--- gunawan <gun_cool182(a)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(a)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.
1. 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.
2. 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.
3. 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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