Hi all
I need a little help with that.
I have installation of ser-0.8.14 and freeradius1.02.
I am checking my register requests with ngrep and it's coming on port
5060 with no problem. The problem is authentication, I can't
authenticate users through radius, freeradius working properly i
checked that with radiusclient, but the register request is not going
through authentication in the radius.( I don't see anything happens in
the radius logs)
If there any way to debug the ser ( i have debug=9 inside ser.cfg). In
order to see what's happening when the request is coming, and if it's
going to the radius or not.
ser.cfg
-----------------------------------
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_radius.so"
--------------------
modparamd"auth_radius",
"radius_config","/usr/local/etc/radiusclient/radiusclient.conf")
modparam("auth_radius", "service_type", 15)
----------------------
if (method=="REGISTER") {
log(1, "REGISTER: Authenticating user\n");
if (!radius_www_authorize("")) {
log(1, "REGISTER: challenging user\n");
www_challenge("", "0");
break;
};
setflag(1);
save("location");
sl_send_reply("200","ok");
break;
};
------------------------
Thanks for any help.
Why do i receive these?
Apr 14 15:35:47 max ser[60541]: record_route(): Double attempt to
record-route
Apr 14 15:35:47 max /kernel: Apr 14 15:35:47 max ser[60541]: record_route():
Double attempt to record-route
Apr 14 15:35:47 max ser[60541]: ERROR: send_rtpp_command: can't connect to
RTP proxy
Apr 14 15:35:47 max /kernel: Apr 14 15:35:47 max ser[60541]: ERROR:
send_rtpp_command: can't connect to RTP proxy
Apr 14 15:35:47 max ser[60541]: WARNING: rtpp_test: can't get version of the
RTP proxy
Apr 14 15:35:47 max ser[60541]: WARNING: rtpp_test: support for RTP proxy
has been disabled temporarily
Apr 14 15:35:47 max ser[60541]: ERROR: force_rtp_proxy2: support for RTP
proxy is disabled
Apr 14 15:35:47 max /kernel: Apr 14 15:35:47 max ser[60541]: ERROR:
force_rtp_proxy2: support for RTP proxy is disabled
Apr 14 15:37:35 max ser[60537]: record_route(): Double attempt to
record-route
Apr 14 15:37:35 max /kernel: Apr 14 15:37:35 max ser[60537]: record_route():
Double attempt to record-route
Apr 14 15:37:35 max ser[60537]: ERROR: send_rtpp_command: can't connect to
RTP proxy
Apr 14 15:37:35 max /kernel: Apr 14 15:37:35 max ser[60537]: ERROR:
send_rtpp_command: can't connect to RTP proxy
Apr 14 15:37:35 max ser[60537]: WARNING: rtpp_test: can't get version of the
RTP proxy
Apr 14 15:37:35 max ser[60537]: WARNING: rtpp_test: support for RTP proxy
has been disabled temporarily
Apr 14 15:37:35 max ser[60537]: ERROR: force_rtp_proxy2: support for RTP
proxy is disabled
Apr 14 15:37:35 max /kernel: Apr 14 15:37:35 max ser[60537]: ERROR:
force_rtp_proxy2: support for RTP proxy is disabled
Hello,
I am a new SER user, and would like to use this application for carrier
class VoIP services. I've got SER to process calls fine using a simple
routing list, but am having some trouble setting up authentication.
I've tried to look through prior mailings but did not see this error
come up.
When trying to add a user using the command "serctl add 7778881000
7778881000 7778881000@localhost", I get "error: 500 command
'ul_show_contact' not available" after entering the mysql password
(doesn't matter if it's correct or not and from the logs I don't think
SQL is even playing a part here yet.)
The debug messages says:
1(12965) ERROR: fifo_server: command ul_show_contact is not available
1(12965) qm_free(e55c8, 102ea8), called from fifo_server.c:
fifo_server(547)
1(12965) qm_free: freeing frag. 102e90 alloc'ed from fifo_server.c:
trim_filename(311)
1(12965) **** done consume
1(12965) ERROR: fifo_server: command must begin with ::
7778881000(a)lwav.net
1(12965) **** done consume
I've tried adding a user called 7778881000(a)lwav.net too but it doesn't
seem to matter (lwav.net is my SIP_DOMAIN variable.)
Here's my SER version info in case that matters:
version: ser 0.9.0 (sparc64/solaris)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST,
DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_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 1.197 2004/12/03 19:09:31 andrei Exp $
main.c compiled on 15:00:54 Apr 6 2005 with gcc 3.4
Thanks,
Brian
when I exec serctl moni show this in syslog
/usr/sbin/ser[30466]: ERROR: open_reply_pipe: open error
(/tmp/ser_receiver_31325): Permission denied
/usr/sbin/ser[30466]: ERROR: fifo_reply: no reply pipe /tmp/ser_receiver_31325
anyone know why ?
