<div>Olle,</div>
<div> </div>
<div>The rr param problem seems to have been fixed in 1.2 HEAD. Great job. Now I've discovered a new problem with Contact header URI parameters in 1.2 cvs HEAD.</div>
<div> </div>
<div>When Asterisk sends a BYE, the RURI does not have the paramaters from the Contact header of the original INVITE. Typicaly this shows up when the original INVITE has a "transport=tcp" in the Contact header.
</div>
<div> </div>
<div>Now I do realize that Asterisk does not support TCP, but this is not the issue. From what I've been told by the SER folks, any parameters in the INVITE Contact header should be in the R-URI of subsequent loose routes requests - like BYE.
</div>
<div> </div>
<div>Mark</div>
<div><br><br> </div>
<div><span class="gmail_quote">On 9/18/05, <b class="gmail_sendername">Mark Aiken</b> <<a href="mailto:aiken.mark@gmail.com">aiken.mark@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Olle,<br><br>Did the issue of rr parms come up? Asterisk doesnt return Record-Route parameters in its Route headers for in-Dialog messages like BYE and reINVITE.
<br><br>I dont know if this is a violation of the SIP RFCs or just a missing feature, but it helps when working with transaction stateful proxies like Ser and OpenSer. Since these proxies are not dialog stateful, RR parms are very useful for saving in dialog state information.
<br><span class="sg"><br>Mark</span>
<div><span class="e" id="q_1066abc6dd85e7cf_2"><br><br>
<div><span class="gmail_quote">On 9/18/05, <b class="gmail_sendername">Olle E. Johansson</b> <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:oej@edvina.net" target="_blank">oej@edvina.net</a>> wrote:
</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid"><a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:steve@daviesfam.org" target="_blank">
steve@daviesfam.org</a> wrote:<br>><br>> On Fri, 16 Sep 2005, Olle E. Johansson wrote:<br>><br>><br>>>Friends,<br>>>I've spent one week turning chan_sip inside out, testing with a lot of <br>>>different test tools, testing with different phones, proxies and other
<br>>>SIP devices. It's been wonderful. Fun, educational, stressful and SIP-crazy.<br><br>> You took Asterisk's SIP to SIPit? You're a brave man. Well done! <br>><br>Thanks. Yes, I hesitated quite a lot, but Rohan Mahoney convinced me...
<br><br>Also note that I ran all tests with pedantic=yes :-)<br><br>/O<br>_______________________________________________<br>Asterisk-Dev mailing list <br><a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:Asterisk-Dev@lists.digium.com" target="_blank">
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