_inv_timer", 27)
modparam("tm", "fr_inv_timer_avp", "inv_timeout")
modparam("permissions", "db_mode", 1)
modparam("permissions", "trusted_table", "trusted")
alias='pcr.ac.id'
route {
#
------------------------------------------------------------
-----
        # Sanity Check Section
        #
------------------------------------------------------------
-----
        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483", "Too Many Hops");
                break;
        };
if (msg:len > max_len) {
                sl_send_reply("513", "Message
Overflow");
                break;
        };
#
------------------------------------------------------------
-----
        # Record Route Section
        #
------------------------------------------------------------
-----
        if (method=="INVITE" && client_nat_test("3"))
{
                # INSERT YOUR IP ADDRESS HERE
record_route_preset("202.xx.xxx.xxx:5060;nat=yes");
        } else if (method!="REGISTER") {        
                record_route(); 
        };
#
------------------------------------------------------------
-----
        # Call Tear Down Section
        #
------------------------------------------------------------
-----
        if (method=="BYE" || method=="CANCEL") {
                end_media_session();
        };
#
------------------------------------------------------------
-----
        # Loose Route Section
        #
------------------------------------------------------------
-----
        if (loose_route()) {
if (has_totag() && (method=="INVITE"
|| method=="ACK"))
{
if (client_nat_test("3") ||
search
("^Route:.*;nat=yes")) {
                                setflag(6);
                                use_media_proxy();
                        };
                };
route(1);
                break;
        };
#
------------------------------------------------------------
-----
        # Call Type Processing Section
        #
------------------------------------------------------------
-----
if (uri!=myself) {
    	    route(5);
                route(1);
                break;
        };
if (uri==myself) {
if (method=="ACK") {
    		route(6);
    		break;}
else if (method=="CANCEL") {
                        route(3);
                        break;
                } else if (method=="INVITE") {
                        route(3);
                        break;
                } else  if (method=="REGISTER") {
                        route(2);
                        break;
                };
lookup("aliases");
                if (uri!=myself) {
                        route(5);
                        route(1);
                        break;
                };
if (!lookup("location")) {
                        sl_send_reply("404", "User Not
Found");
                        break;
                };
        };
route(1);
}
route[1] {
#
------------------------------------------------------------
-----
        # Default Message Handler
        #
------------------------------------------------------------
-----
t_on_reply("1");
if (!t_relay()) {
if (method=="INVITE" || method=="ACK")
{
                        end_media_session();
                };
sl_reply_error();
        };
}
route[2] {
#
------------------------------------------------------------
-----
        # REGISTER Message Handler
        #
------------------------------------------------------------
----
sl_send_reply("100", "Trying");
if (!search("^Contact:\ +*") &&
client_nat_test("7")) {
                setflag(6);
                fix_nated_register();
                force_rport();
        };
if (!www_authorize("pcr.ac.id","subscriber"))
{
                www_challenge("pcr.ac.id","0");
                break;
        };
if (!check_to()) {
                sl_send_reply("401", "Unauthorized");
                break;
        };
consume_credentials();
if (!save("location")) {
                sl_reply_error();
        };
}
route[3] {
#
------------------------------------------------------------
-----
        # CANCEL and INVITE Message Handler
        #
------------------------------------------------------------
-----
if (client_nat_test("3")) {
                setflag(7);
                force_rport();
                fix_nated_contact();
        };
if (method=="INVITE" && !allow_trusted()) {
if (!proxy_authorize("pcr.ac.id","subscriber")) {
 proxy_challenge("pcr.ac.id","0");
 break;
 } else if (!check_from()) {
 sl_send_reply("403", "Use From=ID");
 break;
 };
consume_credentials();
 };
if (uri=~"^sip:081[0-9]*@") {
 route(4);
 break;
 };
lookup("aliases");
        if (uri!=myself) {
    	    route(5);
                route(1);
                break;
        };
if (!lookup("location")) {
    	if (uri=~"^sip:[0-9]{10}@") {
    		route(4);
    		break;
    		};
                sl_send_reply("404", "User Not
Found");
                break;
        };
if (method=="CANCEL") {
                route(1);
                break;
        };
route(5);
        route(1);
}
route[4] {
#
-----------------------------------------------------------------
 # PSTN Handler
 #
-----------------------------------------------------------------
rewritehostport("202.xx.xxx.xxx:5060"); # INSERT YOUR
PSTN GATEWAY IP
ADDRESS
avp_write("i:45", "inv_timeout");
route(5);
 route(1);
 }
route[5] {
#
-----------------------------------------------------------------
 # RTP Proxy Enabler
 #
-----------------------------------------------------------------
if (isflagset(6) || isflagset(7)) {
 use_media_proxy();
 };
 }
route[6] {
#
---------------------------------------------------------------------
# ACK Handler
#
---------------------------------------------------------------------
#
---------------------------------------------------------------------
# Aliases Section
#
---------------------------------------------------------------------
    lookup("aliases");
    if (uri!=myself) {
    	route(5);
    	route(1);
    	break;
    };
lookup("location");
route(1);
}
onreply_route[1] {
if ((isflagset(6) || isflagset(7)) &&
(status=~"(180)|(183)|2
[0-9][0-9]")) {
if (!search("^Content-Length:\ +0")) {
                        use_media_proxy();
                };
        };
if (client_nat_test("1")) {
                fix_nated_contact();
        };
}
and I already configured The Cisco Router... when I
tried dial to Cisco gateway from PSTN.... for the
first
 few seconds I heard sound which like the router
answered the calls, after that I received busy
tone....
Why did I receive busy tone after the router answered
 my call??? Is my configuration in Router wrong????
 I already use debug ccsip messages, but nothing
 appeared in my console... ????? :?
could somebody help me... thanks all..... v(^_^)v
...
--- Steve Blair blairs@isc.upenn.edu wrote:
...
gunawan wrote:
...
Hi, Steve.....
I want to ask about dial-peers u provide in ur
website(http://mit.edu/sip/sip.edu/ciscoGW.html)...
...
Just for clarification this isn't my site. This is
where some of the 
documents associated
with the Internet2 working group of which I am a
member.
...

