Hi List,

i’ve been receiving several sip registration probes in the last month,
and as this server is a testing site (no external lines, no nothing) i
have no fail2ban and still not planning to install. Whenever i have
nagios telling me that there is another ‘guest’, i go and edit iptables
manually and that’s it.

Recently i discovered that these attacks start with some kind of
dictionary, and try to guess valid peer names to use one by one.
Apparently after quarter million tries, they do find a legitim sip peer
name and from that point they stick to that peer name and the attack
continues to guess only passwords. Of course, they can not guess
passwords like p(F9j43/Qgrhjv*&^3 so i’m still not worried, but this
made me believe that asterisk responds differently when probing a valid
sip peer name.

So i was wondering through the sip.conf and found ‘alwaysauthreject’
which was set to default (commented out). I now set its value to yes
(which i thought was the default setting).

Does this setting makes the attacker believe that the first try of sip
peer name was valid, but only the password was incorrect? So in this
case should they stick to the first name tried whatever it was?

thanks
adam

I need in a strange applicatio a way to “detect” the tone (busy, ring
etc. etc.) of analog line (zap channel), while channel UP.

I found the application “NV” line detect, but is very old, and may be
not mantained.

I can patch asterisk to actually support this application but i think
someone other have something like this done.

Thnks.

hi guys,
i have a problem with 1.8 branch no matter which release of 1.8 i’m
using. i can’t make any sip calls, this is the error message i get on
each call:

[Jan 18 19:02:15] ERROR[1698] rtp_engine.c: No RTP engine was found.
Do you have one loaded?
[Jan 18 19:02:15] ERROR[1698] chan_sip.c: Got SDP but have no RTP
session allocated.

i’m sure that the rtp engine is loaded this is the messages i get when
loading rtp engine:

> module load res_rtp_asterisk.so
Loaded res_rtp_asterisk.so
== Registered RTP engine ‘asterisk’
== Parsing ‘/etc/asterisk/rtp.conf’: == Found
== RTP Allocating from port range 1650 -> 4650
Loaded res_rtp_asterisk.so => (Asterisk RTP Stack)

any advice to get rid of this problem?
thanks all
paradise