Hello all, what file directs where SER looks for the mysql.sock file. In my
case mysql.sock is located in /tmp and as you can see SER is lookin for it
in /var/lib/mysql
Mar 10 16:46:33 ttalksvr /usr/sbin/ser[6592]: connect_db(): Can't connect to
local MySQL server through socket '/var/lib/mysql/mysql.sock' (2)
Thanks Mike
We are contemplating building an Admin interface for SER that could run on a
windows os. We like the fact that the server is running on Linux but we
require a more intuitive interface for administration.
A web interface is not acceptable, just not enough functionality. We would
like to create an executable that runs on windows which interfaces with SER
server running on Linux.
I am curious if anyone else could make use of a GUI that would incorporate
all of the SER features and functionality. What do you think? Should we
attempt it? Does anyone have any preference on the language we use, e.g.
JAVA, C++ ?
Erik Lagerway
Xten Networks Inc.
7170-515 West Hastings St
Vancouver, BC
V6B5K3
Ph. 1.604.878.0440 ext.5
Some interesting modifications I have made to improve
acc process.
1)
--------------
Some changes made to acc.c to strip down the user
names for output and also to get missed calls logged
with callee rather than caller. It was showing missed
calls on the callers missed_calls.php which is
incorrect.
Here are the extracts, I can post the full prog if
required:
/*
JF added following block to extract to user
*/
str_copy(&to_user, &rq->to->body);
if ((ul_get_user(&to_user) < 0) ||
!to_user.len) {
LOG(L_ERR, "ERROR: acc_request: Error
while extracting to_username\n");
return -1;
}
/*
str_copy(&user, &rq->first_line.u.request.uri);
JF modified this copy statement to create correct acc
username (from not to!)
*/
str_copy(&user, &rq->from->body);
2)
---------------------
I modified the $q in accounting.php to the below to
include calls where BYE sent by callee as well as
caller (before it was not displaying calls when closed
by callee). Also elimination of double counting of
calls due to
multiple INVITEs being sent by UAC. max(time) is
required as distinct rows are produced if the invites
are > 1sec apart. "Group by" puts all the
INVITEs together so that the latest one can be
selected with max(). Assume that latest invite is
correct for measuring call start time. The max(time)
column has been aliased to c2 so that it can be used
in the output $time=Substr($row->c2,0,16).
Here it is:
$q="select distinct t1.sip_callid, max(t1.time) c2,
t1.sip_to,
sec_to_time(unix_timestamp(t2.time)-unix_timestamp(max(t1.time)))
as length ".
"from ".$config->table_accounting." t1,
".$config->table_accounting." t2 ".
"where (t1.user='".$auth->auth["uname"]."' and
t1.sip_method='INVITE') and
((t2.user='".$auth->auth["uname"]."' or
t2.sip_to='".$auth->auth["uname"]."') and
t2.sip_method='BYE') and ".
"t1.sip_callid=t2.sip_callid group by t1.sip_callid
".
"order by t1.time desc";
3) I modified the &q in missed_calls.php to the below
to include calls to aliases as well. The query looks
in the missed_calls table for the sip_to = logged-in
user and also joins with aliases table to extract
calls to alias of contact. Otherwise missed_calls.php
only provided calls made directly to a user and
ignored calls to aliases.
Here it is:
$q="select distinct t1.user, t1.sip_to, t1.time,
t1.sip_status, t1.sip_callid from
".$config->table_missed_calls."
t1,".$config->table_aliases." t2 where
t1.sip_to='".$auth->auth["uname"]."' OR
('sip:".$auth->auth["uname"]."@".$config->default_domain."'=t2.contact
AND t2.user=t1.sip_to) order by time desc";
echo $q;
4)
-----------------
Any comments will be appreciated.
__________________________________________________
Do you Yahoo!?
Yahoo! Tax Center - forms, calculators, tips, more
http://taxes.yahoo.com/
How about a java applet for use with a web browser? I'd have no problem
with a
requirement for a Windows PC for management, but I can see it being an
issue
for others.
And yes I would use it, in whatever form it took.
