Hi guys,
I'd like to propose another possibility for a highly-available and
scalable system design based on SER:
The problems I've encountered for scalable systems are:
- Distribution of the user location and alias location among the nodes
(user location is based on registrations, alias location comes from
web interfaces and is used for call forwarding).
- Reloading up to date location tables after breakdown and recovery of a
node
So I'm just thinking loud about the following provisioning system:
- Write a client which fulfills the this demands:
- Receive one or more locations from SER via a SER module or from a
web application and distribute them to other
known clients. Take care of retransmissions if a client isn't
reachable or reports a temporary failure.
- Receive one or more locations from other clients and write
them into the SER FIFO. If writing into the FIFO fails, try to
write directly into the database (location-table, alias-table etc.).
Report a temporary failure if this also fails.
Maybe a centralized server should be used which receives the locations
from the clients and distributes them to other clients, so that the
nodes just know about the server and nothing about other nodes. This
would make integration of new nodes easy.
On the other hand, it's another single point of failure, so a
decentralized solution should be considered. But that would mean that
you've to inform every node about the existence of a new node.
The protocol used between the nodes should be simple and fast. So I
think SOAP drops out here. Maybe XMLRPC or ICE
(http://www.zeroc.com/ice.html) could be used.
One might think now why not just use replication on SIP layer, but
t_replicate only supports one peer and you've no possibility to get
locations on a node while it's down. Replication of other location
tables like the alias-table is also not possible.
I'd be willing to release these parts as GPL for creating an open
framework for carrier-grade SER integration, so any feedback,
improvements or flames are highly welcome.
Cheers,
Andy
Does anyone know if there is a way to authenticate SER to MS Active
Directory? I'd like to use an existing user database instead of
having to manage users myself, and most of our users are in Active
Directory.
Has anyone tried this?
What are most SER users using for authentication? Is anyone here
tying into any kind of existing organization user database?
Hi Jiri.
Two questions please.
1.)
Jiri Kuthan HYPERLINK
"mailto:serusers%40iptel.org?Subject=%5BSerusers%5D%20ENUM%20variable%20TLD%
3F&In-Reply-To=4032B769.8000804%40august.net"jiri at iptel.org
Wed Feb 18 02:25:07 CET 2004
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_____
on development branch there is an updated enum module which
allows you to use private enum trees the way you are asking.
If you need it, there should be no conflicts if you take
devel version of enum and put it in your 8.12 source tree.
-jiri
At 01:52 AM 2/18/2004, Greg Fausak wrote:
>Normally I do enum lookups inside my
>domain, however, I want to be able to do
>an enum lookup at other TLD. Is there a
>way to do that? Maybe enum_query("e164.other.com")??
>
I have downloaded the latest code as you suggested but have not been able to
figure out how to specify a different TLD.
I have looked a documentation but that has not helped either.
Can you please tell me what the function I need is called?
2.)
Is there an updated ENUM module that would allow me to preform an ENUM query
based on a spcified prefix?
In other words, what if I would like to preform an ENUM query when a number
comes in with a "**" instead of the "+".
I have modified the c code to do this for me but was wondering if you have
released a "production" version of this code.
Thanks,
David Schwartz
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Version: 6.0.796 / Virus Database: 540 - Release Date: 11/13/2004
Hi list. I have this problem. There's something that's not working,
because, when ever I set up a call, I receive both messages INVITE and
BYE instantly. I think the correct way would it be to send Start
messages (Acct-Status-Type = Start) upon INVITEs and Stop messages
(Acct-Status-Type = Stop) upon BYEs. But this is not happening. Look ...
