*CLI> sip show settings


Global Settings:
----------------
  SIP Port:               5060
  Bindaddress:            0.0.0.0
  Videosupport:           No
  AutoCreatePeer:         No
  Allow unknown access:   Yes
  Promsic. redir:         No
  SIP domain support:     No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Our auth realm          asterisk
  Realm. auth:            No
  User Agent:             Asterisk PBX
  MWI checking interval:  10 secs
  Reg. context:           (not set)
  Caller ID:              asterisk
  From: Domain:           
  Record SIP history:     Off
  Call Events:            Off
  IP ToS:                 0x0
  OSP Support:            No
  SIP realtime:           Disabled

Global Signalling Settings:
---------------------------
  Codecs:                 ulaw,alaw,gsm,g723,g729
  Relax DTMF:             No
  Compact SIP headers:    No
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         No
  Pedantic SIP support:   No
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes

Default Settings:
-----------------
  Context:                home
  Nat:                    RFC3581
  DTMF:                   inband
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               (Defaults to English)
  Musicclass:             default
  Voice Mail Extension:   asterisk