Hi!
> -----Original Message-----
> From: Nils Ohlmeier [mailto:nils@iptel.org]
> Sent: Thursday, December 04, 2003 1:39 PM
> To: franz.edler(a)utanet.at; serusers(a)lists.iptel.org
> Subject: Re: [Serusers] Problem creating mysql database
>
>
> Hello,
> On Thursday 04 December 2003 11:13, Franz Edler wrote:
> > Hello,
...
> > MySQL server is running.
> > I called /usr/sbin/ser_mysql.sh.
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> > But the mySQL management tool command "SHOW DATABASES"
> shows only two
> > databases mySQL and test, where I also expected ser.
> > When I issue the mySQL management tool command "SELECT *
> FROM USER" I get
> > the error "ERROR 1046: No database selected"
> >
> > Has anyone an idea what I can do?
> > Is the problem caused by the ignored inconsistencies of
> > libmysqlclient.so.1x?
>
> Sorry that i have to say: RTFM.
> It is no where mentioned that the installation of the rpm
> also creates the
> databases. That would IMHO often end in a real disaster.
> After you installed the rpm's you have to create the required
> database with
> the ser_mysql.sh script.
To me it looks like as Franz already started the script.
I also suggest compiling from source - never had any problems, and you
never have conflicts because of wrong library versions.
Klaus
Let me rephrase. I don't think it is SER. And I do agree it is Messenger.
When UserA signs out, the update is immediately seen by UserB that UserA is
offline. So, where is the update failing?
Scott Morris
Enterprise Network Engineer
DOE - ORAU / ORISE
865-576-4672
-----Original Message-----
From: Morris, Scott
Sent: Thursday, December 04, 2003 8:29 AM
To: 'Mario Kolberg'; serusers(a)lists.iptel.org
Subject: RE: [Serusers] Windows Messenger 5.0
I can't say it's a messenger issue. Messenger wants a contact list, and to
show you who is online and offline. Your phone doesn't do that. The phone
talks to SER to see if the person is signed in. Or does kphone and sipc
show users signed in?
Scott Morris
Enterprise Network Engineer
DOE - ORAU / ORISE
865-576-4672
-----Original Message-----
From: Mario Kolberg [mailto:mko@cs.stir.ac.uk]
Sent: Thursday, December 04, 2003 5:47 AM
To: Morris, Scott; serusers(a)lists.iptel.org
Subject: Re:[Serusers] Windows Messenger 5.0
I have made exactly the same observation. I treat it as a problem with
Messenger as it works fine between two kphone UAs or between sipc and
kphone. I'm running ser 0.8.12.
Mario
> I have 0.8.1 running on Suse 9.0. It works great with my Zultys
> phone, Polycom IP 600, and the Helmsen agent. When I try to use
> Windows Messenger 5.0 I have a slight problem.
>
> UserA signs in. UserB signs in, and sees that UserA is online, and
> there changes the UserA in the contact list on ONLINE. But UserB
> never shows as ONLINE on UserA's desktop. So UserA can't call UserB.
> The update never goes to UserA's desktop. What am I missing here?
--
The University of Stirling is a university established in Scotland by
charter at Stirling, FK9 4LA. Privileged/Confidential Information may be
contained in this message. If you are not the addressee indicated in this
message (or responsible for delivery of the message to such person), you may
not disclose, copy or deliver this message to anyone and any action taken or
omitted to be taken in reliance on it, is prohibited and may be unlawful.
In such case, you should destroy this message and kindly notify the sender
by reply email. Please advise immediately if you or your employer do not
consent to Internet email for messages of this kind.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Hello,
I am using SuSE Linux 9.0 and try to run SER 0.8.11.
SER is now already running and the next step for me is to add the mySQL
database.
During installation of ser_mysql package I ran into a package inconsistency
because SER 0.8.11 requires libmysqlclient.so.10 and SuSE Linux 9.0 uses
libmysqlclient.so.12. Full of hope I ignored the warning and the situation
is now the following:
MySQL server is running.
I called /usr/sbin/ser_mysql.sh.
But the mySQL management tool command "SHOW DATABASES" shows only two
databases mySQL and test, where I also expected ser.
When I issue the mySQL management tool command "SELECT * FROM USER" I get
the error "ERROR 1046: No database selected"
Has anyone an idea what I can do?
Is the problem caused by the ignored inconsistencies of
libmysqlclient.so.1x?
Franz
Hi,
I am looking for a working ser.cfg which forwards all unanswered calls
to voicemail after 10 seconds..
How is the correct usage of "modparam("tm", "fr_inv_timer", 8 )" ??