radrecv: Accounting Request from host c0a801fd code=4, id=205,
length=270
Acct-Status-Type = Start
Service-Type = Sip-Session
Sip-Response-Code = 200
Sip-Method = Invite
User-Name = "1992001(a)192.168.1.253"
Calling-Station-Id = "sip:1992001@192.168.1.253:5060;user=phone"
Called-Station-Id = "sip:1992005@192.168.1.253:5060;user=phone"
Sip-Translated-Request-URI = "sip:1992005@192.168.1.178:11005"
Acct-Session-Id = "6dfb6640367b1ab7(a)192.168.1.113"
Sip-To-Tag = "54061171"
Sip-From-Tag = "39eeac148f846cb9"
Sip-CSeq = "1544"
NAS-IP-Address = 192.168.1.253
NAS-Port-Id = 5060
Acct-Delay-Time = 0
Sending Accounting Ack of id 205 to c0a801fd (nas linux)
radrecv: Accounting Request from host c0a801fd code=4, id=206,
length=270
Acct-Status-Type = Start
Service-Type = Sip-Session
Sip-Response-Code = 200
Sip-Method = Bye
User-Name = "1992001(a)192.168.1.253"
Calling-Station-Id = "sip:1992001@192.168.1.253:5060;user=phone"
Called-Station-Id = "sip:1992005@192.168.1.253:5060;user=phone"
Sip-Translated-Request-URI = "sip:1992005@192.168.1.178:11005"
Acct-Session-Id = "6dfb6640367b1ab7(a)192.168.1.113"
Sip-To-Tag = "54061171"
Sip-From-Tag = "39eeac148f846cb9"
Sip-CSeq = "1544"
NAS-IP-Address = 192.168.1.253
NAS-Port-Id = 5060
Acct-Delay-Time = 0
Sending Accounting Ack of id 206 to c0a801fd (nas linux)
Do you see ? ...Both outputs show an Acct-Status-Type = Start, but first
output refers to an INVITE and second output refers to BYE message. And
the strange thing is that both messages arrive instantly, as soon as the
peer answers the phone. Why ??? I don't think this is correct, isn't it
?
Any hint ?
here goes my ser.cfg ...
[...]
# -- RADIUS ACC --
modparam("acc", "radius_config", "/etc/radiusclient.conf")
modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag",
2) modparam("acc", "log_level", 1) modparam("acc", "service_type", 15)
[...]
if (loose_route())
{
t_relay();
break;
};
if (uri==myself)
{
if (method=="REGISTER")
{
# Lo siguiente es para auth con RADIUS
if (!radius_www_authorize(""))
{
www_challenge("", "0");
break;
};
save("location");
break;
};
if (method=="INVITE")
{
setflag(1);
};
# native SIP destinations are handled using our USRLOC
DB
if (!lookup("location"))
{
sl_send_reply("404", "Not Found");
break;
};
};
t_relay();
[...]
Regards,
Lucas
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Checked by AVG Anti-Virus.
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Yeap.
Me.!
Cheers
Ricardo Martinez
-----Mensaje original-----
De: O-Zone [mailto:liste@zerozone.it]
Enviado el: Miércoles, 22 de Diciembre de 2004 11:12
Para: ser users
Asunto: [Serusers] SER + RADIUS
Someone use SER with Radius Auth ?
Oz
--
------
O-Zone ! www.zerozone.it
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Hi All,
Any idea, how to change registration timeout value in
ser.cfg so that my phone register with the server
every 5 minutes in place of default 1 minute.
Please advise.
Regards,
Suvendu.
________________________________________________________________________
Yahoo! India Matrimony: Find your life partner online
Go to: http://yahoo.shaadi.com/india-matrimony
Hello,
I'm running ser 0.8.14 and rtpproxy CVS on debian sarge.
You can find my ser.cfg. below.
