Global Settings:
----------------
  UDP Bindaddress:        0.0.0.0:5060
  TCP SIP Bindaddress:    0.0.0.0:5060
  TLS SIP Bindaddress:    213.133.102.85:5061
  Videosupport:           Yes
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        Off
  Match Auth Username:    Yes
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  No
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain support:     No
  Path support :          No
  Realm. auth:            No
  Our auth realm          srv01
  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:             Asterisk PBX 12.4.0
  SDP Session Name:       Asterisk PBX 12.4.0
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Trust RPID:             Yes
  Send RPID:              Yes
  Legacy userfield parse: No
  Send Diversion:         Yes
  Caller ID:              asterisk
  From: Domain:           
  Record SIP history:     Off
  Auth. Failure Events:   Off
  T.38 support:           Yes
  T.38 EC mode:           FEC
  T.38 MaxDtgrm:          4294967295
  SIP realtime:           Enabled
  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:        AF41
  802.1p CoS SIP:         3
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   4
  802.1p CoS RTP text:    3
  Jitterbuffer enabled:   No

Network Settings:
---------------------------
  SIP address remapping:  Disabled, no localnet list
  Externhost:             <none>
  Externaddr:             (null)
  Externrefresh:          10

Global Signalling Settings:
---------------------------
  Codecs:                 (gsm|ulaw|alaw|h263|testlaw)
  Codec Order:            none
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          No
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         Yes
  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 403:0
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      No
  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,TLS
  Outbound transport:	  UDP
  Context:                fromoutside
  Record on feature:      automon
  Record off feature:     automon
  Force rport:            Auto (No)
  DTMF:                   rfc2833
  Qualify:                0
  Keepalive:              0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               
  Tone zone:              <Not set>
  MOH Interpret:          default
  MOH Suggest:            
  Voice Mail Extension:   asterisk

Realtime SIP Settings:
----------------------
  Realtime Peers:         Yes
  Realtime Regs:          No
  Cache Friends:          Yes
  Update:                 Yes
  Ignore Reg. Expire:     No
  Save sys. name:         Yes
  Save path header:       No
  Auto Clear:             120 (Enabled)

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