sip_nat.conf
nat=yes
externip=69.70.114.174
localnet=10.0.0.0/255.255.255.0
sip.conf
; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.
[general]
#include sip_general_additional.conf
language=fr
bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; 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
sip_additionnal.conf
[334]
type=friend
secret=secret
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=334@default
host=dynamic
dtmfmode=rfc2833
disallow=
dial=SIP/334
context=from-internal
canreinvite=no
callgroup=
callerid=device <334>
call-limit=4
allow=
accountcode=
call-limit=50