Hello List,<br><br>I am stuck with this problem for several days... anybody can give me a hint on this??<br><br>I know many of you dealt with problems similar to this, how did you address this??<br><br>Thanks in advance!!!
<br><br>-lars<br><br>---------- Forwarded message ----------<br><span class="gmail_quote">From: <b class="gmail_sendername">Lars Knopf</b> <<a href="mailto:lars.knopf@gmail.com">lars.knopf@gmail.com</a>><br>Date: Jan 11, 2007 1:12 PM
<br>Subject: realtime sipusers and rtcachefriends... big headache!!<br>To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br><br></span>hi folks,<br><br>I am using asterisk 1.2.13 (debian etch).
<br><br>My customer's sip accounts are stored in realtime sipusers.<br><br>I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes<br><br>Each account has nat=yes
<br><br>Now, I have lot of problems.<br><br>for example, when I change the 'secret' field of a user in the database, it doesn't<br>get reflected in Asterisk, who is still expecting the old password.<br><br>Randomly, when trying to dial SIP/xxxxx (a customer's account), especially those behind NAT,
<br>I get in the console the error "no route to...".<br><br>Sometimes, too, they can't even register with asterisk.<br><br>It seems to happen mostly when using realtime.<br><br>I was digging into the bug tracking system, and I see two bugs that seems to be related,
<br>but I can't figure how can I fix it or what step I am supposed to do. The bugs are:<br><br><a href="http://bugs.digium.com/view.php?id=4687" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
http://bugs.digium.com/view.php?id=4687</a><br><a href="http://bugs.digium.com/view.php?id=4832" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
http://bugs.digium.com/view.php?id=4832</a><br><br>So please, anything than can bring me some light on this... is very appreciated.<br><span class="sg"><br>-lars<br><br>
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