Hello,
In fact I started with kamailio.
The calls go smoothly on my router (with NAT) kamailio that the proxy is started or not, the calls pass.
I put what in the conf file for the proxy intercepts all calls sip and then forward.
Thank you in advance.
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Hello all
In my kamailio.cfg I don't use append_branch at all and I listen on
multiple addresses. Before running t_relay I need to know on which
socket will the request be sent. I don't want to change the socket,
but I want to know which socket is chosen. I tried
$(branch(send_socket)[0]) and $(branch(send_socket)) but I got error
"ERROR:pv:pv_get_branchx: error accessing branch [0]"
Does anybody know how I can get this info?
Thanks
Catalina
Hi,
----- Original Message ----
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
To: Iñaki Baz Castillo <ibc(a)aliax.net>
Cc: users(a)lists.kamailio.org
Sent: Monday, May 11, 2009 10:04:49 AM
Subject: Re: [Kamailio-Users] two UACs behind same NAT
>>
>> Bad luck then... :(
>> SIP ALG is the worst enemy for SIP.
>>
> workarounds could be:
> - run sip server also on a different port than 5060 (say 5070) -
> kamailio is just fine doing it. Point the users behind such ALGs to this
> port. Still alg can detect it, but most of them do the detection by port
> 5060
> - use TCP if the phone supports it - most of algs do not touch TCP
> connections, but ...
> - use TLS if the phone supports it - safest - the alg cannot touch it
yes, we are planning to use TLS and I hope we won't have that kind of problems then
> - recommended - send back the router and ask for money return
well, I could do that but the problem is not just with this router. I saw a list of this kind of routers (small home routers) that are not recommended to use with SIP, and I think that TLS is only universal solution.
Best regards
Dubravko
Cheers,
Daniel
Hi,
I am trying to make SER working with FreeRADIUS. However, I got an error
using www_challenge, " 0(3184) parse error (119,25-26): unknown command,
missing loadmodule?" The 119 line is www_challenge("", "0"); I did load
auth module. Any idea?
Here is my config:
#
# $Id: ser-basic.cfg,v 1.1.2.2 2008/06/12 13:15:06 alfredh Exp $ # #
This a very basic config file w aliases and anamed route but # w/o
authentication, accounting, database, multi-domain support etc.
# Please refer to ser.cfg for a more complete example #
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
#memdbg=10 # memory debug message level
#memlog=10 # memory statistics log level #log_facility=LOG_LOCAL0 # sets
the facility used for logging (see syslog(3))
/* 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)
listen=202.158.197.134
port=5060
children=4
#user=ser
#group=ser
#disable_core=yes #disables core dumping
#open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes
# usefull for multihomed hosts, small performance penalty
#disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via
option (see NEWS)
#
# ------------------ module loading ----------------------------------
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/textops.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/ctl.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_radius.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# -- rr params --
# add value to ;lr param to make some broken UAs happy modparam("rr",
"enable_full_lr", 1)
# ctl params
# by default ctl listens on unixs:/tmp/ser_ctl if no other address is #
specified in modparams; this is also the default for sercmd
modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") # listen on the
"standard" fifo for backward compatibility modparam("ctl", "fifo",
"fifo:/tmp/ser_fifo") # listen on tcp, localhost #modparam("ctl",
"binrpc", "tcp:localhost:2046")
# -- auth_radius params --
modparam("auth_radius", "radius_config",
"/usr/local/etc/radiusclient-ng/radiusclient.conf")
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_reply("483","Too Many Hops");
break;
}
if (msg:len >= max_len ) {
sl_reply("513", "Message too big");
break;
}
# 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(FORWARD);
break;
}
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(FORWARD);
break;
}
# 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") {
# authentication
if (!radius_www_authorize("")) {
www_challenge("", "0");
break;
};
save_contacts("location");
break;
}
# native SIP destinations are handled using our USRLOC DB
if (!lookup_contacts("location")) {
sl_reply("404", "Not Found");
break;
}
append_hf("P-hint: usrloc applied\r\n");
}
route(FORWARD);
}
route[FORWARD]
{
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
}
}
Thanks,
Leon
Hello,
Is it possible to ban an IP which scan my servers ?
Thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
Hello,
when i start a sip client (x-lite) i have in kamailio debug message this
error: WARNING:core:fm_free: free(0) called
someone can tell me where is the problem??
thank you
------------------------------------------------------------
BAHA Rachid
Elève ingénieur en 3ème année à l'INPT, Rabat
GSM:(+33) 0619787609
Hello,
RTPProxy 1.2.0 can be instructed to notify and external app when RTP
timeout occurs.
It seems this notification must be enabled on a per session basis. So,
does module nathelper have the means to request this notification?
regards,
takeshi
Hi all,
I have a problem when using two UACs
behind same NAT on Linksys WRT54GL router. Registrations, Invites, ...
are handled correctly, but after 30sec in the call UACs hangs up the
call (RTP goes directly between clients). This happens because UAC#1
(which answerd the call with OK) didn't recive ACK from UAC#2. This isn't happening when I have this same clinets behind NAT on PIX525 and everything is working fine. Everithing is also fine when UAC#1 calls someone on PSTN (RTP is handled by RTPProxy) or someone behind other NAT.
Strange thing is that UAC#1 sends Ringing response and OK message with this information in Contact header:
Contact: <sip:bob@public_ip_address:31479;rinstance=e0f8548cf68b5b6e>
but proxy forwards this Ringing response and OK message to UAC#2 with this information in Contact header:
Contact: <sip:bob@public_ip_address:0;rinstance=b18efeaa8a0e3f0f>
so port is "0" and with ngrep on proxy I saw that this port is replaced
with port "5060" and ACK is sent to the UAC#1 with this parameters
public_ip_address:5060 which isn't correct because UAC#1 is using
31478-31488 port range and expects traffic on these ports.
Some of debug messages that i see on proxy are:
1. sl_filter_ACK: to late to be a local ACK!,
2. e2e proxy ACK found
building branch for end2end ACK
totag for e2e ACK found: 1
I'm using Kamailio 1.5.0 with standard kamailio.cfg and newest XLite (with STUN and ICE options turn on).
I would be greatfull if someone can give me any guidelines what to look next in order to solve this issue.
Best regards
Dubravko
Hi,
Creating openser database on the localhost works fine using "kamdbctl create".
However, I need to create openser database on an external mysql server. I tried the following command but it's not working:
kamdbctl create openser@<external_mysql_server_ip>
What's the correct command?
Thanks,
A R
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Hi all,
I am new to openser.
I have register two sip user in openser (as register server) and call
handling in asterisk.
when 1001 user do a call to 1002 nothing happen.
call rejected.
If posible give a sip.conf and extension.conf snap of this scenario.
any suggestion?
Bhrugu Mehta
