<html><head><meta http-equiv="Content-Type" content="text/html; charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">I think we already hold off update until ack<br class=""><div><br class=""><blockquote type="cite" class=""><div class="">On Jun 28, 2018, at 3:37 AM, Melek Oktay <<a href="mailto:melekoktay@gmail.com" class="">melekoktay@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">Hi,<div class=""><br class=""></div><div class="">I realize very interesting problem of my software stack (FreeSwitch + Kamailio) <span style="background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline" class="">with Yealink 41P Voip phone.. </span></div><div class=""><span style="background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline" class=""><br class=""></span></div><div class=""><span style="background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline" class="">After 3-4 days research, I realize Yealink phone does not try to start RTP session if it gets UPDATE packet before ACK packet !! </span></div><div class=""><br class=""></div><div class="">In Freeswitch side, do we have a option that send UPDATE packet after RTP session is established? (we have a option in FreeSwitch side: do not send UPDATE packet however this is not good case for me, I update CallerID with this UPDATE packet)</div></div></div></blockquote></div><br class=""></body></html>