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<div class="moz-cite-prefix">On 8/6/20 8:09 AM, Jerry Geis wrote:<br>
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<blockquote type="cite"
cite="mid:CABr8-B4VFbq5CU8Z-ZUhnoXhziQTP8f2AowjrHq1KLrZA8Q3rw@mail.gmail.com">
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<div dir="ltr">I am using asterisk 13.33.0 and POlycom phone with
the latest firmware.
<div><br>
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<div>The polycom phone is behind a firewall, the server is in
the cloud.</div>
<div>If the polycom has just booted - it receives a call, after
some time (couple minutes) it no longer receives a ring. I see
no errors in the CLI - looks just like the previous call as
far as I can tell.</div>
<div><br>
</div>
<div>Then reboot the phone and as soon as its ready call it and
it rings just liek before. then some time later no longer
rings.</div>
<div><br>
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</blockquote>
Sounds to me like you need to enable keep alives on the Polycom so
it keeps the NAT pinhole open in the outbound direction. It will
also help to enable the qualify setting on the PBX itself for the
extension so it keeps sending SIP messages to the phone ensuring
connectivity in the inbound direction.<br>
<br>
qualify=yes<br>
<br>
<blockquote type="cite"
cite="mid:CABr8-B4VFbq5CU8Z-ZUhnoXhziQTP8f2AowjrHq1KLrZA8Q3rw@mail.gmail.com">
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<div> -- Executing [something@smvoice-dialout:4]
Dial("SIP/1005-000000ab", "SIP/526,30000,tT") in new stack<br>
== Using SIP RTP CoS mark 5<br>
-- Called SIP/526<br>
-- SIP/526-000000ac is ringing<br>
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<div><br>
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<div>526 is the extension in question. (my definition follows):</div>
<div>[526]<br>
type=friend<br>
defaultname=526<br>
defaultuser=526<br>
secret=XXXXXXXXX<br>
dtmfmode=RFC2833<br>
host=dynamic<br>
description=Polycom
<br>
context=sip<br>
qualify=yes<br>
rtptimeout=60<br>
rtpholdtimeout=60<br>
rtpkeepalive=60<br>
callerid="Polycom "</div>
<div>qualify=no<br>
canreinvite=yes<br>
timezone=1<br>
nat=force_rport,comedia<br>
disallow=all<br>
allow=ulaw<br>
allow=alaw<br>
allow=gsm<br>
</div>
<div><br>
</div>
<div>Thoughts on what is happening here or what to try?</div>
<div><br>
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<div>Jerry </div>
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<pre class="moz-signature" cols="72">--
Andres</pre>
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