Hi there i have compiled the ser with auth_radius module.
During my test in the log i can see that i get:
res: -1
3(25958) radius_authorize_sterman(): Failure
3(25958) build_auth_hf(): 'WWW-Authenticate: Digest
realm="209.250.135.8",
nonce="4175e8b266ce69371b9d12a6b69c4ce3b541b313"^M
i did followed everything in the FAQ and radius documentation ..
But for some reason i cannot get through this problem.
the radius client is the 0.4.5 but it does not help.
Another issue is that i run radius -X and nothing ... nothing is
showing on the log ???
when i run radtest as in document ion i do get the log ... in the radius ..
If any one know where i did get stuck ??? please let me know
Thanks
Hi,
I have just installed ser this week.
It's very cool and happy to find FreeBSD port of it.
Anyway,
I have rtp proxy connection problem of ser+nathelper+rtpproxy system.
To refer console log, it seems no communication between ser and rtpproxy.
Does anyone know how to fix it?
------
gw# /usr/local/sbin/ser -D -E
-Listening on
:
:
stateless - initializing
Maxfwd module- initializing
0(47772) mod_init(): Database connection opened successfuly
textops - initializing
0(0) INFO: udp_init: SO_RCVBUF is initially 41600
0(0) INFO: udp_init: SO_RCVBUF is finally 231936
0(0) WARNING: using only the first listen address (no fork)
2(47774) INFO: fifo process starting: 47774
1(47773) ERROR: send_rtpp_command: can't read reply from a RTP proxy
1(47773) 2(47774) WARNING: rtpp_test: can't get version of the RTP proxy
ERROR: send_rtpp_command: can't read reply from a RTP proxy
1(47773) 2(47774) WARNING: rtpp_test: support for RTP proxyhas been
disabled temporarily
------
Here is my environmnet:
OS : FreeBSD 4.10-RELEASE-p2
ser version: ser 0.8.14-2 (i386/freebsd)
get stable version from CVS, since port version is still 0.8.12...
nathelper : get from CVS 14-Oct-2004
I run ser and rtpporoxy as root user to avoid permission issue.
and /var/run looks no problem.
srwxr-xr-x 1 root wheel 0 Oct 15 14:53 rtpproxy.sock
----
# ----------- 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
*/
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"
#fifo_mode=0666
### temporary for ser to write/read /var/run/rtpproxy.sock
#uid="nobody"
#gid="nobody"
# ------------------ 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_url", "sql://ser:ser@localhost/ser")
#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
:
:
:
Thanks in advance!
*~~*~~*~~*~~*~~*~~*~~*~~*~~*~~*~~*~~*
Akira Okubo <akira(a)access-sys-eu.com>
I am having problems getting exec_dst to work
correctly. I am loading the module, but every time
that I attempt to start SER I am getting an error
saying that I haven't loaded the module.
I am using the latest stable SER 0.8.14 - downloaded
from CVS today.
Here are snipets from my SER.CFG.
loading the module....
-----
loadmodule "/usr/local/lib/ser/modules/exec.so"
-----
calling exec_dst....
-----
#exec external app to determine dest URI
if (exec_dst("/usr/local/sbin/sip-route")) {
route(1);
break;
};
-----
The error that I am getting is...
-----
0(20149) parse error (161,57-58): unknown command,
missing loadmodule?
-----
Line 161 in my ser.cfg file is the "exec_dst" line
shown in my snipet above.
I have verified that the exec.so module exists, and
have even tried recompiling / reinstalling it.
Any suggestions?
Thanks!
Darren Nay
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Hi,
I have a Comtrend SIP UA here and it breaks when I have
'sip_warning=yes' in ser.cfg. It doesn't like the warning header to be
too long. I couldn't find anywhere in the SIP RFC's whether there is a
maximum length to this header.
When I set sip_warning to no, the UA works fine, however I think I read
somewhere once that this warning header is sometimes mandatory. Can
someone confirm or deny that ?
Thanks,
Leon de Rooij
leon(a)scarlet-internet.nl
Anyone who can help me with this issue?
Best Regards
Ricardo
> -----Mensaje original-----
> De: Ricardo Martinez
> Enviado el: Lunes, 18 de Octubre de 2004 12:04
> Para: 'serusers(a)lists.iptel.org'
> Asunto: INVITE change by Radius Authorization
>
> Hello List.
> I'm using SER as my SIP platform, i'm still using it in a lab
> enviroment, i need to solve some issues before move it to a "normal"
> enviroment.
> What i want to do is make some changes in the authorization of a call.
> Now, when a INVITE arrives to my SER platform the user is challenged by
> radius in my DB. This is done by the : radius_proxy_authorize. For the
> other hand I'm able to return any parameter in the radius Access-Response
> message. So, here is my question: Is possible for SER to "understand"
> this access-response and the parameters that contains in it?. For example
> is possible to return in a radius access-response a parameter called
> "Redirect-Number" that changes the original dialed number by the user for
> another number?. In a diagram may look like this :(this is a very easy
> example)
>
>
> User dial 00-562-2408190 ----------- Challenged with Radius --------
> Radius Return Parameter Redirect-Number=56-2-2204567 ---- The call is
> placed.
>
> Is possible to do this?.
> Do i have to use maybe a exec module or something like this?
> Any idea?
>
> Thanks in advace.
