Happy new year everyone!!
first of all, I'd like to thank those who answered my questions last time.
now I've got a question with Portaone RTP prozy (which recommended in
the voip-info.org web site.
I downloaded the tar ball and compile the source code. I executed the
command ./rtpproxy and saw it running by "ps" it. Now when I typed "ser
-E" on my console, I've got the following errors.
[root@localhost ser]# ser -E
0(6085) WARNING: could not rev. resolve 192.168.1.101
Listening on
127.0.0.1 [127.0.0.1]:5060
192.168.1.101 [192.168.1.101]:5060
Aliases: localhost:5060 localhost.localdomain:5060
stateless - initializing
Maxfwd module- initializing
0(6087) mod_init(): Database connection opened successfuly
exec - initializing
print - initializing
textops - initializing
0(0) INFO: udp_init: SO_RCVBUF is initially 65535
0(0) INFO: udp_init: SO_RCVBUF is finally 131070
0(0) INFO: udp_init: SO_RCVBUF is initially 65535
0(0) INFO: udp_init: SO_RCVBUF is finally 131070
[root@localhost ser]# 1(0) ERROR: send_rtpp_command: can't read reply
from a RTP proxy
1(0) WARNING: rtpp_test: can't get version of the RTP proxy
1(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
2(0) ERROR: send_rtpp_command: can't read reply from a RTP proxy
2(0) WARNING: rtpp_test: can't get version of the RTP proxy
2(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
3(0) ERROR: send_rtpp_command: can't read reply from a RTP proxy
3(0) WARNING: rtpp_test: can't get version of the RTP proxy
3(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
4(0) ERROR: send_rtpp_command: can't read reply from a RTP proxy
4(0) WARNING: rtpp_test: can't get version of the RTP proxy
4(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
5(0) ERROR: send_rtpp_command: can't read reply from a RTP proxy
5(0) WARNING: rtpp_test: can't get version of the RTP proxy
5(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
6(0) ERROR: send_rtpp_command: can't read reply from a RTP proxy
6(0) WARNING: rtpp_test: can't get version of the RTP proxy
6(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
7(0) ERROR: send_rtpp_command: can't read reply from a RTP proxy
7(0) WARNING: rtpp_test: can't get version of the RTP proxy
7(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
8(0) ERROR: send_rtpp_command: can't read reply from a RTP proxy
8(0) WARNING: rtpp_test: can't get version of the RTP proxy
8(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
9(0) INFO: fifo process starting: 6130
9(0) ERROR: send_rtpp_command: can't read reply from a RTP proxy
9(0) WARNING: rtpp_test: can't get version of the RTP proxy
9(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
10(0) ERROR: send_rtpp_command: can't read reply from a RTP proxy
10(0) WARNING: rtpp_test: can't get version of the RTP proxy
10(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
11(0) ERROR: send_rtpp_command: can't read reply from a RTP proxy
11(0) WARNING: rtpp_test: can't get version of the RTP proxy
11(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
9(6130) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo...
12(0) ERROR: send_rtpp_command: can't read reply from a RTP proxy
12(0) WARNING: rtpp_test: can't get version of the RTP proxy
12(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
13(0) ERROR: send_rtpp_command: can't read reply from a RTP proxy
13(0) WARNING: rtpp_test: can't get version of the RTP proxy
13(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
14(0) ERROR: send_rtpp_command: can't read reply from a RTP proxy
14(0) WARNING: rtpp_test: can't get version of the RTP proxy
14(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
0(6087) ERROR: send_rtpp_command: can't read reply from a RTP proxy
0(6087) WARNING: rtpp_test: can't get version of the RTP proxy
0(6087) WARNING: rtpp_test: support for RTP proxyhas been disabled
temporarily
15(6161) ERROR: send_rtpp_command: can't read reply from a RTP proxy
15(6161) WARNING: rtpp_test: can't get version of the RTP proxy
15(6161) WARNING: rtpp_test: support for RTP proxyhas been disabled
temporarily
That was asked back to December 8 but there seemed no answer to that
question. Thus can please anyone help?
thanks
Proson
I've never used a 5300 but if I had a guess it would be that the 5300 doesn't realize that the RTP is NAT'd (and should send to a proxy). In other words it may not be sending to your rtpproxy correctly if at all? Do a debug on all devices, ensure the RTP ports match up. Does rtpproxy report any errors such as no payload? You can do an 'rtpproxy -f' to run it at command line.