I tried to integrate the sample voicemail configuration from iptel's ser
admin guide, but now ALL my calls are forwarded to voicemail...
I have read that it can be useful to run 2 instances of ser to integrate
voicemail. Is this necessary ???
Oliver
I have made exactly the same observation. I treat it as a problem with
Messenger as it works fine between two kphone UAs or between sipc and
kphone. I'm running ser 0.8.12.
Mario
> I have 0.8.1 running on Suse 9.0. It works great with my Zultys phone,
> Polycom IP 600, and the Helmsen agent. When I try to use Windows Messenger
> 5.0 I have a slight problem.
>
> UserA signs in. UserB signs in, and sees that UserA is online, and there
> changes the UserA in the contact list on ONLINE. But UserB never shows as
> ONLINE on UserA's desktop. So UserA can't call UserB. The update never
> goes to UserA's desktop. What am I missing here?
--
The University of Stirling is a university established in Scotland by
charter at Stirling, FK9 4LA. Privileged/Confidential Information may
be contained in this message. If you are not the addressee indicated
in this message (or responsible for delivery of the message to such
person), you may not disclose, copy or deliver this message to anyone
and any action taken or omitted to be taken in reliance on it, is
prohibited and may be unlawful. In such case, you should destroy this
message and kindly notify the sender by reply email. Please advise
immediately if you or your employer do not consent to Internet email
for messages of this kind.
hi,
When i turn on log_strerror=yes, I can't not register my SIP UE with SER.
When i comment out "listen" and fork, I can register; When i only comment out listen, it also can not register.
Can anybody take a look ?
Thanks,
hong
--------------------------
debug=4 # debug level (cmd line: -dddddddddd)
listen=192.168.8.96
fork=no
log_stderror=yes # (cmd line: -E)
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"
I tried using MSN 5.0 earlier. And I faced exactly the same problem. After sniffing the network traffic I came to know that -
When UserA sends a SUBSCRIBE message to UserB, if UserB has UserA in its list, it sends back a OK message. However it is not smart enough to update the status of UserA (to online) on its own list. It just waits for its turn to send the SUBSCRIBE message to UserA and then only if it gets an OK response back from UserA will it update the status of UserA on its list. So if you let UserA and UserB to be both online for a certain duration of time ( == time between two SUBSCRIBE attempts atleast), you will see the correct status. Another stupid MS implememtation.
Dhiraj
-----Original Message-----
From: Jan Janak [mailto:jan@iptel.org]
Sent: 04 December 2003 00:13
To: Morris, Scott
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] Windows Messenger 5.0
That is probably because of the instant messaging and presence model
implemented in MS Messenger.
When you UserA signs in, his messenger sends a SUBSCRIBE to UserB to
find out his status (online/offline). Because UserB didn't sign in yet,
the SUBSCRIBE message will be not replied and UserA thinks that UserB is
offline.
Messenger of UserA tries to resend the SUBSCRIBE again later, but
meanwhile you see UserB offline (even if he signs in during this
period).
After some time (I don't know how long) messenger of UserA should retry
and then you will see UserB online.
Jan.
On 03-12 16:24, Morris, Scott wrote:
> I have 0.8.1 running on Suse 9.0. It works great with my Zultys phone,
> Polycom IP 600, and the Helmsen agent. When I try to use Windows Messenger
> 5.0 I have a slight problem.
>
> UserA signs in. UserB signs in, and sees that UserA is online, and there
> changes the UserA in the contact list on ONLINE. But UserB never shows as
> ONLINE on UserA's desktop. So UserA can't call UserB. The update never
> goes to UserA's desktop. What am I missing here?
>
> Scott Morris
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
I am forwarding this message so the rest of the list can benefit as well.
this is what my ngrep looks like now: It appears that the phone is trying to register, but the ser server is saying No!
How can I fix this?
Thanks a whole lot!
Chris
U 192.168.3.201:50207 -> 192.168.3.1:5060
REGISTER sip:192.168.3.1 SIP/2.0..Via: SIP/2.0/UDP 192.168.3.201:5060..From: sip:chris@192.168.3.1..To: sip:chris@192.168
.3.1..Call-ID: 00036b3c-2e330026-71ef1fc6-7d9d0be3@192.168.3.201..CSeq: 101 REGISTER..User-Agent: CSCO/4..Contact: sip:ch
ris@192.168.3.201:5060..Content-Length: 0..Expires: 3600....