Dan
-----Original Message-----
From: Erik Lagerway [mailto:sipdev@xten.com]
Sent: Thursday, March 06, 2003 1:50 PM
To: sip-implementors(a)cs.columbia.edu
Cc: serusers(a)lists.iptel.org
Subject: [Serusers] Admin Interface
We are contemplating building an Admin interface for SER that
could run on a windows os. We like the fact that the server is running
on Linux but we require a more intuitive interface for administration.
A web interface is not acceptable, just not enough
functionality. We would like to create an executable that runs on
windows which interfaces with SER server running on Linux.
I am curious if anyone else could make use of a GUI that would
incorporate all of the SER features and functionality. What do you
think? Should we attempt it? Does anyone have any preference on the
language we use, e.g. JAVA, C++ ?
Erik Lagerway
Xten Networks Inc.
7170-515 West Hastings St
Vancouver, BC
V6B5K3
Ph. 1.604.878.0440 ext.5
Hi,
I tried to use the SER SIP server together with a number of different
UAs. I successfully used a Pingtel phone, pingtel softphone, the
Ubiquity UA, and kphone on linux. However, when I tried to use SIPC from
Columbia Uni I keep getting eror messages 400: Bad Request. As all the
other UAs are working fine I assume this may be a problem with sipc.
Have you any experiences with using sipc and ser?
I attach an error log between the Pingtel phone (139.153.254.222) is
trying to send an INVITE to sipc (139.153.254.34). SER (and ngrep) are
running on 139.153.254.50. (registrations are going through fine and
also invites the other way (from sipc to pingtel) go through ok).
Thanks very much for your help!
Regards,
Mario
--
Mario Kolberg phone: +44 (0)1786 46 7440
Lecturer in Computing Science fax : +44 (0)1786 46 4551
email: mko(a)cs.stir.ac.uk
Department of Computing Science and Mathematics
University of Stirling
Stirling FK9 4LA
Scotland, UK
--
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. Opinions, conclusions and other
information in this message that do not relate to the official
business of the University of Stirling shall be understood as neither
given nor endorsed by it.
I everybody.
where can i download serweb????
at http://developer.berlios.de/projects/serweb/ there is not any release for
download and in home page it says "Web-based user provisioning, serweb,
available"
thank's all
-------------------------------------------------
This mail sent through IMP: http://mail.info.unlp.edu.ar/
Jiri,
Scenario is providing IP Telephony to the household.
I am more concern about the security of the Hardphone. I am thinking of auto-provisioned the hardphone (eg C7960, ATA186) without subsriber intervention. What the subscriber know is their phone # (Just like legacy phone system).
Since the Hardphone is 'hard-coded', the phone can move round the vicinity of the redisential area and still able to make a call. Potentially this will lead to abuse, as someone may take the phone to a different location when owner is not around and make a 'free' call, return back the phone and the billing still charge the original subsriber.
Any other suggestion to counter this issue is much appreacited.
SSng
-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]
Sent: Wednesday, March 05, 2003 12:18 AM
To: Ng, Soo Sim; serusers(a)lists.iptel.org
Subject: Re: [Serusers] multiple registration on one user login
At 03:08 PM 3/4/2003, Ng, Soo Sim wrote:
>I have such requirements. In providing sip-based residential ip telephony, I would like to restrict each home subsriber is only allowed to register one UA per account. This would make easy for billing purposes and for security reasons.
>
>Is there a way to achieve this requirement with SER?
If that is your desparate wish, it is little overhead to make you happy.
I'm still not sure though, it is a useful thing.
Maybe an operator can make more revennues if my wife can accept calls at
any phone in my building and initiate calls in parallel with my doughter.
What are exactly the billing/security reasons here?
-Jiri
Hello,
I fear that such a case can't be avoided with allowing only
a single registration. If I steal your phone away from your
desk, you will not register with it anymore, but I will and
we will have exactly one valid registration. Leaving SIP
phones with hard-wired passwords on your desk has simply the
same potential as leaving your credit-card or cell-phone there.
What can be done about fraud?
User education -- don't leave your money and phone unattended.
Hotline -- report stolen phones to lock the account.
