voip*CLI> sip show peer 334
voip*CLI>

  * Name       : 334
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : from-internal
  Subscr.Cont. : <Not set>
  Language     : fr
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      : 334@default
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 50
  Dynamic      : Yes
  Callerid     : "device" <334>
  MaxCallBR    : 384 kbps
  Expire       : 128
  Insecure     : no
 Nat          : Always
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : 74.56.64.35 Port 60157
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 334
  SIP Options  : (none)
  Codecs       : 0xc (ulaw|alaw)
  Codec Order  : (ulaw:20,alaw:20)
  Auto-Framing:  No
  Status       : OK (61 ms)
  Useragent    : Aastra 57iCT/2.2.0.166
  Reg. Contact : sip:334@74.56.64.35:51620;transport=udp;srcadr=192.168.0.101:5060


sip_nat.conf
nat=yes
externip=69.70.114.174
localnet=10.0.0.0/255.255.255.0

sip.conf
[general]
#include sip_general_additional.conf

language=fr
bindport = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 10.0.0.9    ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; Reported as required for Asterisk 1.4
notifyringing=yes
notifyhold=yes
limitonpeers=yes

; enable and force the sip jitterbuffer. If these settings are desired
; they should be set in the sip_general_custom.conf file as this file
; will get overwritten during reloads and upgrades.
;
; jbenable=yes
; jbforce=yes

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_general_custom.conf
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf
#include sip_custom.conf
#include sip_additional.conf