#
U 192.168.3.1:5060 -> 192.168.3.201:5060
SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 192.168.3.201:5060..From: sip:chris@192.168.3.1..To: sip:chris@192.168.3.1;tag
=b27e1a1d33761e85846fc98f5f3a7e58.ff9c..Call-ID: 00036b3c-2e330026-71ef1fc6-7d9d0be3@192.168.3.201..CSeq: 101 REGISTER..W
WW-Authenticate: Digest realm="sip.telepacket.com", nonce="3fce89df87947262e59f6714c788b940428cf2ab"..Server: Sip EXpress
router (0.8.12 (i386/linux))..Content-Length: 0..Warning: 392 192.168.3.1:5060 "Noisy feedback tells: pid=1081 req_src_
ip=192.168.3.201 req_src_port=50207 in_uri=sip:192.168.3.1 out_uri=sip:192.168.3.1 via_cnt==1"....
#
U 192.168.3.201:50207 -> 192.168.3.1:5060
REGISTER sip:192.168.3.1 SIP/2.0..Via: SIP/2.0/UDP 192.168.3.201:5060..From: sip:chris@192.168.3.1..To: sip:chris@192.168
.3.1..Call-ID: 00036b3c-2e330026-71ef1fc6-7d9d0be3@192.168.3.201..CSeq: 102 REGISTER..User-Agent: CSCO/4..Contact: sip:ch
ris@192.168.3.201:5060..Authorization: Digest username="LINE 1",realm="sip.telepacket.com",uri="sip:192.168.3.1",response
="2146fa3e14386bf104a407f57e879d81",nonce="3fce89df87947262e59f6714c788b940428cf2ab",algorithm=md5..Content-Length: 0..Ex
pires: 3600....
#
U 192.168.3.1:5060 -> 192.168.3.201:5060
SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 192.168.3.201:5060..From: sip:chris@192.168.3.1..To: sip:chris@192.168.3.1;tag
=b27e1a1d33761e85846fc98f5f3a7e58.ff9c..Call-ID: 00036b3c-2e330026-71ef1fc6-7d9d0be3@192.168.3.201..CSeq: 102 REGISTER..W
WW-Authenticate: Digest realm="sip.telepacket.com", nonce="3fce89df87947262e59f6714c788b940428cf2ab"..Server: Sip EXpress
router (0.8.12 (i386/linux))..Content-Length: 0..Warning: 392 192.168.3.1:5060 "Noisy feedback tells: pid=1083 req_src_
ip=192.168.3.201 req_src_port=50207 in_uri=sip:192.168.3.1 out_uri=sip:192.168.3.1 via_cnt==1"....
############exit
-----Original Message-----
From: Jan Janak [mailto:jan@iptel.org]
Sent: Wed 12/3/2003 4:43 PM
To: Chris Monahan
Cc:
Subject: Re: [Serusers] help with the setup of Ser.
Most of them send REGISTER by default when you turn them on.
If it is Cisco 7960 then go to "SIP Configuration" menu and check that
the item "Register with Proxy" (number 10) is set to yes.
Jan.
On 03-12 16:41, Chris Monahan wrote:
> Here is my config. I am working on the ngrep now.
>
> How do I get my phones to send a REGISTER?
>
> #
> # $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/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
> alias="telepacket.com"
> alias="sip.telepacket.com"
> alias="192.168.3.1"
> 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 (uri=~"^sip:9[0-9]*@sip.telepacket.com") {
> # log("Forwarding to PSTN\n");
> #t_relay_to( "207.71.120.6", "5060");
> # break;
> #};
> if (!uri==myself) {
> route(1);
> break;
> };
> if (method=="REGISTER") {route(3); break; };
> lookup("aliases");
> if (!uri==myself) { 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=~"^sip:(.+@)?(192\.168\.3\.1|(sip\.)?telepacket\.com)([:;\?].*)?$") {
> if (uri==myself) {
> if (method=="REGISTER") {
> # Uncomment this if you want to use digest authentication
> if (!www_authorize("sip.telepacket.com", "subscriber")) {
> www_challenge("sip.telepacket.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();
> };
> }
>
>
> -----Original Message-----
> From: Jan Janak [mailto:jan@iptel.org]
> Sent: Wed 12/3/2003 4:08 PM
> To: Chris Monahan
> Cc: Kannaiyan Natesan; serusers(a)iptel.org
> Subject: Re: [Serusers] help with the setup of Ser.
>
>
>
> Hello,
>
> first of all, please remove the change suggested by Kannaiyan, it is
> wrong.
>
> To help you properly we would need to see your configuration file and
> SIP message dumps.
>
> You can create the message dumps using ngrep utility. Run
> "ngrep port 5060" on the SER server, it will show you all traffic coming to the
> server.