I'm forwarding my calls to a Cisco AS5350 whish serves as a PSTN
gateway. I can place a call to a PSTN number and the phne rings, but
then no audio is sent or received. I use kphone as a client behind NAT.
thnaks
Luca
--- ser.cfg routing ---
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;
};
# !! Nathelper
# Special handling for NATed clients; first,NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding is used); also,
# the received test should, if completed, should check all
# vias for rpesence of received if (nat_uac_test("3")) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if (method == "REGISTER" || ! search("^Record-Route:")) {
log("LOG: Someone trying to register from private IP,
rewriting\n");
# This will work only for user agents that support
symmetric
# communication. We tested quite many of them and
majority is
# smart enough to be symmetric. In some phones it takes
a configuration
# option. With Cisco 7960, it is called NAT_Enable=Yes,
with kphone it is
# called "symmetric media" and "symmetric signalling".
fix_nated_contact(); # Rewrite contact with source IP of
signalling
if (method == "INVITE") {
fix_nated_sdp("1"); # Add direction=active to
SDP
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as NATed
};
# 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
if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
rewritehostport("$AS5350_IP:5060");
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==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest
authentication
if (!www_authorize("my.domain",
"subscriber")) {
www_challenge("my.domain", "0");
break;
};
save("location");
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
# native SIP destinations are handled using our USRLOC
DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
# -- nathelper --
route[1]
{
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)"
&& !search("^Route:")){
sl_send_reply("479", "We don't forward to private IP
addresses");
break;
};
# if client or server know to be behind a NAT, enable relay
if (isflagset(6)) {
force_rtp_proxy();
};
# NAT processing of replies; apply to all transactions (for
example,
# re-INVITEs from public to private UA are hard to identify as
# NATed at the moment of request processing); look at replies
t_on_reply("1");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
onreply_route[1] {
# NATed transaction ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
# otherwise, is it a transaction behind a NAT and we did
not
# know at time of request processing ? (RFC1918
contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
Hello,
I'm trying to get asterisk to work with SER - I have ser forwarding
the sip connection to asterisk with the following routing logic.
if (uri=~"^sip:8[0-9]{7}.*") {
forward( 10.0.18.3, 5061 );
break;
Asterisk accepts the connection on port 5061 - but I get the following
once it connects to asterisk.
001 -- Executing Answer("SIP/ast.digicen.com-08289858", "") in new stack
002 -- Executing Dial("SIP/ast.digicen.com-08289858",
"IAX2/kknott@NuFone/12039063173") in new stack
003 -- Called kknott@NuFone/12039063173
004 -- Call accepted by 66.225.202.72 (format gsm)
005 -- Format for call is gsm
006 -- Hungup 'IAX2/NuFone/1'
007 == No one is available to answer at this time (1:0/0/0)
008 == Auto fallthrough, channel 'SIP/ast.digicen.com-08289858'
status is 'NOANSWER'
009 > cdr_odbc: Query Successful!
010 Jan 30 21:41:42 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP
headers...
011 Jan 30 21:41:42 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP
headers...
012 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP
headers...
013 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP
headers...
014 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP
headers...
015 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
016 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
017 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
018 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
019 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
020 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
021 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
022 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
023 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
024 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
025 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
026 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
027 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
028 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
029 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
030 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
031 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
032 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
033 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
034 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
035 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
036 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
037 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
038 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
039 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
040 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
041 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
042 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
043 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
044 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
045 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
046 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
047 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
048 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
049 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
050 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
051 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
052 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
053 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
054 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
055 Jan 30 21:41:46 WARNING[9829]: chan_sip.c:729 retrans_pkt: Maximum
retries exceeded on call
5A2650C2-94B0-44AB-B241-66DB67BE2CF7(a)192.168.1.104 for seqno 52428
(Non-critical Response)
056 Jan 30 21:41:46 WARNING[9829]: chan_sip.c:729 retrans_pkt: Maximum
retries exceeded on call
5A2650C2-94B0-44AB-B241-66DB67BE2CF7(a)192.168.1.104 for seqno 102
(Non-critical Request)
057 Jan 30 21:41:50 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
058
059 *CLI>
060 *CLI>
Asterisk will usually core dump after that - I have no idea where to
start to fix it - No machines are behind NAT.
Thanks,
Patrick
Hello All,
My task is to program the ser.cfg in a way that it
connects to an external database(any machine in LAN,other than the
server on which SER is running)
After getting connected to the database I need to query the
database and get a result set of IPAddresses based on the destination
number that I passed in the query..and further use this set to rewrite
the call to the IP Address retrieved.
To accomplish this task I need--
1.how to extract the destinationno from sip_uri
2.how to use the database interface given in SER programmer's guide
I went through SER programmer's guide,but if anybody can provide
me a simple
example of using database interface,it wud be of great help.
Thanks!
This seems to be a perennial trouble for people.
I see many similar threads.
I have 0.9.0
serweb version from when I previously downloaded 08.14
ser is running with mysql, registration and calls running ok etc
serweb start up as far as the admin login screen
admin/heslo and anything else I throw at it fails to login.
It just returns to login screen.
After failed login I get a little message : error in sql query line 30
which is the line of the print for error in …..
I have seen many many reports of this but no confirmation from any one that
any of the suggested fixes actually did so…
Is this a known problem
Is it a compatibility issue with 090
Anyone have a solution?
Cheers Chris
_____
Hi,
This is my first time installation of serweb on Fedora Core 2.
When I view it from my own linux box and type in
username: admin
password: heslo
Nothing happen.
But when I use I.E. on other computers, I see the error message:
Bad username and password.
I have set php.ini register_globals = on already. I have also restarted
httpd.
What is my problem?
Thomas
On Wed, 2004-12-01 at 08:49, Chris HARIGA wrote:
> Hi,
>
> I have the same problem :((
> The register_global=on is present and if I try to login I get the "Bad
> username or password" message :(
>
> Best regards,
>
> Chris HARIGA
>
>
> -----Original Message-----
> From: HYPERLINK
"http://lists.iptel.org/mailman/listinfo/serusers"serusers-bounces at
iptel.org [mailto:HYPERLINK
"http://lists.iptel.org/mailman/listinfo/serusers"serusers-bounces at
iptel.org] On
> Behalf Of Karel Kozlik
> Sent: Tuesday, November 30, 2004 2:52 PM
> To: support
> Cc: HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"serusers at
iptel.org
> Subject: Re: [Serusers] cannot login to serweb
>
> Any more details? Your serweb version, error messages etc.?
>
> Did you read FAQ in INSTALL file? Special this:
>
> Q: All my login attempts lead to the previous screen without any
> kind of progress or error indication.
> A: Really make sure that register_globals is turned on in your
> php.ini. (Also make sure that you are changing the php.ini in
> use by your server and register_globals is not turned off
> somewhere else in the configuration file.)
> For check real php configuration create file phpinfo.php
> in html root with content <? phpinfo(); ?> and look to it by
> browser. There is _all_ informations about php.
>
> Karel
>
>
> support wrote:
> > Hi everyone,
> >
> >
> > After I have installed serweb by default, with db_name=ser and
> > db_passwd=heslo, when I go to HYPERLINK
"http://localhost/html/admin/index.php"http://localhost/html/admin/index.php
> > login page, using
> >
> > Username: admin
> > Password: heslo
> >
> >
> > I cannot login the page.
> >
> >
> >
> > Thomas
> >
> > _______________________________________________
> > Serusers mailing list
> > HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"Serusers at
iptel.org
> > HYPERLINK
"http://lists.iptel.org/mailman/listinfo/serusers"http://lists.iptel.org/mailm
an/listinfo/serusers
> >
>
> _______________________________________________
> Serusers mailing list
> HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"Serusers at
iptel.org
> HYPERLINK
"http://lists.iptel.org/mailman/listinfo/serusers"http://lists.iptel.org/mailm
an/listinfo/serusers
>
> ______________________________________________________________________
> _______________________________________________
> Serusers mailing list
> HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"Serusers at
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