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Messages - willow3

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16
Users / Re: Asterisk hacked
« on: December 28, 2011, 02:14:03 pm »
Sorry to hear this happened to you, I feel your pain http://forum.linuxmce.org/index.php/topic,12011.0.html

Cheers,
Matt.

I am sorry you lost money too, man. Thanks for the link though. It contains good advice. From the information that you guys have provided, I think the following measures are appropriate:

- Configure fail2ban to stop brute force against SIP extensions. (According to wiki)
- Employ a restrictive set of dial patterns for your outgoing route
- Subscribe to a dial plan with a limited number of monthly minutes or credits or whatever

I did none of these, which gave me a good security lesson worth $300. Btw, the only reason it did not cost me $30000 or $300000 is that my VOIP provider automatically detected the calls as an "unusual usage" and blocked all outgoing international calls for my phone number. I did not know they had this functionality, they do not advertise it. But I am happy they did that.

I am still confused that gufw says the firewall is disabled. And that external calls work even if I removed the rule for port 5060 in the web admin. Any ideas on that?

all the best

17
Users / Re: Asterisk hacked
« on: December 27, 2011, 06:26:42 pm »
Nope, but thanks for the tip. Would that tool be able to stop unauthorized attempts to register SIP extensions to asterisk?

18
Users / Asterisk hacked
« on: December 26, 2011, 03:04:33 pm »
Hi all!

Hope y'all had a very merry Christmas!

Like the subject indicates I got my asterisk hacked the other day resulting in a huge phone bill. I had a look at the phones in the system and the auto generated SIP passwords seem very difficult to guess. Yet someone succeeded. In the call log I could see that there were calls placed from all extensions in my system. Apparently someone took the pain to crack all of my passwords, which surprised me a bit.

I had a peek at the firewall, and maybe some of you guys can shed some light on how this works. There is a rule to open up port 5060 for udp. This seems inevitable if you want to place and receive external calls. But I guess this also opens up a security risk?

Just as a test I removed this rule to see what would happen. To my surprise, external calls still work. Is this normal? I did a quick reload router. Maybe I need to restart the core? Besides, the firewall used, is it a separate LMCE firewall or is it the kernel built-in one? I launched gufw and it indicated that the kernel firewall was turned off.

Is the general recommendation to have a strict dial plan to avoid having hackers placing calls to expensive phone numbers? Or do I have some security problem with my system that I am not aware of?

Any suggestions or information is welcome!

Happy new year everyone!

Regards

19
Installation issues / Re: Sound and video problems with 810 rc1
« on: September 17, 2011, 05:18:33 pm »
*hmm* a couple of restarts of the MD and the sound and video was back. I thought that kind of solution was reserved to the windows community ;-)

20
Installation issues / Re: Sound and video problems with 810 rc1
« on: September 17, 2011, 04:54:15 pm »
No, I haven't been able to resolve this one. Just for clarification: The UI2 works, but I can not play any video media. Neither does the sarah video stream show up during setup.

I installed the  LinuxMCE-8.10-final.iso  from the release directory.

regards

21
Installation issues / [SOLVED] Sound and video problems with 810 rc1
« on: September 11, 2011, 09:29:28 pm »
Hi,

Experiencing several problems with a snapshot from October 2010 I installed rc1. I have an MD with a geforce 9300 that worked out of the box earlier. Now, I did not get any sound nor video. Starting with the sound I got the following:

Code: [Select]
sambahelper@moon35:~$ aplay -l      
aplay: device_list:217: no soundcards found...

Code: [Select]
sambahelper@moon35:~$ lspci -v
00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1)
        Subsystem: nVidia Corporation MCP79 High Definition Audio
        Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 20
        Memory at fe020000 (32-bit, non-prefetchable) [size=16K]
        Capabilities: <access denied>
        Kernel driver in use: HDA Intel
        Kernel modules: snd-hda-intel

The only thing I have done is to upgrade alsa to 1.0.23 (which is what I was using before).

As for video:

Code: [Select]
sambahelper@moon35:~$ grep -i "x driver" /var/log/Xorg.0.log
(II) NVIDIA dlloader X Driver  195.36.15  Thu Mar 11 22:01:49 PST 2010

This is the version I used before.

Does anyone know what has happened since last year that could explain this behavior? Any logs I should examine?

regards



22
Users / flash player issue
« on: July 28, 2011, 09:33:54 pm »
I have a new MD on 8.10 and when I want to install flash player to watch youtube I can not do that since adobe only have for ubuntu versions 9.04+.

I tried

Code: [Select]
sudo apt-get install flashplugin-nonfree
but it said I already have the latest version.

Earlier when flash player was available for older versions on adobes site I know I got it to work. Someone who knows how I should get a working version?

regards

23
Users / Re: [Solved] Block telemarketing numbers
« on: February 06, 2011, 01:42:22 pm »
I added a short description here http://wiki.linuxmce.org/index.php/Telecom_Features#Black_list

regards

24
Users / Re: Problems with create_amp_*.pl
« on: February 04, 2011, 11:48:24 pm »
For those who have a linksys RTP300 collecting dust in their drawer, here is how to integrate it in LinuxMCE:

http://wiki.linuxmce.org/index.php/Linksys_rtp300

regards

25
Users / Re: Block telemarketing numbers
« on: February 03, 2011, 10:58:24 pm »
@pw44: That did the trick. Thanks a lot!

26
Users / Re: Block telemarketing numbers
« on: February 03, 2011, 09:37:32 pm »
@pw44: Interesting, I couldn't find it. When I go to the Module admin I have the list of modules in the attached jpg. I have FreePBX 2.2.0beta3. I guess that is the one shipped with LinuxMCE 8.10.

27
Users / Re: Block telemarketing numbers
« on: February 03, 2011, 10:33:55 am »
Ok. That approach is a little too strict for me. I would prefer a blacklist with numbers that are blocked (simply hang up). Other numbers are passed through to my extensions. At the moment, there is a limited amount of numbers that annoy me, so I can very well populate the blacklist manually.

I would like to know if there is a built in method in linux mce to do this or if I need to hack asterisk manually. I didn't find anything in mce documentation or in forum, so my guess is no. But I would like to be sure first.

regards


28
Users / [Solved] Block telemarketing numbers
« on: February 03, 2011, 12:23:35 am »
Is there a way to block such numbers, e.g via the web admin?

regards

29
Users / Re: Problems with create_amp_*.pl
« on: February 03, 2011, 12:20:11 am »
the syntax wasn't my problem. everything worked fine once i saw the "apply changes" button on top of the page  :)

I also managed to connect the linksys rtp300 to asterisk. When I get the time I will make a wiki page about it.

30
Users / Re: Problems with create_amp_*.pl
« on: January 27, 2011, 08:17:14 pm »
@pointman87: I didn't get your script for affinity to work. It didn't successfully register to the sip server. I don't have much time to troubleshoot for the moment so I tried the sipgate template instead. I got the same result as you. Incoming calls work but not outgoing. I tried experimenting with the outbound dial patterns in the Free PBX settings without success. I also noted there are two sets of dial patterns; one under Basic->Outbound routes-sipgate->Dial patterns and another under Basic->Trunk SIP/Sipgate->Dial rules. They seem unrelated and I don't know the difference. Which one did you change, and what rules did you use?

regards

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