; 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]

bindport=5060			; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
; 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
videosupport=yes

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

[3001]
type=friend
username=3001
secret=hidden
host=dynamic
nat=no
canreinvite=yes
callerid=Joey <3001>
regesten=3001
allow=all
context=from-sip
allowguest=yes
insecure=very


[101]
type=friend
username=101
secret=hidden
host=dynamic
context=internal
callerid=Video Phone <101>
disallow=all ; better for custom-tunning codec selection
allow=ulaw
allow=alaw
allow=gsm
allow=h263 ; H.263 is our video codec
allow=h263p ; H.263p is the enhanced video codec
dtmfmode=rfc2833 ; inband is not supported in compressed codecs like gsm, so we better set it to rfc2833
canreinvite=no ; canreinvite must be set to 'no'