>
> First of all you must make your phones to send a REGISTER message. The
> message should be sent to SER server and the server should reply with
> 200 OK.
>
> Please send us the message dumps once you have them.
>
> Jan.
>
>
> On 03-12 15:31, Chris Monahan wrote:
> > well its doing something different now -- when I dial a call, it adds it to the location table in mysql. on the phone, it doesn't ring or anything, it just brings me right back to the WFC screen....
> >
> > Any ideas?
> >
> > -----Original Message-----
> > From: Kannaiyan Natesan [mailto:nkans@lycos.co.uk]
> > Sent: Wed 12/3/2003 3:29 PM
> > To: Chris Monahan; serusers(a)iptel.org
> > Cc:
> > Subject: Re: [Serusers] help with the setup of Ser.
> >
> >
> >
> > Sorry. that should be
> >
> > if( method =="REGISTER") {
> > save("location");
> > };
> >
> > Kannaiyan
> >
> >
> >
> > ----- Original Message -----
> > From: "Kannaiyan Natesan" <nkans(a)lycos.co.uk>
> > To: "Chris Monahan" <chris(a)loudpacket.com>; <serusers(a)iptel.org>
> > Sent: Wednesday, December 03, 2003 11:25 PM
> > Subject: Re: [Serusers] help with the setup of Ser.
> >
> >
> > > Remove the "if( uri==myself )" condition completely & add the following
> > >
> > > if( method == INVITE) {
> > > save("location");
> > > };
> > >
> > > this should register all the phones coming to your server. and you should
> > > able to call them without any problems.
> > >
> > > Kannaiyan
> > >
> > > http://www.speak2world.com -- Call with your own domain !!!
> > >
> > >
> > > ----- Original Message -----
> > > From: "Chris Monahan" <chris(a)loudpacket.com>
> > > To: <serusers(a)iptel.org>
> > > Sent: Wednesday, December 03, 2003 11:14 PM
> > > Subject: [Serusers] help with the setup of Ser.
> > >
> > >
> > > > Hello people...
> > > >
> > > > I have been working on Ser all day. I have read every manual possible,
> > and
> > > am about to lose it! My setup is pretty simple. Ser is running on
> > > 192.168.3.1... it is a dhcp server, tftp server, and ser server.
> > > >
> > > > I have 2 Cisco IP Phones (7960)...
> > > >
> > > > They both connect to 192.168.3.1 just fine. They download the latest
> > rom,
> > > they have the appropriate lines in the top right of the screen.
> > > >
> > > > Everything seems fine, until I attempt to make a call.
> > > >
> > > > Running serctl monitor shows me that everytime a call is placed, it gets
> > > an error 404 - Not Found. I was getting error 483 earlier, but I read
> > the
> > > documentation some more and fixed it.
> > > >
> > > > I have setup locations for the two users (sip:chris@192.168.3.1) and
> > > (sip:allan@192.168.3.1), and aliases as well..
> > > >
> > > > Still, I cannot make a call between the two phones. Everytime I try, I
> > get
> > > a busy tone on the phone, and error 404...
> > > >
> > > > Can anyone please help?
> > > >
> > > > Thanks!
> > > >
> > >
> > >
> > > --------------------------------------------------------------------------
> > --
> > > ----
> > >
> > >
> > > > _______________________________________________
> > > > Serusers mailing list
> > > > Serusers(a)iptel.org
> > > > http://mail.iptel.org/mailman/listinfo/serusers
> > > >
> > >
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > Serusers(a)iptel.org
> > > http://mail.iptel.org/mailman/listinfo/serusers
> > >
> >
> >
> >
> >
>
> > _______________________________________________
> > Serusers mailing list
> > Serusers(a)iptel.org
> > http://mail.iptel.org/mailman/listinfo/serusers
>
>
>
Hi All,
I am new to SER and have only recently installed and ran the
application. I have a cisco gateway which I forward to for call
termination and an ATA 186 to use as my sip client. I feel the problem is
with my dial plan all I ever get is fast busy signals. When I set debug to
9 I get lots of stuff in /var/log/messages although nothing about the
call. I had stripped everything out of the ser.cfg routing leaving just
a simple dial plan for any digits, this did not work either.
In the beginning I was able to call out however the joy was short lived as
I have mucked up the config so bad I can't seem to find my way back.