--
[]s,
Bruno "Niggas" Oliveira
Belo Horizonte - MG
Msn: n1gg4s(a)gmail.com
"Todo o nosso descontentamento por aquilo
que nos falta procede da nossa falta de
gratidão por aquilo que temos."
hi
when I type the command serctl moni only appears the line
"[cycle #: 1; if constant make sure server lives and fifo is on]"
this is normal? how I arrange this?
--
[]s,
Bruno "Niggas" Oliveira
Belo Horizonte - MG
Msn: n1gg4s(a)gmail.com
"Todo o nosso descontentamento por aquilo
que nos falta procede da nossa falta de
gratidão por aquilo que temos."
Hi ,
could anybody tell me what is the reason why I get the
error message "Sorry, there was an error when sending
mail. Please try again later" when trying to register
a user with serweb? All user details entered in user
management form are stored in "pending" table because
of unsuccesfull emailing.
Thank you in advance,
Nicusor
__________________________________
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Hi,
I have the following problem.
A messenger 5.1 connected to an LCS server 2005. A SIP phone connected to SER.
When I call from messenger to SIP phone I have the voice only from SIP phone to messenger and not other way.
There is not RTP packet send from messenger.
If I declare messenger to as SER user, it's working fine.
Has anybody an hand for me?
Alexis
---------------------------------
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Thanks Paul.
Now I got
sipsak -f bye.txt -s sip:1.2.3.4
** give up retransmissioning....
And the call continued :(
Regards,
Pavel
----- Original Message -----
From: Java Rockx
To: Pavel Siderov
Cc: serusers(a)lists.iptel.org
Sent: Wednesday, April 13, 2005 4:29 PM
Subject: Re: [Serusers] sending bye using sipsak [onsip.org]
Your sipsak command should be
sipsak -f bye.txt -s sip:1.2.3.4
Regards,
Paul
On 4/13/05, Pavel Siderov <pi(a)hostmates.com> wrote:
Hi guys,
I read the tutorial on http://onsip.org ( http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#2 ) about sending byes but has no success with my tries:
bye.txt:
BYE sip:35923456@1.2.3.4 SIP/2.0
From: sip:222485@5.6.7.8;user=phone
To: sip:35923456@1.2.3.4:5060;user=phone
Contact: sip:222485@5.6.7.8;user=phone
CSeq: 100 BYE
Call-ID: 63450d656cbd4f55bec37cbfeb50a181(a)192.168.2.111
Max-Forwards: 16
Content-Length: 0
Call-ID is taken after call is has started. 1.2.3.4 is the ip of my pstn provider and 5.6.7.8 of my ser box.
When I try to send it using sipsak:
sipsak -f bye.txt -s 1.2.3.4
I got the following error:
error: SIPURI doesn't not begin with sip
Any suggestions / ideas what I'm doing wrong.
Thanks,
Pavel
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Does Microsoft Messenger 5.1 have a problem with presence updates? I get
updates for 4.6 clients but not for 5.1.
ser.cfg is:
cat ser.cfg
#
# $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 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=4
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"
alias="xxx.com"
alias="xxx.local"
alias="sip.xxx.local"
# ------------------ 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"
# 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", 1)
# -- 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)
# -- registrar parameters
modparam("registrar","default_expires",1800)
modparam("registrar","use_domain",0)
# ------------------------- 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("iptel.org", "subscriber")) {
# www_challenge("iptel.org", "0");
# break;
# };
log(1,"request for register received");
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();
};
}
Regards,
Nigel
Hi guys,
I read the tutorial on http://onsip.org ( http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#2 ) about sending byes but has no success with my tries:
bye.txt:
BYE sip:35923456@1.2.3.4 SIP/2.0
From: sip:222485@5.6.7.8;user=phone
To: sip:35923456@1.2.3.4:5060;user=phone
Contact: sip:222485@5.6.7.8;user=phone
CSeq: 100 BYE
Call-ID: 63450d656cbd4f55bec37cbfeb50a181(a)192.168.2.111
Max-Forwards: 16
Content-Length: 0
Call-ID is taken after call is has started. 1.2.3.4 is the ip of my pstn provider and 5.6.7.8 of my ser box.
When I try to send it using sipsak:
sipsak -f bye.txt -s 62.244.175.133
I got the following error:
error: SIPURI doesn't not begin with sip
Any suggestions / ideas what I'm doing wrong.
Thanks,
Pavel