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    <p>Thank you for your help</p>
    <p>Jan<br>
    </p>
    <br>
    <div class="moz-cite-prefix">On 08/02/2017 19:15, Michael Jerris
      wrote:<br>
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    <blockquote
      cite="mid:7584EABD-BA57-4CE7-BA0E-5C30996919D9@jerris.com"
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      No, that is a bug in the pbx session timer negotiation.  We don’t
      see that often.  I’d suggest reporting the bug to them, the only
      solution on our side would be to disable.
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          <blockquote type="cite" class="">
            <div class="">On Feb 8, 2017, at 12:56 PM, Jan Furi &lt;<a
                moz-do-not-send="true" href="mailto:furi@vmtele.com"
                class="">furi@vmtele.com</a>&gt; wrote:</div>
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                <p class="">Thank you Michael for your advice. <br
                    class="">
                </p>
                <p class="">I am not  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.  <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.  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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                        0 .8ex;border-left:1px #ccc
                        solid;padding-left:1ex">
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                          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">
                                <br class="gmail_msg">
                            </font></div>
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