dial-peer voice 680010 voip
            description Only peer for inbound

to
...
SIP Proxy 215-746-8001:8009 extensions
               huntstop
               preference 2 
               destination-pattern 6800[1-9]
               progress_ind setup enable 3
               voice-class codec 1
               session protocol sipv2
               session target sip-server
               dtmf-relay rtp-nte
               no vad
what is 680010?????? is that ur phone number???? 
something related to phone number???
No it is not my phon enumber. This is an example
only. The value following
dial-peer voice is just a label to uniquely
identify
...
that dial-peer. The 
value
specified in the destination-pattern parameter is
what is matched 
against the
dialed digits. In this example the 6800[1-9] 
specifies a range of numbers
from 68001 through 68009. These are real Centrex
extensions within our
numbering plan bu tthey are test numbers reserved
for use by our VoIP 
project.
We use 5 digits because that is how many digits
our
...
Centrex provider hands
us in the call setup message.
...

dial-peer voice 61 pots

description Only peer for outbound 5-digit 746
campus calls
           translation-profile outgoing Prefix
                preference 3
                destination-pattern 6....
                direct-inward-dial
                port 1/0:23                  
                prefix 215746
why do u use 61???? something related to ur pone
number????
Same as above. This is an illustration. The 61 is
just a label . In this 
example
61 is the first dial-peer defined to handle
outbound
...
calls to Centrex 
extensions
6xxxx.
...

could u give me some other example

configuration,
...
bcoz I dun use PABX or analog router here.. I
plug
...
in
...
my Telephone line direct to VIC2FXO card...
so I Wish that my SIP client can call to PSTN
client...
my Telephone number that I plug to cisco router
is
...
...
62(761) 53808 , 62 is country code, 761 is area
kode,
...
53808 is my phone number...
The FXS and FXO peers are similar to those
mentioned
...
in the document. You
may need to fiddle with parameters for your
specific
...
type of connection.
You may need to know if you are using wink-start
or
...
loop-start signaling,
what port to use, etc. A very basic setup would
include something like:
voice-port 1/0/0
 input gain 10
 connection plar opx 32766
 description FXO 1/0/0 5732767 for VoIP
dial-peer voice 91 pots
 destination-pattern 9
 port 1/0/0
 prefix 9
!
...
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--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
   215-746-8001

fax: 215-898-9348
sip:blairs@upenn.edu
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