<DIV>Greger,</DIV>
<DIV> </DIV>
<DIV>Since fix_nated_register does not exist with 0.8.14, will fix_nated_contact do instead? Also if I am leaving out the has_totag() at the start of the script, will this greatly effect its functionality?</DIV>
<DIV> </DIV>
<DIV>Thank you,</DIV>
<DIV>Vivienne<BR><BR><B><I>"Greger V. Teigre" <greger@teigre.com></I></B> wrote:</DIV>
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">
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<DIV>Vivienne,</DIV>
<DIV> </DIV>
<DIV>This is the first INVITE going from SER to your public phone. I have prefixed my comments with *==></DIV>
<DIV> </DIV>
<DIV>U 84.203.148.146:5060 -> 157.190.74.151:5060</DIV>
<DIV> INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP 84.203.148.146;branch=z9hG4bK77bc.b54ca216.0..</DIV>
<DIV>Via: SIP/2.0/UDP 172.16.3.31;rport=5060;received=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..</DIV>
<DIV>From: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..</DIV>
<DIV>To: <sip:2092@84.203.148.146>..</DIV>
<DIV>Contact: <sip:2093@84.203.148.14:5060>..</DIV>
<DIV>*==> Correctly changed to the public address and port of 2093</DIV>
<DIV>Supported: replaces..Call-ID: 44e1ae63c476fbf<A href="mailto:44e1ae63c476fbff@172.16.3.31..CSeq">f@172.16.3.31..CSeq</A>: 10327 INVITE..</DIV>
<DIV>User-Agent: Grandstream BT100 1.0.5.18..</DIV>
<DIV>Max-Forwards: 69..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..</DIV>
<DIV>Content-Type: application/sdp..</DIV>
<DIV>Content-Length: 443....v=0..</DIV>
<DIV>o=2093 8000 0 IN IP4 172.16.3.31..</DIV>
<DIV>s=SIP Call..</DIV>
<DIV>c=IN IP4 84.203.148.1484.203.148.146..</DIV>
<DIV>*==> Here it seems that first fix_nated_sdp("3") is called (replace private IP with public IP), then force_rtp_proxy() (replace with proxy IP). You must only call one of them (fix_nated_sdp("1") is ok)</DIV>
<DIV>t=0 0..</DIV>
<DIV>m=audio 35006 RTP/AVP 0 8 4 18 2 15 99 9..</DIV>
<DIV>a=sendrecv..</DIV>
<DIV>a=rtpmap:0 PCMU/8000/3..</DIV>
<DIV>a=rtpmap:8 PCMA/8000/3..</DIV>
<DIV>a=rtpmap:4 G723/8000/3..a=rtpmap:18 G729/8000/3..</DIV>
<DIV>a=rtpmap:2 G726-32/8000/3..</DIV>
<DIV>a=rtpmap:15 G728/8000/3..</DIV>
<DIV>a=rtpmap:99 iLBC/8000/3..</DIV>
<DIV>a=fmtp:99 mode=20..</DIV>
<DIV>a=rtpmap:9 G722/8000/3..</DIV>
<DIV>a=ptime:20..</DIV>
<DIV>a=direction:active..</DIV>
<DIV>*==> Added by fix_nated_sdp("1") (and "3")</DIV>
<DIV>a=oldmediaip:172.16.3.31..</DIV>
<DIV>
<DIV>*==> Added by fix_nated_sdp("3")</DIV>
<DIV>a=nortpproxy:yes..</DIV></DIV>
<DIV>*==> Added by force_rtp_proxy()</DIV>
<DIV> </DIV>
<DIV>So, your call is proxied. Try using fix_nated_sdp("1"). It should make the INVITE correct. You have not posted the OK, so I don't know what is happening there. If you have followed the rtpproxy ONsip.org example in your onreply_route, you should be fine.</DIV>
<DIV> </DIV>
<DIV>Good luck!</DIV>
<DIV>g-)</DIV>
<DIV> </DIV>
<DIV>---- Original Message ----<BR>From: Vivienne Curran<BR>To: serusers@lists.iptel.org<BR>Sent: Tuesday, April 05, 2005 01:23 PM<BR>Subject: [Serusers] Contact Header and SDP not rewritten<BR><BR>> Hello,<BR>> <BR>> I have a problem whereby when a private client rings a public client<BR>> only the public user can hear voice and when the public user rings<BR>> the private user, no audio is transmitted. After looking at the<BR>> messages I have have determined that the contact header and sdp part<BR>> of the invite contains the private address of the natted client. This<BR>> would lead me to believe that the registration process is incorrect.<BR>> My problem is that I believe my script should handle the registration<BR>> process correctly and I suspect that the following code is being<BR>> skipped: I tried changing it to nat_uac_test("19") and<BR>> fix_nated_sdp("3") but that didnt help. <BR>> <BR>>