PIN Lock -- use phones which can log-off and log-on (I'm not aware
of any now -- only 3com used to do that)
-Jiri
ps -- ability to move is a feature. I know people who are very glad
to use Vonage's US phone number and move with their ATAs and the
US phone number around in Europe.
At 11:37 PM 3/5/2003, Ng, Soo Sim wrote:
>Jiri,
>
>Scenario is providing IP Telephony to the household.
>I am more concern about the security of the Hardphone. I am thinking of auto-provisioned the hardphone (eg C7960, ATA186) without subsriber intervention. What the subscriber know is their phone # (Just like legacy phone system).
>
>Since the Hardphone is 'hard-coded', the phone can move round the vicinity of the redisential area and still able to make a call. Potentially this will lead to abuse, as someone may take the phone to a different location when owner is not around and make a 'free' call, return back the phone and the billing still charge the original subsriber.
>
>Any other suggestion to counter this issue is much appreacited.
>
>SSng
>
>-----Original Message-----
>From: Jiri Kuthan [mailto:jiri@iptel.org]
>Sent: Wednesday, March 05, 2003 12:18 AM
>To: Ng, Soo Sim; serusers(a)lists.iptel.org
>Subject: Re: [Serusers] multiple registration on one user login
>
>
>At 03:08 PM 3/4/2003, Ng, Soo Sim wrote:
>>I have such requirements. In providing sip-based residential ip telephony, I would like to restrict each home subsriber is only allowed to register one UA per account. This would make easy for billing purposes and for security reasons.
>>
>>Is there a way to achieve this requirement with SER?
>
>If that is your desparate wish, it is little overhead to make you happy.
>I'm still not sure though, it is a useful thing.
>
>Maybe an operator can make more revennues if my wife can accept calls at
>any phone in my building and initiate calls in parallel with my doughter.
>
>What are exactly the billing/security reasons here?
>
>-Jiri
--
Jiri Kuthan http://iptel.org/~jiri/
I have to dig at it abit, but it may also be a codec issue on the Phone.
I had a similar error before telling the Cisco which codec to use. I see
you are using G711Alaw. Can you try G711Ulaw?
The PSTN hand-off section of your config looks very familiar to me, so if
you did pull it from the Howto, note that it is an example of what worked
for me. I've seen quite a few much more sophisticated scripts on the list.
Dan
-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]
Sent: Wednesday, March 05, 2003 2:29 AM
To: Rikard Westlund; serusers(a)lists.iptel.org
Subject: Re: [Serusers] Messenger 4.7, CIsco and PSTN
I suspect what happens is that you forward the requests with your server's address in its r-uri to gateway "as is" and the Cisco
gateway would like to see its IP address in the r-uri instead. Try rewriting r-uri -- see bellow.
As for the Messenger problem, see our doc http://www.iptel.org/ser/doc/seruser-html/x878.html#AEN890
-Jiri
At 11:04 AM 3/5/2003, Rikard Westlund wrote:
>Hi all,
>
>I have a Ser 0.8.10-2 install on a Redhat 7.3 kernel 2.4.18-3.
>
>As clients I use Pingtel and messenger 4.7. I have followed the setup
>guide on http://www.fitawi.com/ser-Howto.html
>
>I can register the pingtel phone with no problem. I can call from the
>PSTN to the pingtel via a Cisco AS5300 with no problems.