Looking at your config you have a lot of stuff being done before you relay to the 5300, not to mention you have 'forward' under your t_relay..?? Have you tried just sending to route(1) ?? That's where all your 'if natted' statements should be. That way on_reply gets caught correctly as well.
ie:
if ( uri=~"^sip:9[0-9]*@5300_IP" ) {
log(1, "SER22: matched - forward to GW\n");
rewritehostport ("5300_IP:5060");
route(1)
break;
};
-----Original Message-----
From: Victor Chan [mailto:vchan67@hotmail.com]
Sent: Wednesday, December 29, 2004 8:11 PM
To: serusers(a)lists.iptel.org
Subject: [Serusers] Ser, Cisco and NAT
I have some problem about Cisco and NAT device. I have SER and rtpproxy
running on a server. Call is made from a SIP endpoint device to Cisco 5300
gateway via SER. But voice can't hear at my SIP device. I try point to
point call without going to Cisco gateway, both side can hear. Please give
advise to solve it. Thanks a lot.
The following is my ser.cfg
#
# $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj 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= 7 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
# 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)
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/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/textops.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"
# !! Nathelper
loadmodule "/usr/local/lib/ser/modules/nathelper.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)
# !! Nathelper
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
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;
};
# !! 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\n");
# 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 is
# 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
log(1, "loose_route processing\n");
append_hf("P-hint: rr-enforced\r\n");
t_relay();
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") {
save("location");
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
if ( uri=~"^sip:9[0-9]*@5300_IP" ) {
log(1, "SER22: matched - forward to GW\n");
fix_nated_sdp("2");
force_rtp_proxy("5300_IP");
rewritehostport ("5300_IP:5060");
t_relay_to_udp("5300_IP","5060");
forward(5300_IP, 5060);
break;
};
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])\.)" &&
!search("^Route:")){
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();
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();
};
}
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I am a newcome ,I install a SER server ,it's working well
[root@a sbin]# ser -dddd
Listening on
127.0.0.1 [127.0.0.1]:5060
221.202.129.165 [221.202.129.165]:5060
Aliases: a.jzmv.com:5060 localhost:5060
tail -f /var/log/messages
Dec 31 09:14:17 a ser[16586]: mod_init(): Database connection opened successfuly
Dec 31 09:14:17 a ser[16586]: INFO: udp_init: SO_RCVBUF is initially 65535
Dec 31 09:14:17 a ser[16586]: INFO: udp_init: SO_RCVBUF is finally 262142
Dec 31 09:14:17 a ser[16586]: ERROR: tcp_init: bind(6, 0x80d8320, 16) on 127.0.0.1: Address already in use
Dec 31 09:14:17 a ser[16586]: INFO: signal 15 received
#serctl add zhaomin zm9898 zhaomin(a)jzmv.com
OK
but I used the sipphone by the user name zhaomin login .it tell me login is failt.my ser.cfg is blow:
debug=7
fork=yes
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/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"
# 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", 2)
# 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)
# ------------------------- 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;
};
# 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
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
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") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("jzmv.com", "subscriber")) {
www_challenge("jzmv.com", "0");
break;
};
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
Where is my failt ? anyones can help me ?
Thanks
ZhaoMin
recently i had reported 2 problems. while waiting to hear from the list, i
ran some experiments as follows: first, i switched from ser-0.9.0 to
ser-0.8.14. i then re-initialized the ser database. afterwards, i was able
to do a 'serctl add ..." successfuly. also, i'm now able to authenticate my
registration requests. so both problems seem be gone ;-)
i'd like to understand one thing better however: how does the information in
subscriber database get to be used during the authentication process? stated
in other words, which fields of the subscriber table get to be used during
authentication?
thanks & regards - ga
folks
i'd appreciate if i could get some pointers re: following incidences...
1. my attempts to invoke serctl to add a new subscriber fail [error: 400;
check if you use aliases in SER] at all times which suggest i must be doing
something wrong.
