[Serusers] Problem in SER->SIPH323 communication

Hi, I have this network configuration SIP Phone (ph.no: 3322) -> SER -> SIPH323 -> router -> Cisco CM ->Cisco Skinny phones(ph.no: 1133). Their corresponding IP address are 192.168.6.103 -> 192.168.6.100 (port:5060) -> 192.168.6.100 (port 5080 for sip side of SIPH323 converter) -> etc The problem arises in SER -> SIPH323 communication. SIPH323 doesnot get any SIP recognisable packet that SER sends. In the SER configuration, I check for uri sip:1133@192.168.6.100 and then I forward to port 5080 where SIPH323 listens for SIP side messages. I have attached the following: 1) My ser configuration 2)ngrep output on lo (loopback interface) 3) ngrep output on eth0. I would appreciate if you could tell me what is happening and how to solve this problem. I also would like to know if my SER configuartion is correct. I am sure SIPH323 is configured properly. If you see the ngrep on loopback interface, I could see a message going from 192.168.6.100 -> 192.168.6.100 3:3. This message has some weird characters before the INVITE message. I guess SIPH323 gets this message and doenot respond to it because it is not a valid SIP Invite message. I would like to know why this happens. ATTACHED FILES: 1) SER.cfg . . 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 ( len_gt(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 (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; if(uri=~"^sip:11[0-9]*@192.168.6.100") { forward(192.168.6.100, 5080); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # 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(); }; } }; # 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) 2) ngrep on lo (loopback interface): interface: lo (127.0.0.0/255.0.0.0) # U 192.168.6.100:5060 -> 192.168.6.100:5080 INVITE sip:1133@192.168.6.100 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:1133@192.168.6.100;ftag=925f0b001c06323a74 788-57c59f3f;lr=on>..Via: SIP/2.0/UDP 192.168.6.100;branch= 0..Via: SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100>;tag=925f0b001c06323a74788-57c5 9f3f..To: <sip:1133@192.168.6.100>..Call-ID: 000b5f92-63c00 003-69765ab4-289e7e69@192.168.6.103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 INVITE..Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: application/sdp..Contact: sip :3322@192.168.6.103:5060..Content-Type: application/sdp..Co ntent-Length: 225....v=0..o=CiscoSystemsSIP-IPPhone-UserAge nt 11416 8413 IN IP4 192.168.6.103..s=SIP Call..c=IN IP4 19 2.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0 8 18 101..a=rtp map:0 pcmu/8000..a=rtpmap:101 telephone-event/8000..a=fmtp: 101 0-11.. # I 192.168.6.100 -> 192.168.6.100 3:3 ....E..`..@.@..d...d...d.....L.7INVITE sip:1133@192.168.6.1 00 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:1133@192.1 68.6.100;ftag=925f0b001c06323a74788-57c59f3f;lr=on>..Via: S IP/2.0/UDP 192.168.6.100;branch=0..Via: SIP/2.0/UDP 192.168 .6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100 .6.100>..Call-ID: 000b5f92-63c00003-69765ab4-289e7e69(a)192.1 68.6.103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 IN VITE..Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accep t: application/sdp..C # U 192.168.6.100:5060 -> 192.168.6.100:5080 INVITE sip:1133@192.168.6.100 SIP/2.0..Max-Forwards: 10..Re cord-Route: <sip:1133@192.168.6.100;ftag=925f0b001c06323a74 788-57c59f3f;lr=on>..Via: SIP/2.0/UDP 192.168.6.100;branch= 0..Via: SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Bala ji" <sip:3322@192.168.6.100>;tag=925f0b001c06323a74788-57c5 9f3f..To: <sip:1133@192.168.6.100>..Call-ID: 000b5f92-63c00 003-69765ab4-289e7e69@192.168.6.103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 INVITE..Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: application/sdp..Contact: sip :3322@192.168.6.103:5060..Content-Type: application/sdp..Co ntent-Length: 225....v=0..o=CiscoSystemsSIP-IPPhone-UserAge nt 11416 8413 IN IP4 192.168.6.103..s=SIP Call..c=IN IP4 19 2.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0 8 18 101..a=rtp map:0 pcmu/8000..a=rtpmap:101 telephone-event/8000..a=fmtp: 101 0-11.. ...It repeats the same messages 3) ngrep on eth0: interface: eth0 (192.168.6.0/255.255.255.0) ############### U 192.168.6.103:49834 -> 192.168.6.100:5060 INVITE sip:1133@192.168.6.100 SIP/2.0..Via: SIP/2.0/UDP 192.1 68.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100 .100>..Call-ID: 000b5f92-63c00003-69765ab4-289e7e69(a)192.168.6 .103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 INVITE.. Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: appli cation/sdp..Contact: sip:3322@192.168.6.103:5060..Content-Typ e: application/sdp..Content-Length: 225....v=0..o=CiscoSystem sSIP-IPPhone-UserAgent 11416 8413 IN IP4 192.168.6.103..s=SIP Call..c=IN IP4 192.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0 8 18 101..a=rtpmap:0 pcmu/8000..a=rtpmap:101 telephone-event /8000..a=fmtp:101 0-11.. # U 192.168.6.103:49834 -> 192.168.6.100:5060 INVITE sip:1133@192.168.6.100 SIP/2.0..Via: SIP/2.0/UDP 192.1 68.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100 .100>..Call-ID: 000b5f92-63c00003-69765ab4-289e7e69(a)192.168.6 .103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 INVITE.. Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: appli cation/sdp..Contact: sip:3322@192.168.6.103:5060..Content-Typ e: application/sdp..Content-Length: 225....v=0..o=CiscoSystem sSIP-IPPhone-UserAgent 11416 8413 IN IP4 192.168.6.103..s=SIP Call..c=IN IP4 192.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0 8 18 101..a=rtpmap:0 pcmu/8000..a=rtpmap:101 telephone-event /8000..a=fmtp:101 0-11.. # U 192.168.6.103:49834 -> 192.168.6.100:5060 INVITE sip:1133@192.168.6.100 SIP/2.0..Via: SIP/2.0/UDP 192.1 68.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100 .100>..Call-ID: 000b5f92-63c00003-69765ab4-289e7e69(a)192.168.6 .103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 INVITE.. Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: appli cation/sdp..Contact: sip:3322@192.168.6.103:5060..Content-Typ e: application/sdp..Content-Length: 225....v=0..o=CiscoSystem sSIP-IPPhone-UserAgent 11416 8413 IN IP4 192.168.6.103..s=SIP Call..c=IN IP4 192.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0 8 18 101..a=rtpmap:0 pcmu/8000..a=rtpmap:101 telephone-event /8000..a=fmtp:101 0-11.. ### ...it repeats these messages Thanks, Balaji