Any help would be most appreciated
Thank you
Rick
# ----------- global configuration parameters ------------------------
debug=9 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
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"
#
# $Id: pstn.cfg,v 1.2 2003/06/03 03:18:12 jiri Exp $
#
#
# ------------------ module loading ----------------------------------
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/acc.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/uri.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"
loadmodule "/usr/lib/ser/modules/textops.so"
loadmodule "/usr/lib/ser/modules/group.so"
modparam("auth_db", "db_url","sql://ser:passwd@localhost/ser")
modparam("usrloc", "db_url", "sql://ser:passwd@localhost/ser")
# ----------------- setting module-specific parameters ---------------
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("usrloc", "db_mode", 2)
# -- acc params --
modparam("acc", "log_level", 1)
# that is the flag for which we will account -- don't forget to
# set the same one :-)
# modparam("acc", "log_flag", 1 )
# ------------------------- request routing logic -------------------
# main routing logic
route{
/* ********* ROUTINE CHECKS ********************************** */
# filter too old messages
if (!mf_process_maxfwd_header("10")) {
log("LOG: Too many hops\n");
sl_send_reply("483","Too Many Hops");
break;
};
if (msg:len >= max_len ) {
sl_send_reply("513", "Message too big");
break;
};
/* ********* RR ********************************** */
/* grant Route routing if route headers present */
if (loose_route()) { t_relay(); break; };
/* record-route INVITEs -- all subsequent requests must visit us */
if (method=="INVITE") {
record_route();
};
# now check if it really is a PSTN destination which should be handled
# by our gateway; if not, and the request is an invitation, drop it --
# we cannot terminate it in PSTN; relay non-INVITE requests -- it may
# be for example BYEs sent by gateway to call originator
if (!uri=~"sip:\+?[0-9]+@.*") {
if (method=="INVITE") {
sl_send_reply("403", "Call cannot be served here");
} else {
# forward(uri:host, uri:port);
forward(192.168.1.101, 5060); #ip of my cisco gateway
};
break;
};
# account completed transactions via syslog
setflag(1);
# free call destinations ... no authentication needed
if ( is_user_in("Request-URI", "free-pstn") /* free destinations */
| uri=~"sip:[79][0-9][0-9][0-9]@.*" /* local PBX */
| uri=~"sip:98[0-9][0-9][0-9][0-9]") {
log("free call");
} else if (src_ip==192.168.1.101) {
# our gateway doesn't support digest authentication;
# verify that a request is coming from it by source
# address
log("gateway-originated request");
} else {
# in all other cases, we need to check the request against
# access control lists; first of all, verify request
# originator's identity
if (!proxy_authorize( "gateway" /* realm */,
"subscriber" /* table name */)) {
proxy_challenge( "gateway" /* realm */, "0" /* no
qop */ );
break;
};
# authorize only for INVITEs -- RR/Contact may result in weird
# things showing up in d-uri that would break our logic; our
# major concern is INVITE which causes PSTN costs
if (method=="INVITE") {
# does the authenticated user have a permission for
local
# calls (destinations beginning with a single zero)?
# (i.e., is he in the "local" group?)
if (uri=~"sip:0[1-9][0-9]+@.*") {
if (!is_user_in("credentials", "local")) {
sl_send_reply("403", "No permission
for local calls");
break;
};
# the same for long-distance (destinations begin
with two zeros")
} else if (uri=~"sip:00[1-9][0-9]+@.*") {
if (!is_user_in("credentials", "ld")) {
sl_send_reply("403", " no
permission for LD ");
break;
};
# the same for international calls (three zeros)
} else if (uri=~"sip:000[1-9][0-9]+@.*") {
if (!is_user_in("credentials", "int")) {
sl_send_reply("403", "International
permissions needed");
break;
};
# everything else (e.g., interplanetary calls) is denied
} else {
sl_send_reply("403", "Forbidden");
break;
};
}; # INVITE to authorized PSTN
}; # authorized PSTN
# if you have passed through all the checks, let your call go to GW!
rewritehostport("192.168.1.101:5060");
# forward the request now
if (!t_relay()) {
sl_reply_error();
break;
};
if (uri=~"^sip:[0-9]*@.*") {
log("Forwarding to PSTN\n");
t_relay_to_udp ("192.168.1.101","5060"); # IP address of my cisco
gateway
break;
};
}
I have 0.8.1 running on Suse 9.0. It works great with my Zultys phone,
Polycom IP 600, and the Helmsen agent. When I try to use Windows Messenger
5.0 I have a slight problem.
UserA signs in. UserB signs in, and sees that UserA is online, and there
changes the UserA in the contact list on ONLINE. But UserB never shows as
ONLINE on UserA's desktop. So UserA can't call UserB. The update never
goes to UserA's desktop. What am I missing here?
Scott Morris