2. when i enable the authentication code block in ser.cfg, my 1st
registration attempt fails with a response type of 401 (unauthorized) which
contains a WWW-Authenticate header. subsequently, my ua (x-lite softphone)
sends another register message which contains an Authorization header with
some number of credentials. unfortunately, this 2nd attempt also results in
a 401 response. are there certain things that i should do on the
client/server side to get ser authentication to work?
thanks for your help in advance,
gurol
Hi there
after slashing my head with a hammer for a little bit,
reading many times the SER RADIUS Howto and acc man
page, I managed to get acc to log some RADIUS accounting
packets (Stop records and Call Failure records) with
FreeRADIUS
unfortunately, I could not discover how to generate
the START records, no matter what I tried!!
as the Stop record doesn't have RADIUS session-time,
I need the Start record so my current billing system
can be used.
I've searched the list archives quite a bit, and
googled on [serusers] and FreeRADIUS lists but
couldn't find a clue.
any help would be gratelly appreciated
snippets of ser.cfg
[...]
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_radius.so"
# acc compiled with RADIUS support
loadmodule "/usr/lib/ser/modules/acc.so"
loadmodule "/usr/lib/ser/modules/nathelper.so"
# -- acc params --
modparam("acc", "radius_config", "/etc/radiusclient/radiusclient.conf")
modparam("acc", "radius_flag", 1)
modparam("acc", "radius_missed_flag", 2)
#modparam("acc", "log_level", 1)
#modparam("acc", "failed_transactions", 1)
#modparam("acc", "report_cancels", 1)
#modparam("acc", "report_ack", 1)
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# -- auth_radius params --
modparam("auth_radius", "radius_config",
"/etc/radiusclient/radiusclient.conf")
# -- 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", 30) # Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
[...] pretty much nathelper.cfg example [...]
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
log(1,"LOG: not found\n");
setflag(2);
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])\.)"
&& !search("^Route:")){
sl_send_reply("479", "We don't forward to private IP
addresses");
break;
};
log(1, "LOG: route[1]\n");
setflag(1);
# if client or server know to be behind a NAT, enable relay
if (isflagset(6)) {
force_rtp_proxy();
};
t_on_reply("1");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
onreply_route[1] {
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
# CHECK LATER: does this need to be enclosed on
# if(method==BYE || method==CANCEL) ???
force_rtp_proxy();
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
# EOF ---
one thing that I noticed was that, depending on which
side ended the call, the RADIUS Stop packet cames with
that side being the calling-station-id and the peer
being the called-station-id. is it supposed to be that
way?
Cheers
!3runo
P.S. in case it matters:
# ser -V
version: 0.8.14 (i386/linux)
flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.168.4.3 2004/06/28 15:41:21 andrei Exp $
main.c compiled on 07:13:08 Nov 17 2004 with gcc 3.3
Hello SER users,
I need some assistance with SER and NAT. I have issues with users
behind NAT that seem to get disconnected using a soft SIP phone (i.e.
eyePMedia or X-Ten). I am a newbie so configurations are appreciated.
Also any comments and suggestions are welcome.
Thank you
Jason Kline
Sr. Network Engineer
IDS Telcom, LLC
Ph. 305-612-4325
Fax: 305-612-3224
jkline(a)idstelcom.com
http://www.idstelcom.com <http://www.idstelcom.com/>
Hello,
First of all excuse my newbieness. I have just installed SER in a public
IP. I read Dan Austin's howto, and understood that I would be able to
register using the default configuration file. I installed and configured
kphone with these values:
Full Name: Diego M. Vadell
User part of SIP URL: dvadell
Host part of SIP URL: 200.69.255.72 (the public IP of SER)
Outbound proxy (optional): 200.69.255.72
but I get a 404 Not found from SER.
Could anyone guide me? Im lost here. Do I need to setup mysql to register?
Here are some lines of information, both from kphone and from SER.
Thanks in advance,
-- Diego.
Kphone says:
bash-2.04$ kphone
Found 2 interfaces.
