[general]
udpbindaddr=172.20.255.40
transport=udp,tcp
tcpenable=yes
tlsenable=no
tcpbindaddr=172.20.255.40
directrtpsetup=no
directmedia=yes
allowguest=no
match_auth_username=yes
tos_sip=AF31
tos_audio=ef
tos=0xB8
tos_video=af41                 ; Sets TOS for RTP video packets.
tos_text=af41                  ; Sets TOS for RTP text packets.
trustrpid = yes                 ; If Remote-Party-ID should be trusted
sendrpid = yes                 ; If Remote-Party-ID should be sent (defaults to no)
disallow=all
allow=alaw
allow=ulaw
allow=g729
maxforwards=70
relaxdtmf=yes
rpid_update = yes
maxexpiry=400
minexpiry=60
defaultexpiry=300
qualify=yes ;
notifycid = yes ; Control whether caller ID information is sent along with dialog-info+xml notifications (supported by snom phones)
qualifyfreq=300
qualifypeers=1
qualifygap=2000
registertimeout=20
registerattempts=10
progressinband=never
ignoreregexpire=yes


Global Settings:
----------------
  UDP Bindaddress:        172.20.255.40:5060
  TCP SIP Bindaddress:    172.20.255.40:5060
  TLS SIP Bindaddress:    Disabled
  Videosupport:           No
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        No
  Match Auth Username:    Yes
  Allow unknown access:   No
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain 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:             AsteriskPBX
  SDP Session Name:       Asterisk PBX 1.8.14.0
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Legacy userfield parse: No
  Caller ID:              Unknown
  From: Domain:
  Record SIP history:     Off
  Call Events:            Off
  Auth. Failure Events:   Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          -1
  SIP realtime:           Disabled
  Qualify Freq :          300000 ms
  Q.850 Reason header:    No
  Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
  IP ToS SIP:             AF31
  IP ToS RTP audio:       EF
  IP ToS RTP video:       AF41
  IP ToS RTP text:        AF41
  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:                 0xc (ulaw|alaw)
  Codec Order:            ulaw:20,alaw:20
  Relax DTMF:             Yes
  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:      400 secs
  Reg. default duration:  300 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 10
  Notify ringing state:   Yes
    Include CID:          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:     Yes
  Max forwards:           70

Default Settings:
-----------------
  Allowed transports:     TCP,UDP
  Outbound transport:     UDP
  Context:                from-sip-external
  Force rport:            Yes
  DTMF:                   rfc2833
  Qualify:                2000
  Use ClientCode:         No
  Progress inband:        Never
  Language:
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   *97

----
RAI-ASTERISK-01*CLI> sip show peer 3217


  * Name       : 3217
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : uae
  Subscr.Cont. : <Not set>
  Language     :
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  MOH Suggest  :
  Mailbox      : 3217@device
  VM Extension : *97
  LastMsgsSent : 0/0
  Call limit   : 2147483647
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "Zohair Raza" <3217>
  MaxCallBR    : 384 kbps
  Expire       : 13
  Insecure     : no
  Force rport  : Yes
  ACL          : Yes
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : Yes
  Send RPID    : Yes
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       :
  Addr->IP     : 10.100.210.4:51139
  Defaddr->IP  : (null)
  Prim.Transp. : TCP
  Allowed.Trsp : TCP,UDP
  Def. Username: 3217
  SIP Options  : (none)
  Codecs       : 0x100 (g729)
  Codec Order  : (g729:40)
  Auto-Framing :  No
  Status       : OK (589 ms)
  Useragent    : Cisco-CP7942G/9.3.1
  Reg. Contact : sip:3217@10.100.210.4:51139;transport=tcp
  Qualify Freq : 300000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No
  DND          : No
  CallFwd Num. :