dev2*CLI> sip show settings
dev2*CLI> 

Global Settings:
----------------
  UDP SIP Port:           5060
  UDP Bindaddress:        78.105.1.129
  TCP SIP Port:           Disabled
  TLS SIP Port:           Disabled
  Videosupport:           No
  Textsupport:            No
  AutoCreate Peer:        No
  Match Auth Username:    No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Allow promsic. redir:   No
  Enable call counters:   No
  SIP domain support:     No
  Realm. auth:            No
  Our auth realm          asterisk
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    No
  Direct RTP setup:       No
  User Agent:             Asterisk PBX SVN-branch-1.6.1-r162896
  SDP Session Name:       Asterisk PBX SVN-branch-1.6.1-r162896
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Caller ID:              asterisk
  From: Domain:           
  Record SIP history:     Off
  Call Events:            Off
  Auth. Failure Events:   Off
  T38 fax pt UDPTL:       No
  SIP realtime:           Disabled
  Qualify Freq :          60000 ms

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS0
  IP ToS RTP audio:       CS0
  IP ToS RTP video:       CS0
  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:   Yes
  Jitterbuffer forced:    No
  Jitterbuffer max size:  -1
  Jitterbuffer resync:    -1
  Jitterbuffer impl:      
  Jitterbuffer log:       No

Network Settings:
---------------------------
  SIP address remapping:  Disabled, no localnet list
  Externhost:             <none>
  Externip:               0.0.0.0:0
  Externrefresh:          10
  Internal IP:            78.105.1.129:5060
  STUN server:            0.0.0.0:0

Global Signalling Settings:
---------------------------
  Codecs:                 0x170f (g723|gsm|ulaw|alaw|g729|speex|ilbc|g722)
  Codec Order:            alaw:20,ulaw:20,gsm:20,g723:30,speex:20,ilbc:30,g729:20,g722:20
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Compact SIP headers:    No
  RTP Keepalive:          30 
  RTP Timeout:            600 
  RTP Hold Timeout:       3600 
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         Yes
  Pedantic SIP support:   No
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
  Notify hold state:      No
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No
  Outb. proxy:            78.105.1.128 
  Session Timers:         Originate
  Session Refresher:      uas
  Session Expires:        1800 secs
  Session Min-SE:         90 secs
  Timer T1:               500
  Timer T1 minimum:       100
  Timer B:                32000

Default Settings:
-----------------
  Context:                common
  Nat:                    RFC3581
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               
  MOH Interpret:          default
  MOH Suggest:            
  Voice Mail Extension:   asterisk

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