Thank you, I wasn't aware of the prune command! -Lars
On 1/15/07, David Thomas <[EMAIL PROTECTED]> wrote:
On 1/15/07, Lars Knopf <[EMAIL PROTECTED]> wrote: > Hello List, > > I am stuck with this problem for several days... anybody can give me a hint > on this?? > > I know many of you dealt with problems similar to this, how did you address > this?? > > Thanks in advance!!! > > -lars > > ---------- Forwarded message ---------- > From: Lars Knopf <[EMAIL PROTECTED]> > Date: Jan 11, 2007 1:12 PM > Subject: realtime sipusers and rtcachefriends... big headache!! > To: [email protected] > > 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. I think you will need to prune the user/peer after changes. I believe the syntax is something like "sip prune realtime user_or_peer" where user_or_peer is the actual username. - David _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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