<div dir="auto">Hi Antony, <div dir="auto"><br></div><div dir="auto">Sory but I don't understand why your Asterisk accept anon calls with the conf you provide us. </div><div dir="auto"><br></div><div dir="auto">Maybe a full excerpt of an incoming call will help. </div><div dir="auto"><br></div><div dir="auto">Last, there exist dialplan like GROUP and GROUP_COUNT that permits you count the number of calls in a custom group fashion. </div></div><div class="gmail_extra"><br><div class="gmail_quote">El 10/2/2017 11:51, "Антон Сацкий" <<a href="mailto:satskiy.a@gmail.com">satskiy.a@gmail.com</a>> escribió:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Thanks Frank -- but this not a solution </div><div>below my current config</div><div><br></div><div>[general]</div><div> </div><div>;sms</div><div> accept_outofcall_message = yes</div><div> outofcall_message_context = messages</div><div> auth_message_requests = no</div><div> </div><div>;general</div><div> allowguest = no</div><div> jbenable = no</div><div> jbimpl = adaptive</div><div> allow = !all,g722,ulaw,gsm</div><div> udpbindaddr = 0.0.0.0</div><div> transport = udp</div><div><br></div><div> language = ru</div><div> context = public</div><div> alwaysauthreject = yes</div><div> nat = force_rport,comedia</div><div> directmedia = no</div><div> allowoverlap = no</div><div> match_auth_username = yes</div><div><br></div><div> progressinband = yes</div><div> textsupport = yes</div><div> videosupport = yes</div><div> maxcallbitrate = 1384</div><div> ;</div><div> sendrpid = pai</div><div> rpid_update = yes</div><div> pedantic=no</div><div> ;tos</div><div> tos_sip=cs3</div><div> tos_audio=ef</div><div> tos_video=cs4</div></div><div class="gmail_extra"><br><div class="gmail_quote">2017-02-10 16:40 GMT+02:00 Frank Vanoni <span dir="ltr"><<a href="mailto:mailinglist@linuxista.com" target="_blank">mailinglist@linuxista.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span>On Thu, 2017-02-09 at 14:58 +0200, Антон Сацкий wrote:<br>
<br>
<br>
> so the main question is -- how to Disallow CALLS without registering<br>
> on PBX<br>
<br>
</span>sip.conf configuration<br>
In the [general] section, define:<br>
<br>
<br>
[general]<br>
...<br>
allowguest=no<br>
alwaysauthreject=yes<br>
...<br>
<br>
<br>
The "allowguest" line disables anonymous SIP calls to your PBX. Some SIP<br>
providers connect as a guest user, however, so this may be inappropriate<br>
for your situation. Also, if you want to accept anonymous SIP calls,<br>
this line would block them, so you wouldn't want that. But it is listed<br>
here because it is the safest configuration.<br>
<br>
The "alwaysauthreject" line is important. This causes a hacker to get<br>
the same response from your PBX when they try to guess passwords whether<br>
or not they guessed a valid username. This also has the side-effect of<br>
making poorly written scanning scripts (the vast majority of hacker<br>
scripts seem to be poorly written) take less resources on your Asterisk<br>
box, as even if they scan a valid username, they'll think it doesn't<br>
exist.<br>
<br>
(Source: <a href="https://www.voip-info.org/wiki/view/Asterisk+security" rel="noreferrer" target="_blank">https://www.voip-info.org/wiki<wbr>/view/Asterisk+security</a> )<br>
<span class="m_-8076028479491688638HOEnZb"><font color="#888888"><br>
<br>
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