looks like the LCS doesn't find a matching entry in its "routing table".
regards, klaus
Mimmus wrote:
Hi, I'm setting up SER to act as a proxy between Asterisk and Microsoft LCS (to enable TCP2UDP translation needed to speak with Microsoft LCS). I'd like to keep configuration the simplest as possible. Unfortunately, LCS refuses INVITE with a: <504: Server Timeout> as you can see in following output from 'ngrep' (some security sanityzing performed):
####### T xxx.yyy.64.118:32834 -> 10.99.1.9:5060 [AP] INVITE sip:myuser@mydomain.it SIP/2.0. Max-Forwards: 10. Record-Route: sip:myuser@xxx.yyy.64.118;transport=tcp;r2=on;ftag=as12c76469;l r=on. Record-Route: sip:myuser@xxx.yyy.64.118;r2=on;ftag=as12c76469;lr=on. Via: SIP/2.0/TCP xxx.yyy.64.118;branch=z9hG4bK1988.c780cdd1.0. Via: SIP/2.0/UDP xxx.yyy.64.118:5061;branch=z9hG4bK184e7f66. From: "01234564" sip:01234564@xxx.yyy.64.118:5061;tag=as12c76469. To: sip:myuser@xxx.yyy.64.118. Contact: sip:01234564@xxx.yyy.64.118:5061. Call-ID: 3b3a9d59133196ef4dfa27836e47cea9@xxx.yyy.64.118. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Tue, 29 Mar 2005 17:34:32 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Content-Type: application/sdp. Content-Length: 267. . v=0. o=root 14974 14974 IN IP4 xxx.yyy.64.118. s=session. c=IN IP4 xxx.yyy.64.118. t=0 0. m=audio 11980 RTP/AVP 8 3 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -.
# T 10.99.1.9:5060 -> xxx.yyy.64.118:32834 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.yyy.64.118;branch=z9hG4bK1988.c780cdd1.0;ms-received-port=3 2834;ms-received-cid=1d00. Via: SIP/2.0/UDP xxx.yyy.64.118:5061;branch=z9hG4bK184e7f66. From: "01234564" sip:01234564@xxx.yyy.64.118:5061;tag=as12c76469. To: sip:myuser@xxx.yyy.64.118. Call-ID: 3b3a9d59133196ef4dfa27836e47cea9@xxx.yyy.64.118. CSeq: 102 INVITE. Content-Length: 0. .
## T 10.99.1.9:5060 -> xxx.yyy.64.118:32834 [AP] SIP/2.0 504 Server time-out. Via: SIP/2.0/TCP xxx.yyy.64.118;branch=z9hG4bK1988.c780cdd1.0;ms-received-port=3 2834;ms-received-cid=1d00. Via: SIP/2.0/UDP xxx.yyy.64.118:5061;branch=z9hG4bK184e7f66. From: "01234564" sip:01234564@xxx.yyy.64.118:5061;tag=as12c76469. To: sip:myuser@xxx.yyy.64.118;tag=9B5EAEFA3B5764273F459C938E732306. Call-ID: 3b3a9d59133196ef4dfa27836e47cea9@xxx.yyy.64.118. CSeq: 102 INVITE. Content-Length: 0. . #################################################################
Logs from LCS shows this message:
Text: Unable to route the request Result-Code: 0xc3e91002 SIP-Start-Line: INVITE sip:myuser@mydomain.it SIP/2.0 SIP-Call-ID: 572410cc05c62bfe720cbc4c6cde6b0b@xxx.yyy.64.118 SIP-CSeq: 102 INVITE Data: Non-trusted source sent an FQDN/IP that doesn't match a routing table rule ##################################################################
This is my ser.cfg:
debug=9 # debug level (cmd line: -dddddddddd) log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=xxx.yyy.64.118 alias="mydomain.it" alias="xxx.yyy.64.118" fifo="/tmp/ser_fifo" 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" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) 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; }; # reject REGISTER attempts for now if (method=="REGISTER") { sl_send_reply( "503", "Registration Unavailable" ); break; }; # Forward to LCS if text usernames if (uri =~ "sip:[a-zA-Z\.\_]*@*") { log(1, "Forwarding to LCS\n"); rewritehostport("mydomain.it"); # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay_to_tcp("eco.pitagora.it", "5060")) { sl_reply_error(); }; break; } # Forward to Asterisk if numerical username if (method == "INVITE") { if (uri =~ "sip:9[0-9]{10}@*") { log(1, "Forwarding to Asterisk\n"); rewriteport("5061"); # 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(); }; break; } } sl_send_reply("404", "Not Found");} ##############################################################
Any help?
Thanks in advance...
Domenico Viggiani _______________________________________________ Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers