That's a one-way trick to make 5.0 accept 4.7's instant messages.
There is no way to make 4.7 or other standard-based implementation
understand 5.0's proprietary messaing model (sending INVITEs to
establish a messaging session).
-jiri
At 04:05 PM 10/28/2003, Rork, Joseph (J.P.) wrote:
>I've followed the instructions:
>Attention: instant messaging model has changed from 4.7 to 5.0 dramatically and in a non-standard way. 5.0 takes session establishement using INVITE and drops out-of-dialog MESSAGEs on the floor, whereas sending "481 Call Leg/Transaction...". To permit them, apply the following registry setting: HighSecurityMode HKLM\Software\Policies\Microsoft\Messenger\Client\{83D4679F-B6D7-11D2-BF36-0 0C04FB90A03}\_Default\EnableSIPHighSecurityMode DWORD= 0 -- Low Security 1 -- High Security 2(Default, same as not set) --- Medium security You need set it to 0 in order to make peer-to-peer IM. Since WM4.7 client is not registered in the LCS server, I guess that WM5 treats it as peer-to-peer call.
>
>And I'm still getting the following error:
><<messenger5dump>>
>
>Any ideas?
>
>--joe
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
HI, my name is Ivan Dario
First of all, congratulations for iptel.org , I'm a student and i'm really interested about SER and what i can do with it.
I was reading SER description when a friend of mine told me about JTAPI, a java telephony API
I would be gratefuled if you explain me if this is for telephony ip and how is SER relationed with it
I'm also worried about wath kind of development can i do Being based on SER, and one last thing
I only have a computer with 16MB on ram, 2GB on Hard Disk and pentium 120 MHZ, what could i do in order
to install SER on it, i mean what version and all that..
Please.. any be your answer, send me a mail
Thanks a lot
IVAN DARIO
_________________________________________________________
http://www.latinmail.com. Gratuito, latino y en espa�ol.
Hi,
We've just installed ser we've noticed that when placing a call from a
Cisco IP phone the "From:" is set to "username(a)serserver.domain.com"
rather than the expected "username(a)domain.com". Other SIP clients
(we've tested with MS Messenger) don't exhibit this problem.
At first we thought ser was re-writing this, but ngrep revealed that to
not be the case (the phone was actually sending this information).
Inserting "rewritehost("domain.com")" into the config didn't seem to
alter this any.
The phone is configured with the hostname serserver.domain.com as the
SIP server.
Any idea why this is happening?
(Apologies for the skimpiness of the details; I can follow up with
config information, models, etc, but I was hoping someone had seen this
before and had an idea).
--
Dan
I think there is a more appealing case, which is getting clients
of a public SIP service behind any NATs.
an internal SER version does exactly that along with rtpproxy
-- feel free to try it out with our public SIP service. It will
certainly show up in stable release some day.
-jiri
At 03:12 PM 10/29/2003, dhiraj.2.bhuyan(a)bt.com wrote:
>Hello List,
>
>SIP and NAT - this issue has been raised on numerous occassions. I finally got a small working demo for SIP Instant Messaging through a NAT gateway (next on the list is to try it with some RTP traffic). I am using "siproxd" http://sourceforge.net/projects/siproxd on the NAT gateway. Siproxd is an proxy/masquerading daemon for the SIP protocol. It allows SIP clients to work behind an IP masquerading firewall or router.
>
>Demo setup -
>
>[UA1] <------> [siproxd/NAT Gateway]<------->[SER]<------->[UA2]
>
>where UA1 is behind a NAT gateway, siproxd is running on the NAT gateway and UA2 is on the public side. Both UA1 and UA2 uses SER as the registration server in this demo (siproxd is also capable of working as a registration server). UA1 is configured to use siproxd as its outgoing proxy. All SIP packets going through siproxd are re-written to address the NAT traversal issues (so the SIP packets appear to be coming from the NAT gateway). I used Siemens SIP client for this demo.
>
>siproxd is also capable of proxying RTP traffic - although I am yet to test this feature.
>
>Many of us are looking for "siproxd" like utility that addressed the NAT traversal problems. And I think it will be a great idea to add such a capability to SER. In the meantime, it will be great if more people tests and contributes to siproxd.
>
>Thoughts anyone?
>
>Dhiraj Bhuyan
>Security Research Engineer
>BT Exact
>
>Tel: +44 1473 643932
>Mob: +44 7962 012145
>Email: dhiraj.2.bhuyan(a)bt.com
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
I am seeing 2 problems with nathelper and rtpproxy. One I have seen mentioned on the list but no solution or explainations that I can find.