if (nat_uac_test("3")){<BR>> if (method == "REGISTER" || ! search("^Record-Route:")){<BR>> log("Log: Someone trying to register from private IP,rewriting\n");<BR>> fix_nated_contact(); #Rewrite contact with source IP<BR>> if (method == "INVITE"){<BR>> fix_nated_sdp("1"); #Add direction=active to SDP<BR>> force_rtp_proxy();<BR>> };<BR>> force_rport(); # Add rport parameter to topmost Via<BR>> setflag(6); # Mark as Nated<BR>> };<BR>> };<BR>> <BR>> I have included the log message below and my ser.cfg as an<BR>> attachment. Please let me know where I could be going wrong. <BR>> <BR>> Thank you,<BR>> Vivienne.<BR>> <BR>> 2093 (private) ringing 2092 (public)<BR>> <BR>> U 84.203.148.14:5060 -> 84.203.148.146:5060<BR>> INVITE
sip:2092@84.203.148.146 SIP/2.0..Via: SIP/2.0/UDP<BR>> 172.16.3.31;branch <BR>> =z9hG4bK1a48edc121f5bc1f..From: "2093"<BR>> <sip:2093@84.203.148.146>;tag=2dc376 <BR>> dcd4655094..To: <sip:2092@84.203.148.146>..Contact:<BR>> <sip:2093@172.16.3.31>. <BR>> .Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq:<BR>> 10327 IN <BR>> VITE..User-Agent: Grandstream BT100 1.0.5.18..Max-Forwards:<BR>> 70..Allow: INVI <BR>> <BR>> TE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:<BR>> applic <BR>> ation/sdp..Content-Length: 362....v=0..o=2093 8000 0 IN IP4<BR>> 172.16.3.31..s= <BR>> SIP Call..c=IN IP4 172.16.3.31..t=0 0..m=audio 5004 RTP/AVP 0 8 4<BR>> 18 2 15 9 <BR>> 9 9..a=sendrecv..a=rtpmap:0 PCMU/8000/3..a=rtpmap:8<BR>> PCMA/8000/3..a=rtpmap:4 <BR>> G723/8000/3..a=rtpmap:18
G729/8000/3..a=rtpmap:2<BR>> G726-32/8000/3..a=rtpmap: <BR>> 15 G728/8000/3..a=rtpmap:99 iLBC/8000/3..a=fmtp:99<BR>> mode=20..a=rtpmap:9 G722 <BR>> /8000/3..a=ptime:20..<BR>> <BR>> U 84.203.148.146:5060 -> 84.203.148.14:5060<BR>> SIP/2.0 100 trying -- your call is important to us..Via:<BR>> SIP/2.0/UDP 172.16 <BR>> <BR>> .3.31;branch=z9hG4bK1a48edc121f5bc1f;rport=5060;received=84.203.148.14..Fro<BR>> m: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..To:<BR>> <sip:2092@84. <BR>> 203.148.146>..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq: 10327<BR>> INVITE..Se <BR>> rver: Sip EXpress router (0.8.14 (i386/linux))..Content-Length:<BR>> 0..Warning: <BR>> 392 84.203.148.146:5060 "Noisy feedback tells: pid=4732<BR>> req_src_ip=84.203 <BR>> .148.14 req_src_port=5060
in_uri=sip:2092@84.203.148.146<BR>> out_uri=sip:2092@1 <BR>> 57.190.74.151 via_cnt==1"....<BR>> <BR>> U 84.203.148.146:5060 -> 157.190.74.151:5060<BR>> INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP<BR>> 84.203.148.146;bra <BR>> nch=z9hG4bK77bc.b54ca216.0..Via: SIP/2.0/UDP<BR>> 172.16.3.31;rport=5060;receive <BR>> d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"<BR>> <sip:2093@84.2 <BR>> 03.148.146>;tag=2dc376dcd4655094..To:<BR>> <sip:2092@84.203.148.146>..Contact: < <BR>> sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:<BR>> 44e1ae63c476fbf <BR>> f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100<BR>> 1.0.5.18.. <BR>> Max-Forwards: 69..Allow:<BR>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU <BR>> BSCRIBE..Content-Type:
