Thanks for your help, Yes I have this: ACK 0156395148(a)80.118.128.1 Route 0156395148(a)80.118.128.5 When the request ACK is the CONTACT, is loose routing no??? and the call is cut after 20s or there are no voice. But with a Cisco IP PHONE 7940, I have this: ACK 0156395148(a)80.118.128.5 Route 0156395148(a)80.118.128.5 The request ACK is the Route, strict routing no ??? and the call is perfect. Can I put the iptel on strict routing only??? Thanks so much Nicolas RUIZ -----Message d'origine----- De : Martin Koenig [mailto:martin.koenig@toplink-plannet.de] Envoyé : mardi 7 septembre 2004 15:50 À : 'Nicolas RUIZ' Cc : serusers(a)lists.iptel.org Objet : RE: [Serusers] LOOSE routing problem : helppppppppp Hi there, What exactly is the problem? Everything is as it should be. See RFC3261 section 12.2.1.1 for further information: ACK 0156395148(a)80.118.128.1 Route 0156395148(a)80.118.128.5 Is what you should see. Verify with ngrep or something similar. The Loose Route processing of the proxy removes the route header field and forwards then to the Request-URI. Regards, Martin ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Nicolas RUIZ Sent: Tuesday, September 07, 2004 11:52 AM To: serusers(a)lists.iptel.org Subject: [Serusers] LOOSE routing problem : helppppppppp Hi, I have a problem with loose routing and strict routing. - I have a proxy-registrar SER 0.8.12, it says that Iptel is compliant RFC 3261 - I have a CISCO ATA with code 3.1.0 and its work. - I have a CISCO ATA with code 3.1.1 and when the ACK is send to Iptel the request line is ACK : 0156395148(a)80.118.128.1 And 80.118.128.1 is my gateway Cisco AS5300 to the PSTN, - With the ATA 3.1.0, the ACK is : 0156395148(a)80.118.128.5. And 80.118.128.5 is IPTEL ser. In Cisco Web pages, he says that the new version on ATA 3.1.1 is compliant RFC 3261with loose routing. http://www.cisco.com/en/US/products/hw/gatecont/ps514/prod_release_note09186 a0080238c0a.html Is someone can explain what is not good? Thanks in advance for your help. NICOLAS RUIZ VIVVACTION FRANCE, PARIS THAT'S my conf of SER: # 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; }; /* grant Route routing if route headers present */ if (loose_route()) { t_relay(); break; }; /* record-route INVITEs -- all subsequent requests must visit us */ if (method=="INVITE") { record_route(); }; # now check if it really is a PSTN destination which should be handled # by our gateway; if not, and the request is an invitation, drop it -- # we cannot terminate it in PSTN; relay non-INVITE requests -- it may # be for example BYEs sent by gateway to call originator if (uri=~"sip.vivaction.net"){ if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("sip.vivaction.net", "subscriber")) { www_challenge("sip.vivaction.net", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if (!proxy_authorize("sip.vivaction.net", "subscriber")) { proxy_challenge("sip.vivaction.net", "1"); break; }; rewritehostport("80.118.128.6:5060"); }; # forward the request now if (!t_relay()) { sl_reply_error(); break; }; }; }