I'm trying hard to succeed with NAT without STUN.
I have seen the default ser.cfg provided in the mediaproxy module
(0.8.14 as well as 0.9).
To me it seems as if the use_media_proxy() function is called for all
calls and not for NATed only (media is relayed for all calls) because
the setflag(2) set during registration is never checked anywhere..
Am I right in thinking so?
Please help.
try this:
1) make distclean
2) CPPFLAGS='-I/usr/lib/qt3/include -L/usr/lib/qt3/lib' ./configure --prefix=/otp/kphone-4.1.0
3) make
4) make install
bye
Andrea
-----------------------------------------------------------------------------
Some influential environment variables:
CC C compiler command
CFLAGS C compiler flags
LDFLAGS linker flags, e.g. -L<lib dir> if you have libraries in a
nonstandard directory <lib dir>
CPPFLAGS C/C++ preprocessor flags, e.g. -I<include dir> if you have
headers in a nonstandard directory <include dir>
CPP C preprocessor
CXX C++ compiler command
CXXFLAGS C++ compiler flags
CXXCPP C++ preprocessor
-----------------------------------------------------------------------------
--
___________________________________________________________
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Hello,
I'm sure this may have been asked but, I just wanted clarify how this
would be accomplished. I'm looking to have the MWI light on my cisco
phone turn on when a message is left in VM.
Here is my current situation:
sip user -> ser -> asterisk
Best regards,
Patrick b.
hello ser users;
first, please don't pay attention to my english, because i'm french :-)...
I'm using the pbx asterisk and ser as a sip proxy, to pass over nat
problems. Here's my configuration :
ipphone (192.168.0.108) --->
asterisk (192.168.0.4:5061)
SER (192.168.0.4:5060) --->
nat router...
During the call, a sip message comes from asterisk to SER, but when it
arrives at the router, it seems to be corrupted.
Here's the sip/sdp message before SER :
Session Initiation Protocol
Request line: INVITE sip:0467751647@212.94.190.153;user=phone SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.0.4:5061;branch=z9hG4bK653c2038
Route: <sip:22222@212.94.190.153;user=phone>
From: "11111" <sip:11111@192.168.0.4:5061>;tag=as5ea95a2d
To: <sip:22222@192.168.0.4>;tag=1c11534
Contact: <sip:11111@192.168.0.4:5061>
Call-ID: 5de9a6a3391006b40343cfc8370a4aa0(a)192.168.0.4
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 264
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 23042 23043 IN IP4 192.168.0.108
Owner Username: root
Session ID: 23042
Session Version: 23043
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 192.168.0.108
Session Name (s): session
Connection Information (c): IN IP4 192.168.0.108 ...
the same after being routed by SER :
Session Initiation Protocol
Request line: INVITE sip:22222@212.94.190.153;user=phone SIP/2.0
Message Header
Max-Forwards: 10
Via: SIP/2.0/UDP 192.168.0.4;branch=z9hG4bK5fe.bf341295.0
Via: SIP/2.0/UDP 192.168.0.4:5061;rport=5061;branch=z9hG4bK653c2038
From: "11111" <sip:11111@192.168.0.4:5061>;tag=as5ea95a2d
To: <sip:22222@192.168.0.4>;tag=1c11534
Contact: <sip:11111@192.168.0.4:5061>
Call-ID: 5de9a6a3391006b40343cfc8370a4aa0(a)192.168.0.4
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 353
Route: <sip:22222@212.94.190.153;user=phone>
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 23042 23043 IN IP4 192.168.0.108
Owner Username: root
Session ID: 23042
Session Version: 23043
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 192.168.0.108
Session Name (s): session
Connection Information (c): IN IP4
192.168.0.482.231.33.XXX192.168.0.4 ...
...as tou can see, the connection information in the sdp message is a
bit strange...
My ser.cfg :
-------------------------
check_via=yes # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
#loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/mangler.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.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"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
#loadmodule "/usr/local/lib/ser/modules/auth.so"
#loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
#modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
#modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
#modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock")
# ------------------------- 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_send_reply("483","Too Many Hops");
break;
};
if (msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
log(1, "nouvelle trame reçue\n");
if(method=="INVITE"){
log(1, "invite route\n");
add_rport();
fix_nated_sdp("3");
fix_nated_contact();
sdp_mangle_ip("192.0.0.0/255.0.0.0", "82.231.33.XXX");
force_rtp_proxy();
t_on_reply("1");
}
if (loose_route()) {
log(1, "loose_route\n");
t_relay();
break;
}
if(method=="INVITE"){
#log(1, "trame INVITE\n");
record_route();
rewritehostport("212.94.190.153:5060");
}
log(1, "relai de la trame\n\n");
if (!t_relay()) {
log(1, "erreur de relai\n\n");
sl_reply_error();
}
}
onreply_route[1]{
log(1, "on reply\n");
if(status=~"[12][0-9][0-9]")
force_rtp_proxy();
}
----------------------------
I have the same problem with ser 0.8.14 and 0.9.0. So my question is :
is this a bug in ser, or something strange in my ser.cfg ?