application/sdp..Content-Length:<BR>> 443....v=0..o=2093 <BR>> 8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4<BR>> 84.203.148.1484.203.148.146 <BR>> ..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99<BR>> 9..a=sendrecv..a=rtpmap:0 P <BR>> CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4<BR>> G723/8000/3..a=rtpmap:18 G72 <BR>> 9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15<BR>> G728/8000/3..a=rtpmap:99 i <BR>> LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9<BR>> G722/8000/3..a=ptime:20..a=direct <BR>> ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..<BR>> <BR>> U 84.203.148.146:5060 -> 157.190.74.150:5060<BR>> INVITE sip:2092@157.190.74.150 SIP/2.0..Via: SIP/2.0/UDP<BR>> 84.203.148.146;bra <BR>> nch=z9hG4bK77bc.b54ca216.1..Via: SIP/2.0/UDP<BR>> 172.16.3.31;rport=5060;receive <BR>>
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"<BR>> <sip:2093@84.2 <BR>> 03.148.146>;tag=2dc376dcd4655094..To:<BR>> <sip:2092@84.203.148.146>..Contact: < <BR>> sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:<BR>> 44e1ae63c476fbf <BR>> f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100<BR>> 1.0.5.18.. <BR>> Max-Forwards: 69..Allow:<BR>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU <BR>> BSCRIBE..Content-Type: application/sdp..Content-Length:<BR>> 443....v=0..o=2093 <BR>> 8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4<BR>> 84.203.148.1484.203.148.146 <BR>> ..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99<BR>> 9..a=sendrecv..a=rtpmap:0 P <BR>> CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4<BR>> G723/8000/3..a=rtpmap:18 G72 <BR>> 9/8000/3..a=rtpmap:2
G726-32/8000/3..a=rtpmap:15<BR>> G728/8000/3..a=rtpmap:99 i <BR>> LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9<BR>> G722/8000/3..a=ptime:20..a=direct <BR>> ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..<BR>> <BR>> U 157.190.74.151:5060 -> 84.203.148.146:5060<BR>> SIP/2.0 100 Trying..Via: SIP/2.0/UDP<BR>> 84.203.148.146;branch=z9hG4bK77bc.b54c <BR>> a216.0..Via: SIP/2.0/UDP<BR>> 172.16.3.31;rport=5060;received=84.203.148.14;bran <BR>> ch=z9hG4bK1a48edc121f5bc1f..From: "2093"<BR>> <sip:2093@84.203.148.146>;tag=2dc3 <BR>> 76dcd4655094..To: <sip:2092@84.203.148.146>..Call-ID:<BR>> 44e1ae63c476fbff@172. <BR>> 16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100<BR>> 1.0.5.18..Conten <BR>> t-Length: 0....<BR>> <BR>> U 157.190.74.151:5060 -> 84.203.148.146:5060<BR>> SIP/2.0 180 Ringing..Via:
SIP/2.0/UDP<BR>> 84.203.148.146;branch=z9hG4bK77bc.b54 <BR>> ca216.0..Via: SIP/2.0/UDP<BR>> 172.16.3.31;rport=5060;received=84.203.148.14;bra <BR>> nch=z9hG4bK1a48edc121f5bc1f..From: "2093"<BR>> <sip:2093@84.203.148.146>;tag=2dc <BR>> 376dcd4655094..To:<BR>> <sip:2092@84.203.148.146>;tag=10bdf2044401a257..Call-ID: <BR>> 44e1ae63c476fbff@172.16.3.31..CSeq: 10327 INVITE..User-Agent:<BR>> Grandstream <BR>> BT100 1.0.5.18..Content-Length: 0....<BR>> <BR>> Send instant messages to your online friends<BR>> http://uk.messenger.yahoo.com <BR>> <BR>> <BR>> <BR>> #<BR>> # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $<BR>> #<BR>> # simple quick-start config script<BR>> #<BR>> <BR>> # ----------- global configuration parameters ------------------------<BR>> <BR>> #debug=3 # debug level
(cmd line: -dddddddddd)<BR>> #fork=yes<BR>> #log_stderror=no # (cmd line: -E)<BR>> <BR>> /* Uncomment these lines to enter debugging mode<BR>> debug=7<BR>> fork=no<BR>> log_stderror=yes<BR>> */<BR>> <BR>> check_via=no # (cmd. line: -v)<BR>> dns=no # (cmd. line: -r)<BR>> rev_dns=no # (cmd. line: -R)<BR>> #port=5060<BR>> #children=4<BR>> fifo="/tmp/ser_fifo"<BR>> <BR>> alias="157.190.74.152:5060"<BR>> <BR>> # ------------------ module loading ----------------------------------<BR>> <BR>> # Uncomment this if you want to use SQL database<BR>> loadmodule "/usr/lib/ser/modules/mysql.so"<BR>> <BR>> loadmodule "/usr/lib/ser/modules/sl.so"<BR>> loadmodule "/usr/lib/ser/modules/tm.so"<BR>> loadmodule "/usr/lib/ser/modules/rr.so"<BR>> loadmodule "/usr/lib/ser/modules/maxfwd.so"<BR>> loadmodule
