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
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.
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 rtp_engine.c: No RTP engine was found.
Do you have one loaded?
[Jan 18 19:02:15] ERROR 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
== Registered RTP engine 'asterisk'