>
>When i try toi call from pingtel to PSTN iget the following answer:
>
>1. from pingtel to ser - INVITE sip:<pstnnumber>@serserver_ip 2. from
>ser to pingtel - Status: 100 trying 3. from ser to cisco - INVITE
>sip:<pstnnumber>@serserver_ip 4. from cisco to ser - Status: 400 bad
>request - ínvalid IP address' 5. from cisco to ser - Status: 400 bad
>request - ínvalid IP address'
>
>This is my ser.cfg:
>
># $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $
>#
># simple quick-start config script
>#
>
># ----------- global configuration parameters ------------------------
>
>debug=4 # debug level (cmd line: -dddddddddd)
>fork=yes
>log_stderror=no # (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"
>
># ------------------ 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"
>
># ----------------- 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", "secret", "alsdkhglaksdhfkloiwr") modparam("auth",
>"calculate_ha1", yes) #
># If you set "calculate_ha1" parameter to yes (which true in this config),
># uncomment also the following parameter)
>#
>modparam("auth", "password_column", "password")
>
># ------------------------- request routing logic -------------------
>
># main routing logic
>
>route{
>
> # initial sanity checks -- messages with
> # max_forwars==0, or excessively long requests
> if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","Too Many Hops");
> break;
> };
> if (len_gt( max_len )) {
> sl_send_reply("513", "Message too big");
> break;
> };
>
>
> # Do strict routing if pre-loaded route headers present
> rewriteFromRoute();
>
> # 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("norrtull.nexus.se", "subscriber")) {
> www_challenge("norrtull.nexus.se", "0");
> break;
> };
>
> save("location");
> break;
> };
>
># attempt handoff to PSTN
>
> if (uri=~"^sip:1[0-9]*@norrtull.nexus.se") { ## This assumes that the caller is
> log("Forwarding to PSTN\n"); ## registered in our realm
*** here *** rewrite uri prior to fwd-ing.
rewritehostport("cisco_ip:5060");
> t_relay_to( "cisco_ip", "5060"); ## Our Cisco router
> 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
> if (!t_relay()) {
> sl_reply_error();
> };
>
>}
>
>---------------------------------
>
>In the cisco I have the following config:
>
>!
>dail-peer voice 25 voip
>destination-pattern XXXX
>session protocol sipv2
>codec g711alaw
>no vad
>session target ipv4:serserver_ip
>!
>
>I have added 2 subscribers with the serctl command and registration is
>working well from pingtel. In Messenger 4.7 it's not working at all. I
>get 401 Unauthorized.
>
>Well I think thats about it..
>
>Please feel free to contact me if you need more information
>
>Best regards
>
>Rikard Westlund
>
>
>
>_________________________________________________________________
>Tired of spam? Get advanced junk mail protection with MSN 8.
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>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hi all,
I have a Ser 0.8.10-2 install on a Redhat 7.3 kernel 2.4.18-3.
As clients I use Pingtel and messenger 4.7. I have followed the setup guide
on
http://www.fitawi.com/ser-Howto.html
I can register the pingtel phone with no problem. I can call from the PSTN
to the pingtel via a Cisco AS5300 with no problems.
When i try toi call from pingtel to PSTN iget the following answer:
1. from pingtel to ser - INVITE sip:<pstnnumber>@serserver_ip
2. from ser to pingtel - Status: 100 trying
3. from ser to cisco - INVITE sip:<pstnnumber>@serserver_ip
4. from cisco to ser - Status: 400 bad request - ínvalid IP address'
5. from cisco to ser - Status: 400 bad request - ínvalid IP address'
This is my ser.cfg:
# $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=4 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (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"
# ------------------ 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"
# ----------------- 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", "secret", "alsdkhglaksdhfkloiwr")
modparam("auth", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth", "password_column", "password")
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwars==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (len_gt( max_len )) {
sl_send_reply("513", "Message too big");
break;
};
# Do strict routing if pre-loaded route headers present
rewriteFromRoute();
# 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("norrtull.nexus.se", "subscriber")) {
www_challenge("norrtull.nexus.se", "0");
break;
};
save("location");
break;
};
# attempt handoff to PSTN
if (uri=~"^sip:1[0-9]*@norrtull.nexus.se") { ## This
assumes that the caller is
log("Forwarding to PSTN\n"); ## registered in
our realm
t_relay_to( "cisco_ip", "5060"); ## Our Cisco
router
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
if (!t_relay()) {
sl_reply_error();
};
}
---------------------------------
In the cisco I have the following config:
!
dail-peer voice 25 voip
destination-pattern XXXX
session protocol sipv2
codec g711alaw
no vad
session target ipv4:serserver_ip
!
I have added 2 subscribers with the serctl command and registration is
working well from pingtel. In Messenger 4.7 it's not working at all. I get
401 Unauthorized.
Well I think thats about it..
Please feel free to contact me if you need more information
Best regards
Rikard Westlund
_________________________________________________________________
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