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">http://bugs.digium.com/view.php?id=4687</a><br><a href="http://bugs.digium.com/view.php?id=4832">
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><br>-lars<br><br>