> Best Regards
>
> Ricardo Martinez
What's the status of this problem from September? I can't find any info - using sems and ser from CVS 2 weeks ago.
Separate question: when I move the loose-route processing further down in my ser.cfg (after the lookup(aliases); and if(uri==myself){stuff}; blocks), then my PBX gateway (multitech MVP130) drops calls after 1 minute as "unsuccessfull" - Multitech support says that the gateway isn't getting a certain "200 OK" message. I don't know SIP well enough to know exactly what's up there - has anyone heard of this, or do I get to find a hub so I can sniff out the traffic going to the MVP130?
Rob
From:Jan Janak jan at iptel.org
Date: Wed Sep 15 15:28:53 CEST 2004
Subject: [Serdev] tm module and voicemail of sems
Hello,
t_write_req currently does not work voicemail properly because some
parts (to load the email from the dabase) seem to be missing. We will
fix it asap, sorry for the inconvenience.
Jan.
On 15-09 17:09, Zhang Wei wrote:
> hello,
> My ser does not work well with voicemail of sems. No email address is passed to sems . In ser.cfg I wrote :
> -------------------------------------------------
>
> if( is_user_in("Request-URI", "voicemail")){
> log(1, "yes,incoming voicemail call\n");
> if(!t_write_req("/tmp/am_fifo","voicemail")){
> log("could not contact voicemail server\n");
> t_reply("500","could not contact voicemail serv
> };
> }
> ---------------------------------------------------------------
> and the part debug information of sems is :
> ------------------------------------------------------------------
>
> (21112) DEBUG: execute (AmServer.cpp:226): cmd.method= <INVITE>
> (21112) DEBUG: execute (AmServer.cpp:227): cmd.user= <zhangwei>
> (21112) DEBUG: execute (AmServer.cpp:228): cmd.email= <>
> ......
>
> (21112) DEBUG: execute (AmServer.cpp:276): everything is OK !
> (21112) ERROR: startSession (AmSession.cpp:458): 404 voicemail: no email address for user <zhangwei>
> (21112) DEBUG: sendToFIFO (AmRequest.cpp:230): msg=<:t_reply:0000527827DE965E
> 404
> voicemail: no email address for user <zhangwei>
> 5491:1720953846
> 000052782F008C5D
> Contact: <sip:zhangwei at 210.51.11.47>
> .
> ------------------------------------------------------
> In source code, t_write_req()-->assemble_msg()-->search_first_avp()
> In search_first_avp function of usr_avp.c :
> ------------------------------------------------------
> struct usr_avp *avp;
>
> assert( crt_avps!=0 );
>
> if (*crt_avps==0)
> return 0; -------------------->return
> ----------------------------------------------------------
>
> Why ???
HI All;
I have the following network:
cisco ATA (valid ip) ---------------------------ser+rtpproxy---------------------------quintum GW(valid ip)
62.220.101.8 62.220.101.2 62.220.101.3
I want to pass media via ser+rtpproxy, ser can communicate with rtpproxy via unix socket sucessfully.
The problem is when i call from ATA to quintum the voice path is one way??????????/
My SER.CFG and RTPPROXYLOG are as follows:
#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#
# ----------- 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/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/nathelper.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)
# ------------------------- 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) {
log(1, "REQUEST FOR SERVED DOMAIN------------------");
if (uri=~"^sip:3000@ 62.220.101.2"){ *********************SER VALID IP ADDRESS***********************
rewritehost("62.220.101.3");****************************QUINTUM VALID IP ADDRESS**********************
force_rtp_proxy();
log(1, "-------MEDIA IS BEINGED PROXIED--------------");
forward(62.220.101.3, 5060);********************************QUINTUM VALID IP ADD**************************
break;
};
#we forwad to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
++++++++++++++++++++++++++++++++++++++++++++++++++++RTYPROXY DEBUG LOG+++++++++++++++++++++++++++++++++++++++
# rtpproxy -2f -l 62.220.101.2 -s unix:/var/run/rtpproxy.sock -t 40
rtpproxy: rtpproxy started, pid 18398
rtpproxy: new session 633554326(a)62.220.101.8, tag 3269848300 requested
rtpproxy: new session on a port 35000 created, tag 3269848300
rtpproxy: pre-filling caller's address with 62.220.101.8:16384
rtpproxy: session timeout
rtpproxy: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped
rtpproxy: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped
rtpproxy: session on ports 35000/0 is cleaned up
Appreciate any suggections
mohammad
Hi,
when I make a call from UserA to UserB and UserB hangs up first, the Sip-To-Tag and Sip-From-Tag is switching place in
radius stop report compared to the start report. This only seems to happen with calls between "on-net" phones. If the
call is routed to the gw, this does not happen. Does anyone know if this is normal behavior? I'm using ser 0.8.12 and
radiusclient 0.3.2.
Regards,
Tor
Can you try Asterisk calling card platform? I use SER as a proxy server, all
phones connected to it and uses asterisk as a gateway to the pstn and GSM
network. So far the Asterisk calling card works fine.
http://www.voip-info.org/tiki-print.php?page=ASTCC
Does the "ser-0.8.18.src.tar.gz" distribution includes "mysql"
related modules ? I just can't find "mysql.so" in
usr/local/lib/ser/modules.
Any help would be appreciated !