SipClient: Listening UDP on port: 5060
SipClient: Our address: 200.126.216.62
SipRegister: Auth is '(null)'
SipRegister: Proxy Auth is '(null)'
SipClient: Sending: 15:54:15.756
--------------------------------
REGISTER sip:200.69.255.72 SIP/2.0
Via: SIP/2.0/UDP 200.126.216.62;branch=z9hG4bK7578380A
CSeq: 107 REGISTER
To: "Diego M. Vadell" <sip:dvadell@200.69.255.72>
Expires: 900
From: "Diego M. Vadell" <sip:dvadell@200.69.255.72>
Call-ID: 1930116421(a)200.126.216.62
Content-Length: 0
User-Agent: kphone/4.0.5
Event: registration
Allow-Events: presence
Contact: "Diego M. Vadell"
<sip:dvadell@200.126.216.62;transport=udp>;methods="INVITE, MESSAGE, INFO,
SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER"
SipClient: Sending to '200.69.255.72:5060'
SipClient: Receiving message...
SipClient: Received: 15:54:15.869
---------------------------------
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 200.126.216.62;branch=z9hG4bK7578380A
CSeq: 107 REGISTER
To: "Diego M. Vadell"
<sip:dvadell@200.69.255.72>;tag=b27e1a1d33761e85846fc98f5f3a7e58.cf28
From: "Diego M. Vadell" <sip:dvadell@200.69.255.72>
Call-ID: 1930116421(a)200.126.216.62
Server: Sip EXpress router (0.8.14 (i386/linux))
Content-Length: 0
Warning: 392 200.69.255.72:5060 "Noisy feedback tells: pid=14352
req_src_ip=200.126.216.62 req_src_port=5060 in_uri=sip:200.69.255.72
out_uri=sip:200.69.255.72 via_cnt==1"
And SER says:
19(14352) SIP Request:
19(14352) method: <REGISTER>
19(14352) uri: <sip:200.69.255.72>
19(14352) version: <SIP/2.0>
19(14352) parse_headers: flags=1
19(14352) Found param type 232, <branch> = <z9hG4bK7578380A>; state=16
19(14352) end of header reached, state=5
19(14352) parse_headers: Via found, flags=1
19(14352) parse_headers: this is the first via
19(14352) After parse_msg...
19(14352) preparing to run routing scripts...
19(14352) DEBUG : is_maxfwd_present: searching for max_forwards header
19(14352) parse_headers: flags=128
19(14352) get_hdr_field: cseq <CSeq>: <107> <REGISTER>
19(14352) end of header reached, state=9
19(14352) DEBUG: get_hdr_field: <To> [47]; uri=[sip:dvadell@200.69.255.72]
19(14352) DEBUG: to body ["Diego M. Vadell" <sip:dvadell@200.69.255.72>
]
19(14352) DEBUG: get_hdr_body : content_length=0
19(14352) found end of header
19(14352) DEBUG: is_maxfwd_present: max_forwards header not found!
19(14352) end of header reached, state=9
19(14352) parse_headers: flags=256
19(14352) find_first_route(): No Route headers found
19(14352) loose_route(): There is no Route HF
19(14352) check_self - checking if host==us: 13==9 && [200.69.255.72] ==
[127.0.0.1]
19(14352) check_self - checking if port 5060 matches port 5060
19(14352) check_self - checking if host==us: 13==10 && [200.69.255.72] ==
[10.5.5.242]
19(14352) check_self - checking if port 5060 matches port 5060
19(14352) check_self - checking if host==us: 13==13 && [200.69.255.72] ==
[200.69.255.72]
19(14352) check_self - checking if port 5060 matches port 5060
19(14352) lookup(): '' Not found in usrloc
19(14352) parse_headers: flags=-1
19(14352) check_via_address(200.126.216.62, 200.126.216.62, 0)
19(14352) DEBUG:destroy_avp_list: destroing list (nil)
19(14352) receive_msg: cleaning up
Hi,
I need some help with digest authentication.
When I uncomment those lines in ser.cfg, the register msg stops to work. In the trace, see below, you can see the nonce being sent in the re-register msg, but the server still responds with 401 Unauthorized. I've tried with both 0 and 1 in the www_challenge.
Without the digest authentication the register works fine.
