17 Oct
2005
17 Oct
'05
7:59 p.m.
Daniel-Constantin Mierla wrote:
That was exactly my first thought too. I use a noname SIP phone, and
than I used firefly. The exaclty same behaviour.
never used firefly.
I tried Xlite and it is exactly the same, ... I set the sip registration
port to 5062 and it still registers in Asterisk, ...
I tried to use instead of the domain name the IP address, but it is the
same.
I thought already on the possibility that the DNS SRV entry redirects
all to 5060, but that would not be anymore true, when I used the IP
address, ....
grep -v ^[#,$] openser.cfg
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5062
fifo="/tmp/openser_fifo"
fifo_mode=0666
loadmodule "/usr/local/lib/openser/modules/mysql.so"
loadmodule "/usr/local/lib/openser/modules/sl.so"
loadmodule "/usr/local/lib/openser/modules/tm.so"
loadmodule "/usr/local/lib/openser/modules/rr.so"
loadmodule "/usr/local/lib/openser/modules/maxfwd.so"
loadmodule "/usr/local/lib/openser/modules/usrloc.so"
loadmodule "/usr/local/lib/openser/modules/registrar.so"
loadmodule "/usr/local/lib/openser/modules/textops.so"
loadmodule "/usr/local/lib/openser/modules/auth.so"
loadmodule "/usr/local/lib/openser/modules/auth_db.so"
modparam("usrloc", "db_mode", 0)
modparam("usrloc", "db_mode", 2)
modparam("auth_db", "calculate_ha1", yes)
modparam("rr", "enable_full_lr", 1)
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
return;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
return;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
return;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
return;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
if (!www_authorize("vpbx.mydomain.com",
"subscriber")) {
www_challenge("vpbx.mydomain.com",
"0");
return;
};
save("location");
return;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
return;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
return;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
I found a part of my problem, but I cannot explain it.
The phone (hard phone as well as a softphone) have been set to
connect to port 5062, but I see the registration still on 5060,
where Asterisk is running (on the same machine). The user 800000
has no account on Asterisk, therefore the login fails there.
Any idea where this could come from?
maybe there is a special field to set for registrar address in
phone's configuration, or the phone is broken. What kind of phones
have you tried?
Is there any command line utility to test that?
if you want tot test if the registration works, then you can use
sipsak, from http://sipsak.org, it is a tiny command line utility.
Otherwise, you can try some free sip soft phones: kphone, minisip,
xten xlite.