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<p class=MsoNormal><span style='color:#1F497D'>> </span>I'm struggling to
get the MWI set up on a few Polycom phones.<span style='color:#1F497D'><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Sorted. From voip-info. <o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>http://www.voip-info.org/wiki/view/Asterisk+RealTime<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal>The <strong><span style='font-family:"Calibri","sans-serif"'>database
peers/users</span></strong> are not kept in memory. These are only loaded when
we have a call and then deleted, so <strong><span style='font-family:"Calibri","sans-serif"'>there's
no support for NAT keep-alives (qualify=) or voicemail indications</span></strong>
for these peers. <br>
<strong><span style='font-family:"Calibri","sans-serif"'>NOTE:</span></strong>
If you enable RealTime caching in your sip.conf, Voicemail MWI works and so
does 'sip show peers' - see rtcachefriends=yes. The downside to this is that if
you change anything in the database, you need to do a 'sip reload' (for major
changes) or 'sip prune realtime PEERNAME' (for single peer changes) before they
become active.<span style='color:#1F497D'><o:p></o:p></span></p>
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