<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Yes, FS(13263) send out 482 request merged to my voip client.<div><br></div><div>I guess, for some reason, FS doesn't respond to the REGISTER, and when the client start REGISTER again using another call-id, it merged the request to one. Anyone ever met this before? See the call-id and cseq below :</div><div><br></div><div><div>recv 631 bytes from udp/[69.131.94.250]:3270 at 12:40:32.811280:</div><div>REGISTER <a href="sip:voip.xxx.com">sip:voip.xxx.com</a> SIP/2.0</div><div>CSeq: 208 REGISTER</div><div>Content-Length: 0</div><div><br></div><div><br></div></div><div><div>recv 631 bytes from udp/[69.131.94.250]:3270 at 12:40:38.814237:</div><div>REGISTER <a href="sip:voip.xxx.com">sip:voip.xxx.com</a> SIP/2.0</div><div>CSeq: 210 REGISTER</div><div><br></div><div><br></div></div><div><div>recv 631 bytes from udp/[69.131.94.250]:3270 at 12:40:48.821027:</div><div>REGISTER <a href="sip:voip.xxx.com">sip:voip.xxx.com</a> SIP/2.0</div><div>Via: SIP/2.0/UDP 192.168.1.100:3270;rport;branch=z9hG4bKa9b7ba70783b617e9998dc4dd82eb3c5</div><div>From: <<a href="sip:cc@voip.xxx.com:5090">sip:cc@voip.xxx.com:5090</a>>;tag=a9b7ba70783b617e9998dc4dd82eb3c5</div><div>To: "cc" <<a href="sip:cc@voip.xxx.com:5090">sip:cc@voip.xxx.com:5090</a>></div><div>Call-ID: <a href="mailto:a9b7ba70783b617e9998dc4dd82eb3c5@192.168.1.100">a9b7ba70783b617e9998dc4dd82eb3c5@192.168.1.100</a></div><div>CSeq: 214 REGISTER</div><div>Contact: <<a href="sip:cc@192.168.1.100:3270;rinstance=1242647429">sip:cc@192.168.1.100:3270;rinstance=1242647429</a>></div><div>max-forwards: 70</div><div>expires: 300</div><div>Content-Length: 0</div><div><br></div><div><br></div></div><div><div>recv 629 bytes from udp/[69.131.94.250]:3270 at 12:40:52.841591:</div><div>REGISTER <a href="sip:voip.xxx.com">sip:voip.xxx.com</a> SIP/2.0</div><div>Via: SIP/2.0/UDP 192.168.1.100:3270;rport;branch=z9hG4bKb8c37e33defde51cf91e1e03e51657da</div><div>From: "cc" <<a href="sip:cc@voip.xxx.com:5090">sip:cc@voip.xxx.com:5090</a>>;tag=b8c37e33defde51cf91e1e03e51657da</div><div>To: "cc" <<a href="sip:cc@voip.xxx.com:5090">sip:cc@voip.xxx.com:5090</a>></div><div>Call-ID: <a href="mailto:b8c37e33defde51cf91e1e03e51657da@192.168.1.100">b8c37e33defde51cf91e1e03e51657da@192.168.1.100</a></div><div>CSeq: 1 REGISTER</div><div>Contact: <<a href="sip:cc@192.168.1.100:3270;rinstance=1242650454">sip:cc@192.168.1.100:3270;rinstance=1242650454</a>></div><div>max-forwards: 70</div><div>expires: 300</div><div><br></div><div><br></div></div><div><div>sent 439 bytes to udp/[69.131.94.250]:3270 at 12:40:52.841767:</div><div>SIP/2.0 482 Request merged</div><div>Via: SIP/2.0/UDP 192.168.1.100:3270;rport=3270;branch=z9hG4bKb8c37e33defde51cf91e1e03e51657da;received=69.131.94.250</div><div>From: "cc" <<a href="sip:cc@voip.xxx.com:5090">sip:cc@voip.xxx.com:5090</a>>;tag=b8c37e33defde51cf91e1e03e51657da</div><div>To: "cc" <<a href="sip:cc@voip.xxx.com:5090">sip:cc@voip.xxx.com:5090</a>>;tag=tjgccmtraDHFc</div><div>Call-ID: <a href="mailto:b8c37e33defde51cf91e1e03e51657da@192.168.1.100">b8c37e33defde51cf91e1e03e51657da@192.168.1.100</a></div><div>CSeq: 1 REGISTER</div><div>Content-Length: 0</div><div><br></div><div><br></div><div>And I also noticed the CSeq if not continues, seems it lost some. but why the CSeq so big while the client directly logins to FS without any proxy and I don't think there is a loop?</div><div><br></div><div>Anyway, don't know why FS does not respond to REGISTER sometimes. I updated FS to 13374, will see if it happen again.</div><div><br></div><div><html>On May 19, 2009, at 3:37 AM, Brian West wrote:</html><br class="Apple-interchange-newline"><blockquote type="cite"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Is this in regards to FreeSWITCH or something else you're writing?<div><br></div><div>/b</div><div><br><div><div>On May 18, 2009, at 2:34 PM, dujinfang wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; ">On register, sometimes my voip client got SIP/2.0 482 Request merged <br>sometimes got 200 ok.<br><br>482 also means loop detected. my client only has one account logged in <br>only one place, and no proxy, can I take 482 as 200 OK?<br><br>Thanks.<br><br>from RFC 3261:<br><br>"8.2.2.2 Merged Requests<br><br> If the request has no tag in the To header field, the UAS core MUST<br> check the request against ongoing transactions. If the From tag,<br> Call-ID, and CSeq exactly match those associated with an ongoing<br> transaction, but the request does not match that transaction (based<br> on the matching rules in Section 17.2.3), the UAS core SHOULD<br> generate a 482 (Loop Detected) response and pass it to the server<br> transaction."<br><br></span></blockquote></div><br><div apple-content-edited="true"> <span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div>Brian West</div><div><a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a></div><div><br></div></div></span>-- Meet us at ClueCon! <a href="http://www.cluecon.com/">http://www.cluecon.com</a><br><div><br></div></div></span><br class="Apple-interchange-newline"></div></span><br class="Apple-interchange-newline"> </div><br></div></div>_______________________________________________<br>Freeswitch-users mailing list<br><a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></blockquote></div><br></div></body></html>