1. I am seeing the following in my messages file
Oct 29 11:01:34 serv1 ser[9229]: ERROR: extract_mediaip: no `c=' in SDP
Oct 29 11:01:35 serv1 ser[9229]: ERROR: extract_mediaip: no `c=' in SDP
Oct 29 11:01:36 serv1 ser[9227]: ERROR: extract_mediaip: no `c=' in SDP
2. In some situations the BYE messages don't hangup the call
I am using yesterday unstable CVS tree. Client I am using in the test lab are Budgetone and x-lite softphones
and hear is my ser.cfg
Thanks
#
# $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
/* Uncomment these lines to enter debugging mode
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
*/
# /* Uncomment these lines to enter debugging mode
debug=10
fork=yes
log_stderror=no
# */
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"
alias=mydomain.dyndns.org
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
#loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
#loadmodule "/usr/local/lib/ser/modules/auth.so"
#loadmodule "/usr/local/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") {
force_rport();
fix_nated_contact();
save("location");
# Uncomment this if you want to use digest authentication
# if (!www_authorize("iptel.org", "subscriber")) {
# www_challenge("iptel.org", "0");
# break;
# };
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
#inserted by klaus
if (method=="INVITE") {
record_route();
force_rtp_proxy();
/* set up reply processing */
t_on_reply("1");
};
# 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();
};
}
#inserted by klaus
# all incoming replies for t_onrepli-ed transactions enter here
onreply_route[1] {
if (status=~"2[0-9][0-9]")
force_rtp_proxy();
}
i'm behind gprs service that doesn't allow any incoming udp packets or
establishment of inbound tcp connections. when i register, a tcp
connection is established between my sip ua and ser.
if there is an existing tcp connection open between sip ua and ser, is
ser supposed to reuse it for all incoming sip traffic, such as
subscribes and notifies?
-- juha
Hello List,
SIP and NAT - this issue has been raised on numerous occassions. I finally got a small working demo for SIP Instant Messaging through a NAT gateway (next on the list is to try it with some RTP traffic). I am using "siproxd" http://sourceforge.net/projects/siproxd on the NAT gateway. Siproxd is an proxy/masquerading daemon for the SIP protocol. It allows SIP clients to work behind an IP masquerading firewall or router.
Demo setup -
[UA1] <------> [siproxd/NAT Gateway]<------->[SER]<------->[UA2]
where UA1 is behind a NAT gateway, siproxd is running on the NAT gateway and UA2 is on the public side. Both UA1 and UA2 uses SER as the registration server in this demo (siproxd is also capable of working as a registration server). UA1 is configured to use siproxd as its outgoing proxy. All SIP packets going through siproxd are re-written to address the NAT traversal issues (so the SIP packets appear to be coming from the NAT gateway). I used Siemens SIP client for this demo.
siproxd is also capable of proxying RTP traffic - although I am yet to test this feature.
Many of us are looking for "siproxd" like utility that addressed the NAT traversal problems. And I think it will be a great idea to add such a capability to SER. In the meantime, it will be great if more people tests and contributes to siproxd.
Thoughts anyone?
Dhiraj Bhuyan
Security Research Engineer
BT Exact
Tel: +44 1473 643932
Mob: +44 7962 012145
Email: dhiraj.2.bhuyan(a)bt.com
Hi all,
I'm new to the list so sorry if this question has already been answered.
I have has a search through all the posts and all the info at iptel.org
but alas am still having a problem with my ser.
What I am trying to do is set ser up as an off net solution for our
current VOIP/SIP systems.
I have most of it nailed and have thus far found ser very easy to work
with the help great. I have now hit a bit of a road block.
I am trying to get the rtpproxy working with ser so that ALL rtp streams
go through the ser proxy and not from the off net phone directly to the
called phone. I have the rtpproxy and it runs fine (so far as I can
tell) but when trying to get ser to use it I keep getting an error in
the logs about loadmodule and unknown module, the error is: unknown
command, missing loadmodule?
This is being trigered by the line in the ser.cfg file force_rtp_proxy.
I used the example from the voip-info.org so am a little unsure of what
I have done wrong. Is there something else I need to do? There is little
documentation about rtpproxy...
I am running Debian on a 2.4.21 kernel and have used the ser .deb
packeges for the ser install.
Any help would be great.
Thanks,
Stephen
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listen=0.0.0.0 is not advicable -- it prints 0.0.0.0 in Via, which makes
many UASs unable to respond to it.
-jiri
At 10:48 PM 10/28/2003, David Quenzler wrote:
>SER Team,
>
>Working with two Windows Messenger clients that don not show as 'online' to
>each other using 'listen=0.0.0.0'
>
>On a two node cluster, I have SER running with listen=0.0.0.0 and the mysql
>module. One node has the clientIP, the other node is on warm standby.
>Clients connect to a movable IP, which SER picks up as needed
>
>Client wants to talk to his buddy
>+========================+
>Client sends a SUBSCRIBE notice to the SER server via the movable IP
>A different, valid, non-movable SER server IP sends the SUBSCRIBE to the
>buddy
>- the header record-route notes sip:user@0.0.0.0 and Via: 0.0.0.0
>
>How to ensure Windows Messenger clients show valid info re: their contact
>lists?
>
>Thanks,
>
>- Dave
>
>BluePages:
>http://bluepages.ibm.com/cgi-bin/bluepages.pl?Selection=Name&selectOn=quenz…
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
SER Team,
Working with two Windows Messenger clients that don not show as 'online' to
each other using 'listen=0.0.0.0'
On a two node cluster, I have SER running with listen=0.0.0.0 and the mysql
module. One node has the clientIP, the other node is on warm standby.
Clients connect to a movable IP, which SER picks up as needed
Client wants to talk to his buddy
+========================+
Client sends a SUBSCRIBE notice to the SER server via the movable IP
A different, valid, non-movable SER server IP sends the SUBSCRIBE to the
buddy
- the header record-route notes sip:user@0.0.0.0 and Via: 0.0.0.0
How to ensure Windows Messenger clients show valid info re: their contact
lists?
Thanks,
- Dave
BluePages:
http://bluepages.ibm.com/cgi-bin/bluepages.pl?Selection=Name&selectOn=quenz…
