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On 08/06/10 14:50, J wrote:
<blockquote
cite="mid:AANLkTikC4MUC1xiOM9yarbFcuSWxYMOFGT02L0STdTvL@mail.gmail.com"
type="cite">
<pre wrap="">I'm fairly new to FreePBX/Asterisk/Trixbox, but have Googled myself
into submission here, so any assistance is appreciated.
We had a user with a weak SIP secret recently that allowed it to be
used by an outside user. The extension was 3799. I could see the
intruder's calls (including the destination phone numbers) in the
trixbox call report log. Because the extension was no longer used, I
went ahead and deleted it, thinking that would solve the problem. I
also discovered approximately the same time that the Asterisk Call
Manager port was open to the outside world, which has since been
closed. The web interface, ssh, etc. have never been exposed to the
outside world. Since taking these actions, I restarted the asterisk
server.
Now, here's the issue. I don't think deleting the extension helped.
Now I see entries like this in the reports log:
Calldate Channel Source Clid Dst Disposition Duration
1. 2010-06-07 16:47:38 SIP/206.20... 3799 "asterisk"
<3799> s ANSWERED 00:14
The "Dst" field being "s", where it used to be the phone number being
dialed. How is this extension able to be used even after it has been
deleted?
Strangely, what I've done to keep the user out in the mean time is
re-created the 3799 extension with a better secret. This results in
log entries like the following:
[Jun 7 17:04:16] NOTICE[7422] chan_sip.c: Failed to authenticate user
"asterisk" <a class="moz-txt-link-rfc2396E" href="sip:3799@206.205.124.247"><sip:3799@206.205.124.247></a>;tag=as23bacb61
Why can <a class="moz-txt-link-freetext" href="sip:3799">sip:3799</a> connect and make calls when the extension doesn't
exist? Is this person somehow using a "user" account? I've checked
both /etc/asterisk and the MySQL tables and am not coming up with
much. What does it mean that their destination is "s", not a phone
number?
Thanks for any assistance!
J
</pre>
</blockquote>
<br>
Hi<br>
<br>
Were you using RealTime and/or allowing realtime caching? If so it is
possible that the user/peer is still in the realtime cache even though
the sip extension has been deleted from the DB<br>
<br>
Open the console and execute the following<br>
<br>
sip prune realtime 3799<br>
<br>
If you get a response of pruned then that was the problem, if you get a
response of not found then it's back to the drawing board<br>
<br>
Ish<br>
<div class="moz-signature">-- <br>
<title></title>
<span
style="font-family: Arial,Helvetica,sans-serif; color: rgb(0, 0, 0);">Ishfaq
Malik<br>
Software Developer<br>
PackNet Ltd<br>
<br>
Office: 0161 660 3062<br>
</span>
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