hi folks, I am using asterisk 1.2.13 (debian etch).
My customer's sip accounts are stored in realtime sipusers. I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes Each account has nat=yes Now, I have lot of problems. for example, when I change the 'secret' field of a user in the database, it doesn't get reflected in Asterisk, who is still expecting the old password. Randomly, when trying to dial SIP/xxxxx (a customer's account), especially those behind NAT, I get in the console the error "no route to...". Sometimes, too, they can't even register with asterisk. It seems to happen mostly when using realtime. I was digging into the bug tracking system, and I see two bugs that seems to be related, but I can't figure how can I fix it or what step I am supposed to do. The bugs are: http://bugs.digium.com/view.php?id=4687 http://bugs.digium.com/view.php?id=4832 So please, anything than can bring me some light on this... is very appreciated. -lars
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