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--></style></head><body lang="EN-ZA" link="blue" vlink="purple"><div class="WordSection1"><p class="MsoNormal"><span style="color:#1f497d">Also, I find quite interesting…</span></p><p class="MsoNormal"><span style="color:#1f497d"> </span></p>
<p class="MsoNormal"><span style="color:#1f497d">Asterisk when receiving a 420, ends the call without sending a bye, and logs a -- Got SIP response 420 "Option Disabled" back from x.x.x.x</span></p><p class="MsoNormal">
<span style="color:#1f497d"> </span></p><p class="MsoNormal"><span style="color:#1f497d">But SPCE still thinks the call is going. I have no RTP audio from one side now. My rtp_timeout = 60, and rtp_timeout_onhold = 3600. </span></p>
<p class="MsoNormal"><span style="color:#1f497d">The caller is still on the dead line now for 10 minutes. What registers that the call was on hold? Because I the call was on hold at a point, but is no longer. (Seems like it doesn’t unset this).</span></p>
<p class="MsoNormal"><span style="color:#1f497d"> </span></p><p class="MsoNormal"><span style="color:#1f497d">I’m busy testing if I never put the call on hold, if the rtp_timeout works@60sec. Perhaps I’m mistaken and the rtp_timeout requires both ends to send no rtp?</span></p>
<p class="MsoNormal"><span style="color:#1f497d"> </span></p><p class="MsoNormal"><span style="color:#1f497d">Regards</span></p><p class="MsoNormal"><span style="color:#1f497d">Matthew</span></p><p class="MsoNormal"><span style="color:#1f497d"> </span></p>
<p class="MsoNormal"><span style="color:#1f497d"> </span></p><div style="border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm 4.0pt"><div><div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0cm 0cm 0cm">
<p class="MsoNormal"><b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> Matthew Ogden [mailto:<a href="mailto:matthew@tenacit.net">matthew@tenacit.net</a>] <br>
<b>Sent:</b> 11 October 2013 02:36 PM<br><b>To:</b> <a href="mailto:spce-user@lists.sipwise.com">spce-user@lists.sipwise.com</a><br><b>Subject:</b> 2 asterisk subscribers - subscriber sst setting ignored?</span></p></div>
</div><p class="MsoNormal"> </p><p class="MsoNormal">Hi all,</p><p class="MsoNormal"> </p><p class="MsoNormal">I have two asterisk subscribers, and SPCE sitting in the middle. My domain is setup to ignore SST by default and rely on SEMS to dead call detection. The caller asterisk configuration itself has SST set to refuse. The called party doesn’t have this set to Accept.</p>
<p class="MsoNormal"> </p><p class="MsoNormal">Then the caller calls the callee, its passing the “supported: replaces, timer” in the INVITE to the callee. The callee asterisk box, picks this up, an in its 200 OK, says “require: timer”. All goes well for 15 minutes, and then the calls ends, because the callee, does a REINVITE, the REINVITE goes through to the caller, and the caller says “420 Option Disabled”. </p>
<p class="MsoNormal"> </p><p class="MsoListParagraph" style><span style>1)<span style="font:7.0pt "Times New Roman""> </span></span>If SST is disabled, why is it passing “supported: replaces, timer” to the callee?</p>
<p class="MsoListParagraph" style><span style>2)<span style="font:7.0pt "Times New Roman""> </span></span>Why is the callee reinvite being passed to the caller? (Surely the reinvite is only for its leg of the call?)</p>
<p class="MsoNormal"> </p><p class="MsoNormal">I can disable it on my callee, but I was wondering if 1 or 2 above is actually a bug in SPCE as well that should be looked at? </p><p class="MsoNormal"> </p><p class="MsoNormal">
If its SPCE or Homer teammembers I can give you the pcap, and any logs you need if you want.</p><p class="MsoNormal"> </p><p class="MsoNormal">Regards</p><p class="MsoNormal">matthew</p><p class="MsoNormal"> </p></div></div>
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