Global Settings:
----------------
  UDP SIP Port:           5060
  UDP Bindaddress:        0.0.0.0
  TCP SIP Port:           Disabled
  TLS SIP Port:           Disabled
  Videosupport:           Yes
  Textsupport:            No
  AutoCreate Peer:        No
  Ignore SDP sess. ver.:  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:    Yes
  Call limit peers only:  Yes
  Direct RTP setup:       No
  User Agent:             Asterisk PBX 1.6.0.26-FONCORE-r78
  SDP Session Name:       Asterisk PBX 1.6.0.26-FONCORE-r78
  SDP Owner Name:         *******
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Caller ID:              Unknown
  From: Domain:
  Record SIP history:     Off
  Call Events:            Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          -1
  SIP realtime:           Disabled
  Qualify Freq :          60000 ms
trixvoip*CLI>
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:   Yes
  Jitterbuffer forced:    Yes
  Jitterbuffer max size:  750
  Jitterbuffer resync:    -1
  Jitterbuffer impl:      adaptive
  Jitterbuffer log:       No
trixvoip*CLI>
Network Settings:
---------------------------
  SIP address remapping:  Enabled using externip
  Externhost:             <none>
  Externip:               x.x.x.x:5060
  Externrefresh:          10
  Internal IP:            10.0.0.10:5060
  Localnet:               10.0.0.0/255.255.255.0
  STUN server:            0.0.0.0:0
trixvoip*CLI>
Global Signalling Settings:
---------------------------
  Codecs:                 0x28000c (ulaw|alaw|h263|h264)
  Codec Order:            ulaw:20,alaw:20
  Relax DTMF:             No
  RFC2833 Compensation:   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:   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:      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:     No
trixvoip*CLI>
Default Settings:
-----------------
  Context:                from-sip-external
  Nat:                    Always
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   *97