Since you mentioned FreePBX, unfortunately, it's not only the GUI that drives the system and it can be that at some point someone planted the extension in one of your .conf or other file if they had access to SSH or some other way. <div>
<br></div><div>Going back to occurrence in sip.conf as mentioned, of course FreePBX regenerates sip.conf every time and you can't tamper with it but sip_custom.conf or any other file can be called to just create an extension in the non-GUI section and it will still work and not show up in FreePBX GUI.<div>
<br></div><div>Recreating the extension probably over-wrote that or maybe supersedes that and hence the failed authentication attempts.</div><div><br></div><div>If you can live with no SIP from outside, temporarily block any incoming on 5060 and 10000-20000.</div>
<div><br></div><div>To find the extension occurances in the .conf files, try this:</div><div><br></div><div>=> Delete the extension from FreePBX first and then:</div><div>cd /etc/asterisk</div><div>grep -o "3799" *.*</div>
<div><br></div><div>However, I think it's also possible to have the 3799 created in a totally different directory in your server as long as it has the right asterisk.asterisk permissions and it can be called by an #include from sip_custom.conf. So, check that file out extensively.</div>
<div><br></div><div>-Bruce</div><div><br><div><br></div><div><br><div class="gmail_quote">On Tue, Jun 8, 2010 at 10:56 AM, Steve Murphy <span dir="ltr"><<a href="mailto:murf@parsetree.com">murf@parsetree.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">I hope I'm correct, I don't have time to verify every bit of this,<br>but....<br><br>The message <br><div class="im">
<br>[Jun 7 17:04:16] NOTICE[7422] chan_sip.c: Failed to authenticate user<br>
"asterisk" <<a href="mailto:sip%3A3799@206.205.124.247" target="_blank">sip:3799@206.205.124.247</a>>;tag=as23bacb61<br>
<br></div>indicates the user "asterisk". Do you have a sip account for "asterisk"?<br><br>Why it would take 14 seconds and an ANSWERED dial for an unathenticated<br>use is something to investigate!<br>
<br>As to the more general question of how 3799 could be unexpectedly matched<br>
in the dialplan, I would respond that there are several possibilities...<br><br>One is, Is the account with the weak<br>pword removed from sip.conf? The 3799 account? Because, it looks like <br>SIP/206.20... (you abbreviated here in the CDR you listed) is where<br>
the call is originating.<br><br>b. Did you *really* get rid of all 3799 occurrences in the dialplan? What patterns<br>do you have in the dialplan that might match 3799, after the explicit 3799 is removed?<br>Any _XXXX type patterns included or in the context in question?<br>
<br>c. I uncovered a pattern matching bug, and reported it in bug <a href="https://issues.asterisk.org/view.php?id=17366" target="_blank">https://issues.asterisk.org/view.php?id=17366</a><br>where unexpected patterns are matched. Sorry, I haven't had time to correct it myself, it's probably<br>
a simple 1-line fix, but oh, what it might take to figure out what the line should say, and where it is!<br><br>d. "s" is the "start" extension, and an incoming call will tend to get routed into an "s" extension.<br>
<br>You can quickly determine (b) or (c), by going to the CLI, and saying <br>"dialplan show 3799@whatever-context and see what turns up.<br><br>murf<br><br><br><br><br><br><div class="gmail_quote"><div><div></div><div class="h5">
On Tue, Jun 8, 2010 at 7:50 AM, J <span dir="ltr"><<a href="mailto:jmaurer@2ergo.com" target="_blank">jmaurer@2ergo.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex"><div><div></div><div class="h5">I'm fairly new to FreePBX/Asterisk/Trixbox, but have Googled myself<br>
into submission here, so any assistance is appreciated.<br>
<br>
We had a user with a weak SIP secret recently that allowed it to be<br>
used by an outside user. The extension was 3799. I could see the<br>
intruder's calls (including the destination phone numbers) in the<br>
trixbox call report log. Because the extension was no longer used, I<br>
went ahead and deleted it, thinking that would solve the problem. I<br>
also discovered approximately the same time that the Asterisk Call<br>
Manager port was open to the outside world, which has since been<br>
closed. The web interface, ssh, etc. have never been exposed to the<br>
outside world. Since taking these actions, I restarted the asterisk<br>
server.<br>
<br>
Now, here's the issue. I don't think deleting the extension helped.<br>
Now I see entries like this in the reports log:<br>
<br>
Calldate Channel Source Clid Dst Disposition Duration<br>
1. 2010-06-07 16:47:38 SIP/206.20... 3799 "asterisk"<br>
<3799> s ANSWERED 00:14<br>
<br>
The "Dst" field being "s", where it used to be the phone number being<br>
dialed. How is this extension able to be used even after it has been<br>
deleted?<br>
<br>
Strangely, what I've done to keep the user out in the mean time is<br>
re-created the 3799 extension with a better secret. This results in<br>
log entries like the following:<br>
<br>
[Jun 7 17:04:16] NOTICE[7422] chan_sip.c: Failed to authenticate user<br>
"asterisk" <<a href="mailto:sip%3A3799@206.205.124.247" target="_blank">sip:3799@206.205.124.247</a>>;tag=as23bacb61<br>
<br>
Why can sip:3799 connect and make calls when the extension doesn't<br>
exist? Is this person somehow using a "user" account? I've checked<br>
both /etc/asterisk and the MySQL tables and am not coming up with<br>
much. What does it mean that their destination is "s", not a phone<br>
number?<br>
<br>
Thanks for any assistance!<br>
J<br>
<br></div></div>
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