<html><head></head><body>Indeed, Section 3.1 in the referenced RFC covers this pretty well. <br><br><div class="gmail_quote">On 4 February 2014 09:47:38 GMT-05:00, "Olle E. Johansson" <oej@edvina.net> wrote:<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<pre class="k9mail"><br />On 04 Feb 2014, at 15:43, Moritz Graf <moritz.graf@g-fit.de> wrote:<br /><br /><blockquote class="gmail_quote" style="margin: 0pt 0pt 1ex 0.8ex; border-left: 1px solid #729fcf; padding-left: 1ex;"> <br /> Shortly explained what RFC3578 is: In a open numbering plan you never<br /> know if the INVITE you received is already complete, or if there are<br /> more numbers coming in. One way of accomplishing this is to set up a<br /> timer. If the timer elapses you assume the number is complete. If not,<br /> you are receiving a new INVITE with one digit more. Now you have to<br /> close the old transaction with a "484 - Address Incomplete"-response and<br /> start the timer again. (Find the algorithm I want to implement attached)<br /></blockquote>You are not allowed to have two open INVITEs, the second one SHOULD get<br />a 491 response. The client should not send a new INVITE if the old INVITE<br />transaction is not complete.<br /><br />I don't think
overlap dialing in SIP was ever meant to be timer based. Consider<br />that the first INVITE can go to a different proxy than the second. There's no<br />dialog, no route set.<br /><br />/O<br /><hr /><br />SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br />sr-users@lists.sip-router.org<br /><a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br /></pre></blockquote></div><br>
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