Hi!
I just came along a problem in the following scenario: user with public
IP calls a NATed user. The first INVITE causes a lookup("location")
which also sets the natflag (usually flag 6).
Then I check if flag 6 is set and will force the RTP proxy if it is set.
If the public client sends an in-dialog reINVITE, the reINVITE is
processed by the loose_route block and there is no lookup("location")
for those request. So, how can I find out that the request will be
forwarded to a NATed user (to force the RTP proxy)?
regards,
Klaus
Hello,
Two times in the past week I have had some issues with the bindings of the
sip mapping expiring:
Binding 'user','sip:user@x.x.x.x:5060' has expired
Does anyone know what this is and why it happens? I appreciate any input
you may have.
Thanks,
Hekate
_________________________________________________________________
All the action. All the drama. Get NCAA hoops coverage at MSN Sports by
ESPN. http://msn.espn.go.com/index.html?partnersite=espn
Aliases have impact only on incoming calls, they don't mangle
outgoing signaling. They just rewrite r-uri, that's all they do.
To set callerid down in pstn, use rpid.
-jiri
At 05:42 PM 3/22/2004, Ralph.Wabel(a)swisscom.com wrote:
>Content-class: urn:content-classes:message
>Content-Type: multipart/alternative;
> boundary="----_=_NextPart_001_01C4102C.A43065FC"
>
>Hi,
>
>I have the problem that when I make an outgoing call over a Cisco Gateway to a PSTN phone. I�ve defined an alias for the user, works fine for incoming calls, but for the outgoing calls it shows always the number of the main number of the number block. When I make a debug I see that it goes out with the username instead of the alias. Here is my ser.cfg, maybe I�ve done something wrong in the config file. If someone could help me it would be great. Let me know if the config from the Cisco Gateway is also important.
Jan,
Any ideas on this?
Scott Morris
Enterprise Network Engineer
DOE - ORAU / ORISE
865-576-4672
-----Original Message-----
From: Morris, Scott
Sent: Thursday, March 18, 2004 2:38 PM
To: serusers(a)lists.iptel.org
Subject: [Serusers] Radius client - ldconfig not adding the libarry
I installed the radiusclient 0.4.1, and ran into a slight problem.
I run ldconfig -v /usr/local/lib and the following is output:
/usr/local/lib:
libradiusclient.so.2 -> libradiusclient.so.2.0.0
Here is the directory of /usr/local/lib
-rw-r--r-- 1 root root 130780 Mar 18 12:37
libradiusclient.a
-rwxr-xr-x 1 root root 853 Mar 18 12:37
libradiusclient.la
lrwxrwxrwx 1 root root 24 Mar 18 12:37
libradiusclient.so -> libradiusclient.so.2.0.0
lrwxrwxrwx 1 root root 24 Mar 18 12:37
libradiusclient.so.2 -> libradiusclient.so.2.0.0
-rwxr-xr-x 1 root root 95088 Mar 18 12:37
libradiusclient.so.2.0.0
Ldconfig will not process the libradiusclient.so because it is a symbolic
link. So when I run "ser -ddddddddd start" I get the following:
0(7710) ERROR: load_module: could not open module
</usr/lib/ser/modules/auth_radius.so>: libradiusclient.so.0: cannot open
shared object file: No such file or directory
Ser can't locate the reference for libradiusclient.so.0 since ldconfig won't
process the link libradius.so.0. Any ideas of getting around this. I
checked the man pages for ldconfig, and it states it does not process
symbolic links:
"ldconfig ignores symbolic links when scanning for libraries."
Scott Morris
Enterprise Network Engineer
DOE - ORAU / ORISE
865-576-4672
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
I change serweb php files, add lookup("aliases") to /etc/ser/ser.cfg
it's ok but I have a problem with my user account page
Warning: Invalid argument supplied for foreach() in
/var/www/virtual/02/html/page.php on line 158
Warning: Invalid argument supplied for foreach() in
/var/www/virtual/02/html/page.php on line 161
##################################################
foreach($config->enable_tabs as $i => $value)
if ($value) $lasttab=$i;
foreach($config->enable_tabs as $i => $value){
if ($value){
####################################################
Harry
Le lun 22/03/2004 à 18:31, Atle Samuelsen a écrit :
> Without any config files and error messages it's hard to know what the
> exact problem is.. but if I guess right, your fifo buffer
> (/tmp/ser_fifo) has the wrong chmod. try to set fifo_mode in your ser
> config.
