<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">No, that is a bug in the pbx session timer negotiation. &nbsp;We don’t see that often. &nbsp;I’d suggest reporting the bug to them, the only solution on our side would be to disable.<div class=""><br class=""></div><div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On Feb 8, 2017, at 12:56 PM, Jan Furi &lt;<a href="mailto:furi@vmtele.com" class="">furi@vmtele.com</a>&gt; wrote:</div><br class="Apple-interchange-newline"><div class="">
  
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  <div bgcolor="#FFFFFF" text="#000000" class=""><p class="">Thank you Michael for your advice. <br class="">
    </p><p class="">I am not&nbsp; an expert in session timers therefore I would be happy
      if anybody could explain whether current behavior is correct or
      not.</p><p class="">Customer makes a call : <br class="">
    </p><p class="">Customer PBX -&gt; INVITE requires Session-Expires:
      90;refresher=uac <br class="">
    </p><p class="">our FS1.6 returns in 200 OK : Session-Expires: 120;refresher=uas<br class="">
    </p><p class="">After 90 seconds Customer's PBX sends BYE - I think because of
      Session-Expires: 90</p><p class="">For testing, I changes my sip profile to &lt;param
      name="enable-timer" value="false"/&gt;</p><p class="">This time, call was not ended by customer's PBX.&nbsp; <br class="">
    </p><p class="">Do you think it can be NAT problem ? Honestly I am confused a bit
      with this issue. <br class="">
    </p><p class="">BR,</p><p class="">Jan<br class="">
    </p>
    <div class="moz-cite-prefix">On 08/02/2017 16:18, Michael Jerris
      wrote:<br class="">
    </div>
    <blockquote cite="mid:CAEwTsAj=mj2oKV3wuHxZXU1ODqYa2BHHur7qPiaZ5256f-xL5Q@mail.gmail.com" type="cite" class="">
      <div class="">this typically happens in misconfiguted nat scrnarios, which
        would otherwise break habgup for the same call.&nbsp; Troubleshoot
        why the call is breaking and fix the issue is the best approach.</div>
      <div class=""><br class="">
        <div class="gmail_quote">
          <div class="">On Wed, Feb 8, 2017 at 8:48 AM Ján Füri &lt;<a moz-do-not-send="true" href="mailto:furi@vmtele.com" class="">furi@vmtele.com</a>&gt;
            wrote:<br class="">
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            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <div bgcolor="#FFFFFF" text="#000000" class="gmail_msg"><p class="gmail_msg">Hi all,</p><p class="gmail_msg">I have a question about disabling
                enable-timer.</p><p class="gmail_msg">In my sip_profile i have &lt;param
                name="enable-timer" value="true"/&gt;. But one of our
                customer has a problem with it and his calls are ended
                after 90 sec. <br class="gmail_msg">
              </p><p class="gmail_msg">I think 90 sec. is time that
                customer's pbx is set for rtp timer. <br class="gmail_msg">
              </p><p class="gmail_msg">I don't want to change my sip profile
                enable-timer to false because it would influence all our
                customers. Is that possible to disable rtp timer in a
                dialplan ? <br class="gmail_msg">
              </p><p class="gmail_msg">BR,<br class="gmail_msg">
              </p>
              <div class="gmail_msg m_-3172178460692726109moz-signature">J<font style="font-family:'Tahoma',sans-serif;font-size:9pt;color:#333" class="gmail_msg"><font style="font-weight:bolder;color:#333333;font-size:11pt" class="gmail_msg">an Furi</font><br class="gmail_msg">
                  &nbsp; <br class="gmail_msg">
                </font></div>
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