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: asterisk-users@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
_______________________________________________
--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
