I threw off sip show settings, how to secure my freePBX?

Question

Asterisk

As I understand FreePBX has completely different configuration files, the client asks for freepbx

localhost*CLI>         
Disconnected from Asterisk server  
Asterisk cleanly ending (0).       
Executing last minute cleanups     
[email protected]:/# ^C   
[email protected]:/# asterisk -rx 'sip show settings'         
           
           
Global Settings:       
----------------       
  UDP Bindaddress:        0.0.0.0:5160         
  TCP SIP Bindaddress:    Disabled 
  TLS SIP Bindaddress:    Disabled 
  Videosupport:           No       
  Textsupport:No       
  Ignore SDP sess. ver.:  No       
  AutoCreate Peer:        Off      
  Match Auth Username:    No       
  Allow unknown access:   Yes      
  Allow subscriptions:    Yes      
  Allow overlap dialing:  Yes      
  Allow promisc. redir:   No       
  Enable call counters:   No       
  SIP domain support:     No       
  Path support :          No       
  Realm. auth:No       
  Our auth realm          asterisk 
  Use domains as realms:  No       
  Call to non-local dom.: Yes      
  URI user is phone no:   No       
  Always auth rejects:    Yes      
  Direct RTP setup:       No       
  User Agent: FPBX-13.0.190.9(13.13.1)  
  SDP Session Name:       Asterisk PBX 13.13.1 
  SDP Owner Name:         root     
  Reg. context:           (not set)
  Regexten on Qualify:    No       
  Trust RPID: No       
  Send RPID:  No       
  Legacy userfield parse: No       
  Send Diversion:         Yes      
  Caller ID:  Unknown  
  From: Domain:        
  Record SIP history:     Off      
  Auth. Failure Events:   Off      
  T.38 support:           No       
  T.38 EC mode:           Unknown  
  T.38 MaxDtgrm:          4294967295           
  SIP realtime:           Disabled 
  Qualify Freq :          60000 ms 
  Q.850 Reason header:    No       
  Store SIP_CAUSE:        No       
           
Network QoS Settings:  
---------------------------        
  IP ToS SIP: CS3      
  IP ToS RTP audio:       EF       
  IP ToS RTP video:       AF41     
  IP ToS RTP text:        CS0      
  802.1p CoS SIP:         4        
  802.1p CoS RTP audio:   5        
  802.1p CoS RTP video:   6        
  802.1p CoS RTP text:    5        
  Jitterbuffer enabled:   No       
           
Network Settings:      
---------------------------        
  SIP address remapping:  Disabled, no localnet list       
  Externhost: <none>   
  Externaddr: (null)   
  Externrefresh:          10       
           
Global Signalling Settings:        
---------------------------        
  Codecs:     (ulaw|alaw|gsm|g726) 
  Relax DTMF: No       
  RFC2833 Compensation:   No       
  Symmetric RTP:          Yes      
  Compact SIP headers:    No       
  RTP Keepalive:          0 (Disabled)         
  RTP Timeout:30       
  RTP Hold Timeout:       300      
  MWI NOTIFY mime type:   application/simple-message-summary           
  DNS SRV lookup:         No    
  Pedantic SIP support:   Yes      
  Reg. min duration       60 secs  
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs 
  Sub. min duration       60 secs  
  Sub. max duration:      3600 secs
  Outbound reg. timeout:  20 secs  
  Outbound reg. attempts: 0        
  Outbound reg. retry 403:No       
  Notify ringing state:   Yes      
    Include CID:          No       
  Notify hold state:      Yes      
  SIP Transfer mode:      open     
  Max Call Bitrate:       384 kbps 
  Auto-Framing:           No       
  Outb. proxy:<not set>
  Session Timers:         Accept   
  Session Refresher:      uas      
  Session Expires:        1800 secs
  Session Min-SE:         90 secs  
  Timer T1:   500      
  Timer T1 minimum:       100      
  Timer B:    32000    
  No premature media:     Yes      
  Max forwards:           70       
           
Default Settings:      
-----------------      
  Allowed transports:     UDP      
  Outbound transport:     UDP      
  Context:    from-sip-external    
  Record on feature:      automon  
  Record off feature:     automon  
  Force rport:Yes       
   DTMF:       rfc2833  
  Qualify:    0        
  Keepalive:  0        
  Use ClientCode:         No       
  Progress inband:        No       
  Language:   en       
  Tone zone:  <Not set>
  MOH Interpret:          default  
  MOH Suggest:         
  Voice Mail Extension:   *97      
           
----       
[email protected]:/#

Answer 1

Turn it off and never turn it back on.
Seriously though, have you even googled? Iptables, SElinux, fail2ban

Answer 2

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