>
> - AtlE
> * gaillac harry <gaillacharry(a)yahoo.fr> [040322 17:13]:
> > hello,
> >
> > i have a problem with serweb registration. my apache server is
> > configured with User apache Group root because of /tmp/ser_fifo
> > permissions but when i try to confirme my account ser or apache failed,
> > nothing is going on ????
> >
> > Harry
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
ok great let me check it
>From: Klaus Darilion <klaus.mailinglists(a)pernau.at>
>To: Kapil Dhawan <sersavvy(a)hotmail.com>,Serusers <serusers(a)lists.iptel.org>
>Subject: Re: [Serusers] SER Notify
>Date: Mon, 22 Mar 2004 16:17:16 +0100
>
>You can use the OPTIONS method to find out if a user is available or not.
>
>More details can be found in the SIP standard: RFC 3261
>
>regards,
>klaus
>
>Kapil Dhawan wrote:
>>na its not clear to me..pls write atleast a line or two
>>
>>
>>>From: Klaus Darilion <klaus.mailinglists(a)pernau.at>
>>>To: Kapil Dhawan <sersavvy(a)hotmail.com>
>>>CC: serusers(a)lists.iptel.org
>>>Subject: Re: [Serusers] SER Notify
>>>Date: Mon, 22 Mar 2004 15:38:34 +0100
>>>
>>>not NOTIFY, but OPTIONS.
>>>
>>>Klaus
>>>
>>>Kapil Dhawan wrote:
>>>
>>>>is there any way i can check the status of a user by sending some
>>>>request from the server....like some NOTIFY kid of thing ...and come to
>>>>know the status of user whether he is busy with some call or anything
>>>>like......
>>>>
>>>>regards
>>>>
>>>>_________________________________________________________________
>>>>Contact brides & grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag
>>>>Only on www.shaadi.com. Register now!
>>>>
>>>>_______________________________________________
>>>>Serusers mailing list
>>>>serusers(a)lists.iptel.org
>>>>http://lists.iptel.org/mailman/listinfo/serusers
>>>>
>>>>
>>>
>>
>>_________________________________________________________________
>>Easiest Money Transfer to India. Send Money To 6000 Indian Towns.
>>http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home!
>>
>>
>
_________________________________________________________________
Easiest Money Transfer to India. Send Money To 6000 Indian Towns.
http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home!
Hi,
I have the problem that when I make an outgoing call over a Cisco
Gateway to a PSTN phone. I've defined an alias for the user, works fine
for incoming calls, but for the outgoing calls it shows always the
number of the main number of the number block. When I make a debug I see
that it goes out with the username instead of the alias. Here is my
ser.cfg, maybe I've done something wrong in the config file. If someone
could help me it would be great. Let me know if the config from the
Cisco Gateway is also important.
# $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=no
#log_stderror=yes # (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"
fifo_mode=0666
alias="swissptt.ch"
alias="inoitasip.swissptt.ch"
# ------------------ 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"
"/etc/ser/ser.cfg" 138L, 3639C
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;
};
# attempt handoff to PSTN
if (uri=~"^sip:0[0-9]*@inoitasip.swissptt.ch") { ## This
assumes that the caller is
log("Forwarding to PSTN\n"); ## registered in our
realm
t_relay_to_udp("193.5.228.202", "5060"); ## Our Cisco
router
break;
forward( 193.5.228.202, 5060 );
break;
forward( 193.5.228.202, 5060 );
};
# 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=~inoitasip.swissptt.ch) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("inoitasip.swissptt.ch",
"subscriber")) {
www_challenge("inoitasip.swissptt.ch",
"0");
break;
};
save("location");
break;
};
#needed for alias
lookup("aliases");
# 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();
break;
forward( 193.5.228.202, 5060 );
};
# 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=~inoitasip.swissptt.ch) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("inoitasip.swissptt.ch",
"subscriber")) {
www_challenge("inoitasip.swissptt.ch",
"0");
break;
};
save("location");
break;
};
#needed for alias
lookup("aliases");
# 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();
};
}
Thanks
Ralph
hello,
i have a problem with serweb registration. my apache server is
configured with User apache Group root because of /tmp/ser_fifo
permissions but when i try to confirme my account ser or apache failed,
nothing is going on ????
Harry