1 Dec
2003
1 Dec
'03
5:23 p.m.
You missed the most interesting part of the network dump. Because after
receiving the 401 from Ser kphone should send the REIGSTER again, this time
with Authentication header included. If kphone does not send such a request,
then start your kphone from a terminal and watch at the debug output, maybe
it gives you a hint.
Regards
Nils
On Monday 01 December 2003 14:38, Jorge Merlino wrote:
Hello:
I'm trying to register kphone to my ser proxy using authentication with
user and password but I can't make it work.
In the identity configuration of kphone I have set "jorge" in "user part
of
SIP URL" and "authentication username" and I have set "asterix"
in "host
part of SIP URL" and "outbound proxy"
The same configuration with user and password works fine with windows
messenger 4.7
---------------------------------------------------------------------------
------------------------------
The error messages I get from kphone are these:
SipClient: Sending: 10:09:30.743
--------------------------------
REGISTER sip:asterix SIP/2.0
Via: SIP/2.0/UDP 192.168.2.175:5062
CSeq: 928 REGISTER
To: "jorge" <sip:jorge@asterix>
Expires: 900
From: "jorge" <sip:jorge@asterix>
Call-ID: 1234450217(a)192.168.2.175
Content-Length: 0
User-Agent: KPhone/3.12
Event: registration
Allow-Events: presence
Contact: "jorge"
<sip:jorge@192.168.2.175:5062;transport=udp>;q=1.0;methods="INVITE,
MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Receiving message...
SipClient: Received: 10:09:30.755
---------------------------------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.175:5062
CSeq: 928 REGISTER
To: "jorge"
<sip:jorge@asterix>;tag=b27e1a1d33761e85846fc98f5f3a7e58.fbe3
From: "jorge" <sip:jorge@asterix>
Call-ID: 1234450217(a)192.168.2.175
WWW-Authenticate: Digest realm="asterix",
nonce="3fcb14af87f1bd1110526a8d29ad258492f165bd"
Server: Sip EXpress router (0.8.11 (i386/linux))
Content-Length: 0
---------------------------------------------------------------------------
------------------------------
And my ser.cfg file is this:
alias="asterix"
alias=192.168.2.176
sip_warning=no
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
fifo="/tmp/ser_fifo"
# Uncomment this if you want to use SQL database
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"
modparam("usrloc", "db_mode", 2)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("rr", "enable_full_lr", 1)
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (len_gt( max_len )) {
sl_send_reply("513", "Message too big");
break;
};
record_route();
if (loose_route()) {
t_relay();
break;
};
if (uri==myself) {
if (uri =~ "^sip:9[0-9]*@asterix") {
log ("enviando al asterisk\n");
t_relay_to_udp("192.168.2.175", "5060");
break;
};
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("asterix", "subscriber")) {
www_challenge("asterix", "0");
break;
};
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
if (!t_relay()) {
sl_reply_error();
};
}
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