Did anybody ever see something like this ???
thanks.
thibault.
Conditions as follows :
* SER runs on a Public IP
* SER works without auth & database modules,
* Nearly all user behind NAT (but routers configured to do port forwarding for TCP/UDP 5060) to help SER in some cases,
* Users numbers in format of 833XXXXXXX 834XXXXXXX and they should call each P2P-SIP-Calls (if not behind NAT),
* If a user need to call PSTN end point (SIP Gateway located at 212.154.32.154) the call traffic should flow over SER to SIP Gateway via T1 connection already located between that systems so SER handles all voice traffic by help of RTP Proxy.
* UA's registers on SER (Zyxel Prestige 2000, Zyxel Prestige 200W, Cisco ATA186 etc.)
Problem is users cannot call each other (if i comment lines for nathelper they can call)
It's clear i think, and below is my ser.cfg, what do i need extra or erase.
<-<-<-<-< MY SER.CFG STARTS HERE >->->->->
#
# $Id nathelper.cfg,v 1.1.2.1 20050301 by Ozan Blotter Exp $
#
# simple quick-start config script including nathelper support
# This default script includes nathelper support. To make it work
# you will also have to install Maxim's RTP proxy. The proxy is enforced
# if one of the parties is behind a NAT.
#
# If you have an endpoing in the public internet which is known to
# support symmetric RTP (Cisco PSTN gateway or voicemail, for example),
# then you don't have to force RTP proxy. If you don't want to enforce
# RTP proxy for some destinations than simply use t_relay() instead of
# route(1)
#
# Sections marked with !! Nathelper contain modifications for nathelper
#
# NOTE !! This config is EXPERIMENTAL !
#
# ----------- global configuration parameters ------------------------
# debug=3 # debug level (cmd line -dddddddddd)
# fork=yes
# log_stderror=no # (cmd line -E)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line -v)
dns=no # (cmd. line -r)
rev_dns=no # (cmd. line -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
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/textops.so"
# !! Nathelper
loadmodule "/usr/local/lib/ser/modules/nathelper.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)
# !! Nathelper
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 10) # Ping interval 10 seconds
modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
# ------------------------- 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_send_reply("483","Too Many Hops");
break;
};
if (msg:len > max_len ) {
sl_send_reply("513", "Message Too Big");
break;
};
# if ((method=="NOTIFY")&& search("^Event: Keep-Alive")) {
# ls_send_reply("200","OK");
# break;
# };
# !! Nathelper
# Special handling for NATed clients; first, NAT test is
# executed it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding is used); also,
# the received test should, if completed, should check all
# vias for rpesence of received
if (nat_uac_test("3")) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if (method=="REGISTER" || ! search("^Record-Route:")) {
# log("LOG: Someone trying to register from private IP, rewriting\$
# This will work only for user agents that support symmetric
# communication. We tested quite many of them and majority is
# smart enough to be symmetric. In some phones it takes a configuration
# option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it $
# called symmetric media and symmetric signalling.
fix_nated_contact(); # Rewrite contact with source IP of signalling
if (method == "INVITE") {
fix_nated_sdp("1"); # Add direction=active to SDP
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as NATed
};
};
# 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);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
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") {
if (!(uri=~"sip:(833)|(834)")) {
t_relay_to_udp("212.154.32.154","5060");
save("location");
break;
};
# lookup(aliases);
# if (!uri==myself) {
# append_hf("P-hint: outbound alias\r\n");
# route(1);
# break;
# };
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# !! Nathelper
if (uri=~"[@:](192\.168\.|10\.172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !searc$
sl_send_reply("479", "We don't forward to private IP addresses");
break;
};
# if client or server know to be behind a NAT, enable relay
if (isflagset(6)) {
force_rtp_proxy();
};
# NAT processing of replies; apply to all transactions (for example,
# re-INVITEs from public to private UA are hard to identify as
# NATed at the moment of request processing); look at replies
t_on_reply("1");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
# !! Nathelper
onreply_route[1] {
# NATed transaction
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
if (!search("^Content-Length:\0")){
force_rtp_proxy();
};
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
if (!search("^Content-Length:\0")){
force_rtp_proxy();
};
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
<-<-<-<-< MY SER.CFG ENDS HERE >->->->->
I have the following error while restarting the SER:
connect_db(): Client does not support authentication protocol
requested by server. Consider upgrading MySQL client.
I have upgraded the server, the client and the librarys, but it still does
not work. Can anybody help me please?
Thank you.
Sandra García Espada
DMR Consulting
PLEASE HELP ME...
________________________________
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Lakshmi Sharma
Sent: Thursday, February 17, 2005 6:04 PM
To: serusers(a)lists.iptel.org
Subject: [Serusers] SIP user authentication
Hello,
I've installed the latest version of SER on SUSE linux 9.0. Setup MYSQL
as well to test sutehntication/challenge. When SER receives Register
request, I'm seeing these msgs on console:
"Received SIP_msg. No mem for the SIP_msg".
It would be great if someone could help me with this.
Thanks & Regards
Lakshmi
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