Thanks in advance,
//Magnus
ser.cfg (ser 0.8.12 running on a Fedora box. Used for test purpose only):
====================================================
# ----------- 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"
sip_warning=no
alias="sip_server_ip"
# ------------------ module loading ----------------------------------
# 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/pa.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/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)
# ------------------------- 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;
};
# 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
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
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=="SUBSCRIBE") {
if(t_newtran()){
handle_subscription("registrar");
break;
};
};
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("sip_server_ip", "subscriber")) {
www_challenge("sip_server_ip", "1");
break;
};
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
Register trace:
==========
REGISTER sip:sip_server_ip SIP/2.0
Via: SIP/2.0/UDP local_pc_ip:5060;rport;branch=z9hG4bK4268DFDFE5EE410C8DB113A6223C800C
From: Magnus <sip:magnus@sip_server_ip>;tag=470300110
To: Magnus <sip:magnus@sip_server_ip>
Contact: "Magnus" <sip:magnus@local_pc_ip:5060>
Call-ID: EB7272E371C24F6C8F24DB47A53EE7CB@sip_server_ip
CSeq: 6590 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP local_pc_ip:5060;rport=5060;branch=z9hG4bK4268DFDFE5EE410C8DB113A6223C800C
From: Magnus <sip:magnus@sip_server_ip>;tag=470300110
To: Magnus <sip:magnus@sip_server_ip>;tag=b27e1a1d33761e85846fc98f5f3a7e58.0d0e
Call-ID: EB7272E371C24F6C8F24DB47A53EE7CB@sip_server_ip
CSeq: 6590 REGISTER
WWW-Authenticate: Digest realm="sip_server_ip", nonce="41d1321431d402c1af9617eb73deccbce7e532d5", qop="auth"
Server: Sip EXpress router (0.8.12 (i386/linux))
Content-Length: 0
REGISTER sip:sip_server_ip SIP/2.0
Via: SIP/2.0/UDP local_pc_ip:5060;rport;branch=z9hG4bK1813C486770C442BB51E58686A61921F
From: Magnus <sip:magnus@sip_server_ip>;tag=470300110
To: Magnus <sip:magnus@sip_server_ip>
Contact: "Magnus" <sip:magnus@local_pc_ip:5060>
Call-ID: EB7272E371C24F6C8F24DB47A53EE7CB@sip_server_ip
CSeq: 6591 REGISTER
Expires: 1800
Authorization: Digest username="magnus",realm="sip_server_ip",nonce="41d1321431d402c1af9617eb73deccbce7e532d5",response="27ea80aed1b9f5086b396c8f86bcec60",uri="sip:sip_server_ip",qop=auth,cnonce="9F5BBA98D6724D909C6560E8A045A300",nc=00000006
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP local_pc_ip:5060;rport=5060;branch=z9hG4bK1813C486770C442BB51E58686A61921F
From: Magnus <sip:magnus@sip_server_ip>;tag=470300110
To: Magnus <sip:magnus@sip_server_ip>;tag=b27e1a1d33761e85846fc98f5f3a7e58.9cf2
Call-ID: EB7272E371C24F6C8F24DB47A53EE7CB@sip_server_ip
CSeq: 6591 REGISTER
WWW-Authenticate: Digest realm="sip_server_ip", nonce="41d1321431d402c1af9617eb73deccbce7e532d5", qop="auth"
Server: Sip EXpress router (0.8.12 (i386/linux))
Content-Length: 0
Erik,
CDRTool guarantees 100% accurate accounting for call detail records
generated by SIP Express Router in combination with MediaProxy
regardless of the presence of BYE messages. This solution is available
under commercial agreement.
http://www.ag-projects.com/CDRTool.html
Regards,
Adrian
>>>>>>>>>>>>
OK, i've run into a problem with the CDR Creation.
If you have 2 users who call each other, talk a while and then instead
of hanging up decide to pull the plugs from their phones (or terminitate
their application), no complete CDR is generated, the INVITE and ACK are
logged, but since no one realy hangs up there won't be a BYE record, so
no CDR end/total time and no billable time.
Use the / a rtpproxy you would say, to bad it suffers from the same
problem, if both partes put the call on hold and pull the plug the call
keeps existing and once again an incomplete CDR.
Has anyone ever suffered the same problem? And what is the most reliable
way to generate CDRs?
Kind regards,
E. Versaevel
