Hi all,
I'm currently working with a cisco3600 with SER 0.9.2. Thanks to Onsp.org, I
can land calls from SER to Cisco 3600 without any problems. Yet I'm
experiencing some problems with Cisco3600 passing calls onto SER.
My setup is as follow:
Cisco 3600 ==> SER+Mediaproxy ===> IP Phone
When a call come in via DID method through the Cisco3600, SER received the all
SIP messages (I got it via ngrep), yet the call couldn't pass onto the IP
Phone.
Currently, all devices (including cisco) have been configured to use port 5080
instead of port 5060 due to port restrictions on the network, and that I have
specified the ip in the trusted table under mysql. Yet I don't know why the
call still can not get through. The following are the logs I received from
ngrep and cisco..... Any help is welcome and is GREATLY APPRCIATED!!
------------------- ngrep from SER server
U cisco-ip:55299 -> SER-ip:5080
INVITE sip:6119203@SER-ip;user=phone;phone-context=local SIP/2.0..Via:
SIP/2.0/UDP cisco-ip:5060..From: "6113640" <sip:6113640@cisco-ip>..To:
<sip:6119203@SER-ip;user=phone;phone-context=local>..............
#
U SER-ip:5080 -> cisco-ip:5060
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
222.50.103.247:5060..From: "6113640" <sip:6113640@cisco-ip>..To:
<sip:6119203@SER-ip;user=phone;phone-context=local>...........................
#
U cisco-ip:55299 -> SER-ip:5080
CANCEL sip:6119203@SER-ip;user=phone;phone-context=local SIP/2.0..Via:
SIP/2.0/UDP 222.50.103.247:5060..From: "6113640"<sip:6113640@cisco-ip>..To:
<sip:6119203@SER-ip;user=phone;phone-context=local>.................
---------------- mysql trusted table
+----------------+-------+--------------+
| src_ip | proto | from_pattern |
+----------------+-------+--------------+
| cisco-ip | any | ^sip:.*$ |
+----------------+-------+--------------+
----------------- log from cisco3600
Jun 23 14:23:53: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:
SER-ip:5080
Jun 23 14:23:57: Received:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP cisco-ip:5060
From: "6113640" <sip:6113640@cisco3600>
To:
<sip:6119203@SER-ip;user=phone;phone-context=local>;tag=bc78c497f1336138e83d95d77d963f1b-f81d
Call-ID: 22B8BA3E-E32911D9-82A1EC22-7AD29AB0(a)222.50.103.247
CSeq
Hi
If I use avpops for IP based auth, and drop the normal username/password
combo aside from spoofing what is the downside if any. Also if I do IP
based auth, can I auth once, and be done with it, or is it auth once per
call, I guess its once per call, if so is there any way to bypass auth
completely for a particular IP address, again I am assuming no, since
the IP will still need to be checked for each request.
Iqbal
Hi there:
Have install the ser and serweb 0.93 official version.
When I try to open the admin page, It just returns me
the following message.
IPaddress admin interface
And I try to see the return html source code, it stops
at tag <div>. The detail as the follow:
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd">
<html>
<head>
<title>SIP Express Router - web interface</title>
<meta http-equiv="Content-Type" content="text/html;charset=utf-8">
<meta name="Author" content="Karel Kozlik <karel at iptel dot org>">
<meta http-equiv="PRAGMA" content="no-cache">
<meta http-equiv="Cache-control" content="no-cache">
<meta http-equiv="Expires" content="Wed, 29 Jun 2005 04:18:07 GMT">
<LINK REL="StyleSheet" HREF="/serweb/html/domains/_default/styles.css" TYPE="text/css">
</head>
<script language="JavaScript" src="/serweb/html/js/login_completion.js.php"></script><body><h1>
IPaddress admin interface</h1><hr>
<div class="swMain">
Can anybody help me?
Thanks a lot.
Kind Regards,
wilson
2005-06-29
Hi there:
Have install the ser and serweb 0.93 official version.
When I try to open the admin page, It just returns me
the following message.
IPaddress admin interface
And I try to see the return html source code, it stops
at tag <div>. The detail as the follow:
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd">
<html>
<head>
<title>SIP Express Router - web interface</title>
<meta http-equiv="Content-Type" content="text/html;charset=utf-8">
<meta name="Author" content="Karel Kozlik <karel at iptel dot org>">
<meta http-equiv="PRAGMA" content="no-cache">
<meta http-equiv="Cache-control" content="no-cache">
<meta http-equiv="Expires" content="Wed, 29 Jun 2005 04:18:07 GMT">
<LINK REL="StyleSheet" HREF="/serweb/html/domains/_default/styles.css" TYPE="text/css">
</head>
<script language="JavaScript" src="/serweb/html/js/login_completion.js.php"></script><body><h1>
IPaddress admin interface</h1><hr>
<div class="swMain">
Can anybody help me?
Thanks a lot.
Kind Regards,
wilson
2005-06-29
I'm updating my ser 0.9.2 to ser 0.9.3
Whe I run the ser command I'm getting the following error for the module
acc:
0(32050) ERROR: module version mismatch for
/usr/local/lib/ser/modules/acc.so; core: ser 0.9.3 (i386/linux); module:
ser 0.9.2 (i386/linux)
I compile ser in the following way:
make clean
make proper
make all
make install
I already checked the creation date for the acc.so file and it has the
same like the other modules.