"/usr/lib/ser/modules/usrloc.so"<BR>> loadmodule "/usr/lib/ser/modules/registrar.so"<BR>> loadmodule "/usr/lib/ser/modules/textops.so"<BR>> loadmodule "/usr/lib/ser/modules/nathelper.so"<BR>> #loadmodule "/usr/lib/ser/modules/pa.so"<BR>> loadmodule "/usr/lib/ser/modules/cpl-c.so"<BR>> <BR>> # Uncomment this if you want digest authentication<BR>> # mysql.so must be loaded !<BR>> loadmodule "/usr/lib/ser/modules/auth.so"<BR>> loadmodule "/usr/lib/ser/modules/auth_db.so"<BR>> <BR>> # ----------------- setting module-specific parameters ---------------<BR>> <BR>> # -- usrloc params --<BR>> <BR>> #modparam("usrloc", "db_mode", 0)<BR>> <BR>> # Uncomment this if you want to use SQL database<BR>> # for persistent storage and comment the previous line<BR>> modparam("usrloc", "db_mode", 2)<BR>> <BR>> # -- auth params --<BR>> # Uncomment if you are using auth module<BR>> #<BR>> modparam("auth_db",
"calculate_ha1", yes)<BR>> #<BR>> # If you set "calculate_ha1" parameter to yes (which true in this<BR>> config), <BR>> # uncomment also the following parameter)<BR>> #<BR>> modparam("auth_db", "password_column", "password")<BR>> <BR>> # -- rr params --<BR>> # add value to ;lr param to make some broken UAs happy<BR>> #NB Had to up this value from 1 to 11 because reinvites were<BR>> bombarding called phone <BR>> modparam("rr", "enable_full_lr", 11)<BR>> <BR>> #!! Nathelper<BR>> modparam("registrar", "nat_flag", 60)<BR>> modparam("nathelper", "natping_interval", 30) #Ping interval 30 s<BR>> modparam("nathelper", "ping_nated_only", 1) #Ping only clients<BR>> behind NAT <BR>> modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock")<BR>> <BR>> modparam("tm", "fr_inv_timer", 20)<BR>> <BR>> #modparam("pa", "default_expires", 3600)<BR>> <BR>> modparam("usrloc", "db_url",
"sql://root:1alfa156@localhost/ser")<BR>> <BR>> #modparam("cpl-c", "cpl_db", "mysql://root:1alfa156@localhost/ser")<BR>> #modparam("cpl-c", "cpl_table", "cpl")<BR>> #modparam("cpl-c", "cpl_dtd_file",<BR>> "/work/sip_router/modules/cpl-ccpl-06.dtd") <BR>> <BR>> # ------------------------- request routing logic -------------------<BR>> <BR>> # main routing logic<BR>> <BR>> route{<BR>> <BR>> # initial sanity checks -- messages with<BR>> # max_forwards==0, or excessively long requests<BR>> if (!mf_process_maxfwd_header("10")) {<BR>> sl_send_reply("483","Too Many Hops");<BR>> break;<BR>> };<BR>> if ( msg:len > max_len ) {<BR>> sl_send_reply("513", "Message too big");<BR>> break;<BR>> };<BR>> <BR>> #######################################################################################<BR>> if (nat_uac_test("3")){<BR>> if (method == "REGISTER" || ! search("^Record-Route:")){<BR>> log("Log: Someone
trying to register from private IP,rewriting\n");<BR>> fix_nated_contact(); #Rewrite contact with source IP<BR>> if (method == "INVITE"){<BR>> fix_nated_sdp("1"); #Add direction=active to SDP<BR>> force_rtp_proxy();<BR>> };<BR>> force_rport(); # Add rport parameter to topmost Via<BR>> setflag(6); # Mark as Nated<BR>> };<BR>> };<BR>> ########################################################################################<BR>> <BR>> # we record-route all messages -- to make sure that<BR>> # subsequent messages will go through our proxy; that's<BR>> # particularly good if upstream and downstream entities<BR>> # use different transport protocol<BR>> <BR>> if (method =="REGISTER") record_route();<BR>> <BR>> # loose-route processing<BR>> if (loose_route()) {<BR>> #commented 11/02/05<BR>> #t_relay();<BR>> route(1);<BR>> break;<BR>> };<BR>> <BR>> # if the request is for other domain use UsrLoc<BR>> # (in
