<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. We don’t see that often. 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 <<a href="mailto:furi@vmtele.com" class="">furi@vmtele.com</a>> 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 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 -> 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 <param
name="enable-timer" value="false"/></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 <<a moz-do-not-send="true" href="mailto:furi@vmtele.com" class="">furi@vmtele.com</a>>
wrote:<br class="">
</div>
<blockquote class="gmail_quote" style="margin:0 0 0
.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 <param
name="enable-timer" value="true"/>. 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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