Does anybody know why this is happening?
Regards
Alberto Cruz
> -----Org. message-----
> Fra: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]
På
> vegne af Sebastian Kühner Finally I have my serweb running (it had
> problems with apache2 and php5... many problems!).
If i may ask, what problems were these...? I myself am having problem
with array in my_account.php - you to?
> Now I have the problem that serweb doesn't show me the missed
> calls. I looked at the mysql-database and noticed that ser
> doesn't save the domain and username - information in the
> table "missed_calls"...
For this problem, i also would like to know a solution..
René
SYSTEM: BSD5.3, http 2.0.54, php 5.0.3, sql 4.0.24
Hello everyone,
I am testing ser 0.10.99 dev8, radiusclient-ng 0.5.1
with freeradius 1.0.2 on the same machine. It is not
working. I am not sure whether it is a version or a
configuration problem. The question is, do this
versions work all together? Is there any
radiusclient-ser compatibility matrix?
I compile acc module with this options in Makefile:
# uncomment the next line if you wish to enable SQL
accounting
#DEFS+=-DSQL_ACC
# uncomment the next two lines if you wish to enable
RADIUS accounting
DEFS+=-DRAD_ACC -I$(LOCALBASE)/include
LIBS=-L$(LOCALBASE)/lib -lradiusclient-ng
# uncomment the next two lines if you wish to enable
DIAMETER accounting
#DEFS+=-DDIAM_ACC
I don't have in my system any lradiusclient-ng, what I
have is libradiusclient-ng.so. Is that OK??
Any clue is apreciated,
Pablo.
__________________________________
Do you Yahoo!?
Yahoo! Mail - Helps protect you from nasty viruses.
http://promotions.yahoo.com/new_mail
Hello,
I read lot of example ducumentations from net about
the ser configuration. I found one example from
voip-info.org. i configured the ser.cfg file
accordidng to the guide. but when i am trying to
restart the ser using /usr/local/sbin/serctl restart i
am getting some pid file error messate like :
----------------------------------
[root@localhost root]# /usr/local/sbin/serctl stop
Stopping SER : No PID file found! SER probably not
running
[root@localhost root]# /usr/local/sbin/serctl start
Starting SER : PID file /var/run/ser.pid does not
exist -- SER start failed
[root@localhost root]#
--------------------------------
Here is my ser.cfg file configurations.
I wanted to explain how i am going to use it.
I have one VoIP Gatekeeper (MVTS) Which supports to
register SER i want to register the SER in MVTS and
the IP Phones i want to register in SER. So the call
can be pass to destination using like:
IP Phone > SER > MVTS > Destination PSTN.
MVTS IP : 195.22.146.20
SER : 10.0.0.20
IP Phone : Should use the ser IP and uid/pws
--------------- ser.cfg ----------------------
#
# $Id: pstn.cfg,v 1.2 2003/06/03 03:18:12 jiri Exp $
#
#
# ------------------ module loading
----------------------------------
loadmodule "modules/sl/sl.so"
loadmodule "modules/tm/tm.so"
loadmodule "modules/acc/acc.so"
loadmodule "modules/rr/rr.so"
loadmodule "modules/maxfwd/maxfwd.so"
loadmodule "modules/mysql/mysql.so"
loadmodule "modules/auth/auth.so"
loadmodule "modules/auth_db/auth_db.so"
loadmodule "modules/group/group.so"
loadmodule "modules/uri/uri.so"
# ----------------- setting module-specific
parameters ---------------
modparam("auth_db",
"db_url","sql://ser:heslo@10.0.0.26/ser")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
# 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 (len_gt( max_len )) {
sl_send_reply("513", "Wow Message too
large");
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);
};
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==192.168.0.10) {
# 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("192.168.0.10:5060");
# forward the request now
if (!t_relay()) {
sl_reply_error();
break;
};
}
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
Tel: +974 - 4883068
ICQ: 276994704
YM!: abdul_zu
Fax: +974 - 4883063
Doha Qatar
http://www.hatif.com
__________________________________
Yahoo! Mail Mobile
Take Yahoo! Mail with you! Check email on your mobile phone.
http://mobile.yahoo.com/learn/mail
Hello all.
I'm trying to install SER with Radius support for accounting, but it's not
compiling right.
I installed the freeradius server 0.9.1, radius client 0.3.2. and SER
0.9.0.
The freeradius server and radius client appear to be right, I made the
Radius server test just like the ser_radius.html manual, and the output is
right.
But...when I try to complie the ser I get some errors with the cpl,
cpl_parser, pidf, db_con and acc.o modules... Please, can anybody help me?
Thanks a lot
Rosa
Hello,
Finally I have my serweb running (it had problems with apache2 and php5...
many problems!).
Now I have the problem that serweb doesn't show me the missed calls. I
looked at the mysql-database and noticed that ser doesn't save the domain
and username - information in the table "missed_calls"...
How can I make ser to save theese informations, too?
Thanks!
Sebastian
