[asterisk-users] ANY ADVICE ON THIS????

[Lars Knopf]

Thank you, I wasn't aware of the prune command!

-Lars

On 1/15/07, David Thomas <punknow at gmail.com> wrote:
>
> On 1/15/07, Lars Knopf <lars.knopf at gmail.com> 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 <lars.knopf at gmail.com>
> > Date: Jan 11, 2007 1:12 PM
> > Subject: realtime sipusers and rtcachefriends... big headache!!
> > To: asterisk-users at lists.digium.com
> >
> > 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
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