<span style="font-family: arial,helvetica,sans-serif;">Hi -</span><br style="font-family: arial,helvetica,sans-serif;"><br style="font-family: arial,helvetica,sans-serif;"><span style="font-family: arial,helvetica,sans-serif;">
I&#39;m looking into realtime and I&#39;m having a bit of a problem with the SIP part.&nbsp; 
</span><br style="font-family: arial,helvetica,sans-serif;"><br style="font-family: arial,helvetica,sans-serif;"><span style="font-family: arial,helvetica,sans-serif;">My review of the posts seems to indicate that I should use realtime static for the [general] part of my 
sip.conf including the registration commands:</span><br style="font-family: arial,helvetica,sans-serif;"><br style="font-family: arial,helvetica,sans-serif;"><div style="margin-left: 40px; font-family: arial,helvetica,sans-serif;">
register=&gt;&lt;did&gt;:&lt;secret&gt;@&lt;domain&gt;/&lt;did context&gt;
<br></div><br style="font-family: arial,helvetica,sans-serif;"><span style="font-family: arial,helvetica,sans-serif;">and use realtime realtime (funny name!) for peers and friends:</span><br style="font-family: arial,helvetica,sans-serif;">
<br style="font-family: arial,helvetica,sans-serif;"><div style="margin-left: 40px; font-family: arial,helvetica,sans-serif;">[myprovider]
<br>type=peer
<br>auth=md5
<br>username=...<br>fromuser=...<br>fromdomain=...
<br>secret=...<br>host=...
<br>port=5060
<br>nat=yes
<br>canreinvite=yes
<br>qualify=no
<br>disallow=all
<br>allow=ulaw<br>dtmfmode=rfc2833
<br>insecure=port,invite<br>context=incoming-sip<br></div><br style="font-family: arial,helvetica,sans-serif;"><span style="font-family: arial,helvetica,sans-serif;">Is this correct?&nbsp; What&#39;s throwing me off is this statment found 
</span><a style="font-family: arial,helvetica,sans-serif;" href="http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static" target="_blank">here:</a><span style="font-family: arial,helvetica,sans-serif;">
</span><br style="font-family: arial,helvetica,sans-serif;"><div style="margin-left: 40px; font-family: arial,helvetica,sans-serif;"><b><br>NOTE:</b> You can only store a static config OR a RealTime config.
You cannot, for example, store sip.conf and use sipfriends via
RealTime.<br><br></div><span style="font-family: arial,helvetica,sans-serif;">This would suggest that I&#39;ll have to do a reload when I add a DiD, but a reload won&#39;t be necessary if a new SIP client is added.&nbsp; Do I have it right?
</span><br style="font-family: arial,helvetica,sans-serif;"><br style="font-family: arial,helvetica,sans-serif;"><span style="font-family: arial,helvetica,sans-serif;">Also, what&#39;s the difference between a peer and a user?&nbsp; I used to think that a &quot;user&quot; 
was an agent&nbsp; authorized to call in to my * box, a &quot;peer&quot; was an agent I could reach and a &quot;freind&quot; was both.&nbsp; What&#39;s throwing me off now is the statement found </span><a style="font-family: arial,helvetica,sans-serif;" href="http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static">
here:</a><br style="font-family: arial,helvetica,sans-serif;"><br style="font-family: arial,helvetica,sans-serif;"><div style="margin-left: 40px; font-family: arial,helvetica,sans-serif;">With newer versions of Asterisk the concept of SIP &#39;users&#39; will be phased out.
<br><br></div><span style="font-family: arial,helvetica,sans-serif;">I can&#39;t understand this especially in the context of extconfig.conf that uses both a sipuser and sippeer entry.&nbsp; Could someone clarify for me?&nbsp; </span>
<br style="font-family: arial,helvetica,sans-serif;"><br style="font-family: arial,helvetica,sans-serif;"><span style="font-family: arial,helvetica,sans-serif;">Thanks,</span><br style="font-family: arial,helvetica,sans-serif;">
<span style="font-family: arial,helvetica,sans-serif;">H</span><br>