case, it does not work, use the following command<BR>> # with proper names and addresses in it)<BR>> if (uri==myself) {<BR>> <BR>> log(1,"into loop");<BR>> if (method=="REGISTER") {<BR>> <BR>> # Uncomment this if you want to use digest authentication<BR>> # if (!www_authorize("157.190.74.152", "subscriber")) {<BR>> # www_challenge("157.190.74.152", "0");<BR>> # break;<BR>> # };<BR>> <BR>> #cpl_process_register();<BR>> <BR>> save("location");<BR>> break;<BR>> };<BR>> <BR>> # if (method=="SUBSCRIBE")<BR>> # {<BR>> # log(1, "Subscribe\n");<BR>> # if(t_newtran())<BR>> # {<BR>> # log(1, "Registrar\n");<BR>> # handle_subscription("registrar");<BR>> # };<BR>> # break;<BR>> # };<BR>> <BR>> lookup("aliases");<BR>> if (!uri==myself) {<BR>> append_hf("P-hint: outbound alias\r\n");<BR>> route(1);<BR>> break;<BR>> };<BR>> <BR>> if (method=="INVITE"){<BR>> <BR>> log(1,"in invite
loop");<BR>> #break; #no 100 trying<BR>> #if (!cpl_run_script("incoming","is_stateless"))<BR>> #{<BR>> # #script execution failed<BR>> # t_reply("500", "CPL script execution failed");<BR>> #};<BR>> <BR>> t_on_failure("1");<BR>> };<BR>> <BR>> # native SIP destinations are handled using our USRLOC DB<BR>> if (!lookup("location")) {<BR>> #sl_send_reply("404", "Not Found");<BR>> route(2);<BR>> break;<BR>> };<BR>> };<BR>> <BR>> # forward to current uri now; use stateful forwarding; that<BR>> # works reliably even if we forward from TCP to UDP<BR>> #commented 11/02/05#######################<BR>> if (!t_relay()) {<BR>> sl_reply_error();<BR>> };<BR>> <BR>> #route(1);<BR>> }<BR>> <BR>> ######################################entered<BR>> 11/02/05############################################################ <BR>> route[1]<BR>> {<BR>> #!!Nathelper<BR>>
if(uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&<BR>> !search("^Route:")){ <BR>> sl_send_reply("479", "We don't forward to private IP addresses");<BR>> break;<BR>> };<BR>> <BR>> if (isflagset(6)){<BR>> force_rtp_proxy();<BR>> }<BR>> <BR>> t_on_reply("1");<BR>> <BR>> if(!t_relay()){<BR>> sl_reply_error();<BR>> break;<BR>> };<BR>> <BR>> }<BR>> <BR>> ######################################entered<BR>> 11/02/05############################################################ <BR>> #!! Nathelper<BR>> onreply_route[1] {<BR>> if(isflagset(6) && status =~ "(183)|2[0-9][0-9]"){<BR>> fix_nated_contact();<BR>> force_rtp_proxy();<BR>> }else if (nat_uac_test("1")){<BR>> fix_nated_contact();<BR>> };<BR>> }<BR>> #################################################################################################################<BR>> <BR>> # ------------- handling of unavailable
user ------------------<BR>> route[2] {<BR>> <BR>> # non-Voip -- just send "off-line"<BR>> if (!(method == "INVITE" || method == "ACK" || method ==<BR>> "CANCEL")) { <BR>> sl_send_reply("404", "Not Found");<BR>> break;<BR>> };<BR>> <BR>> # forward to voicemail now<BR>> rewritehostport("157.190.74.152:5062");<BR>> t_relay_to_udp("157.190.74.152", "5062");<BR>> }<BR>> <BR>> # if forwarding downstream did not succeed, try voicemail running<BR>> # at
172.16.2.120:5062<BR>> <BR>> failure_route[1] {<BR>> revert_uri();<BR>> rewritehostport("157.190.74.152:5062");<BR>> append_branch();<BR>> t_relay_to_udp("157.190.74.152", "5062");<BR>> }<BR>> <BR>> <BR>> <BR>> <BR>> <BR>> _______________________________________________<BR>> Serusers mailing list<BR>> serusers@lists.iptel.org<BR>> http://lists.iptel.org/mailman/listinfo/serusers</DIV></BLOCKQUOTE><p>Send instant messages to your online friends http://uk.messenger.yahoo.com