Hi,
I have a strange problem with my ser router, which already worked
correctly since a few weeks:
I cannot dial any address without the prefix "sip:" and the suffix
"@mydomain.com" anymore.
For example, when I just type in "echo" instead of
"sip:echo@mydomain.com", I get an "not found" error.
When I dial a no. through my ISDNGW, i.e. "1004" instead of
"sip:1004@mydomain.com", I get an "Interplanetary Toodle Noodle" error.
Does anyone know a reason for this ?
Oliver
Hello!
I'm trying to setup SER to forward calls to a cisco pstn (AS53xx), and have some
dificults.
Well, I'm get confused about Ser configs to do this, so maibe somebody can help-me!
I have some clients, using Cisco ATA-186 trying to logging on my Ser server, and
I want to forward the calls to our PSTN, across dial plan.
Now I have a Ser working and registering clients (I think), but I can't do Ser
to interact with Cisco,
How can I do configure the ser.cfg, maibe someone ser.cfg with samples will be
helpful.
Regards.
--
|o
|o
|o Fabio Silvestri
|o fabio(a)informatec.com.br
|o ICQ: 1667351
|o
Phone im trying to call is a regular phone hooked up to a PBX then PBX is
connected to a cisco router (acting as sip gateway), so the phone cannot
register with the server. How can I get around it as x-lite will only call
number(a)sipserv.foo.com. But only calls are being forwared to cisco gateway
is number(a)foo.com (dunno why). Please look at my ser.cfg and see what I can
change so calls to number(a)sipserv.foo.com will also be forwaded to cisco
gateway.
Thanks
Andy
-----Original Message-----
From: Jan Janak [mailto:jan@iptel.org]
Sent: Wednesday, January 07, 2004 6:39 PM
To: Andy Singh
Cc: 'serusers(a)lists.iptel.org'
Subject: Re: [Serusers] dialing PSTN using x-lite
404 means that the phone you are trying to call is not registered on the
server. Make sure that you have proper domain in subscriber table
(sipserv.foo.com and not just foo.com).
Jan.
On 06-01 15:21, Andy Singh wrote:
> Hello all,
>
> My sip domain is sipserv.foo.com, i have a user1(a)sipserv.foo.com. i can
log
> in fine via messenger and via x-lite 2.0, but when i dial a phone number
> let's say 1212(a)sipserv.foo.com i immediatly get 404 not found. but i can
> dial 1212(a)foo.com from windows messenger, since i don't have the option
to
> specify just @foo.com in x-lite it always dials
> 1212(a)sipserv.foo.com. How can i make x-lite dial differently or how can i
> make sipserv.foo.com work. Here's my ser.cfg file
>
>
> #
> # $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=yes # (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/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", 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("sipserv.foo.com", "subscriber")) {
> www_challenge("sipserv.foo.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();
> };
> attempt handoff to PSTN
> if (uri=~"^sip:3[0-9]*") { ## This assumes that the caller is
> log("Forwarding to PSTN\n"); ## registered in our realm
> forward( 156.151.96.253, 5060 ); ## Our Cisco router
> break;
> };
>
> }
>
>
> Please help
>
> Thanks
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
You can use the same utility and create an alias.
Jan.
On 07-01 07:22, Nadav Baron wrote:
> Hello,
>
> Thank you very much for your helpful and quick reply.
> I am not using "serweb" but only "serctl" and MySql
> utilities to control users. When using MySql to see
> the ser database, I can see that there is an admin
> acount with phone "123". I can then generate a new
> acount with "serctl add Joe password". However that
> does not create a phone number for new user Joe.
> How can I create a phone number for Joe? Is it with
> MySql?
>
> Thank you for your help
>
>
>
> --- Jan Janak <jan(a)iptel.org> wrote:
> > By default, when the account is created through
> > serweb, the user gets a
> > number alias automatically. There is some initial
> > value (82000 if my
> > memory serves), the first registered user will get
> > this number, the
> > second will get 82001 and so on, so forth.
> >
> > Jan.
> >
> > On 05-01 13:51, Nadav Baron wrote:
> > > Hello,
> > >
> > > I am trying to set up public sip server using SIP
> > > Express router. I have installed SER and it seems
> > to
> > > work fine. I can add and remove new users using
> > the
> > > "serctl" utilit however I have not control on the
> > sip
> > > numbers. That is, when a new user signs up, how
> > can I
> > > assign a sip number so that calling to this number
> > > will be directed to that user.
> > >
> > > Thank you
> > >
> > > __________________________________
> > > Do you Yahoo!?
> > > Find out what made the Top Yahoo! Searches of 2003
> > > http://search.yahoo.com/top2003
> > >
> > > _______________________________________________
> > > Serhelp mailing list
> > > serhelp(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serhelp
>
>
> __________________________________
> Do you Yahoo!?
> Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes
> http://hotjobs.sweepstakes.yahoo.com/signingbonus
Hello all,
My sip domain is sipserv.foo.com, i have a user1(a)sipserv.foo.com. i can log
in fine via messenger and via x-lite 2.0, but when i dial a phone number
let's say 1212(a)sipserv.foo.com i immediatly get 404 not found. but i can
dial 1212(a)foo.com from windows messenger, since i don't have the option to
specify just @foo.com in x-lite it always dials
1212(a)sipserv.foo.com. How can i make x-lite dial differently or how can i
make sipserv.foo.com work. Here's my ser.cfg file
#
# $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=yes # (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/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", 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("sipserv.foo.com", "subscriber")) {
www_challenge("sipserv.foo.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();
};
attempt handoff to PSTN
if (uri=~"^sip:3[0-9]*") { ## This assumes that the caller is
log("Forwarding to PSTN\n"); ## registered in our realm
forward( 156.151.96.253, 5060 ); ## Our Cisco router
break;
};
}
Please help
Thanks
Hi,
is there any way to get the mysql database modul running with the current
CVS version of SER? If not, any pointers on what i need to do to patch
the modul so it will work with the current CVS version?
regards,
Arnd
--
NetHead Network Design and Security
Arnd Vehling av(a)nethead.De
Gummersbacherstr. 27 Phone: +49 221 8809210
50679 Cologne, Germany Fax : +49 221 8809212
Hi,
Can anyone help? I want to get from the URI the user value and insert into
the <rewrite("sip:user@domain")> command in the ser.cfg. Please
help...Thanks.
Regards,
Shirley
Hi!
Does anybody knows what's ideal configuration for setup SER with ata186 inside NAT?
When I dial for a other ata186 the other side can hear me but I can't hear
anything...
--
|o
|o
|o Fabio Silvestri
|o fabio(a)informatec.com.br
|o ICQ: 1667351
|o
