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

[David Thomas]

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