[general]
context=default			; Default context for incoming calls
allowoverlap=no			; Disable overlap dialing support. (Default is yes)
bindport=5060			; UDP Port to bind to (SIP standard port is 5060)
				; bindport is the local UDP port that Asterisk will listen on
bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes			; Enable DNS SRV lookups on outbound calls

;----------------------------------------- NAT SUPPORT ------------------------
; The externip, externhost and localnet settings are used if you use Asterisk
; behind a NAT device to communicate with services on the outside.

;externip = 200.201.202.203	; Address that we're going to put in outbound SIP
				; messages if we're behind a NAT

				; The externip and localnet is used
				; when registering and communicating with other proxies
				; that we're registered with
externhost=gbhatia.homeip.net	; Alternatively you can specify an 
				; external host, and Asterisk will 
				; perform DNS queries periodically.  Not
				; recommended for production 
				; environments!  Use externip instead
externrefresh=10		; How often to refresh externhost if 
				; used
				; You may add multiple local networks.  A reasonable 
				; set of defaults are:
localnet=192.168.1.2/255.255.0.0; All RFC 1918 addresses are local networks

; The nat= setting is used when Asterisk is on a public IP, communicating with
; devices hidden behind a NAT device (broadband router).  If you have one-way
; audio problems, you usually have problems with your NAT configuration or your
; firewall's support of SIP+RTP ports.  You configure Asterisk choice of RTP
; ports for incoming audio in rtp.conf
;
;nat=yes				; Global NAT settings  (Affects all peers and users)
                                ; yes = Always ignore info and assume NAT
                                ; no = Use NAT mode only according to RFC3581 (;rport)
                                ; never = Never attempt NAT mode or RFC3581 support
				; route = Assume NAT, don't send rport 
				; (work around more UNIDEN bugs)

;canreinvite=no   		; Asterisk by default tries to redirect the
				; RTP media stream (audio) to go directly from
				; the caller to the callee.  Some devices do not
				; support this (especially if one of them is behind a NAT).
				; The default setting is YES. If you have all clients
				; behind a NAT, or for some other reason wants Asterisk to
				; stay in the audio path, you may want to turn this off.

				; In Asterisk 1.4 this setting also affect direct RTP
				; at call setup (a new feature in 1.4 - setting up the
				; call directly between the endpoints instead of sending
				; a re-INVITE).

[john]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
context=john
type=friend
host=dynamic			; This device needs to register
nat=yes		   	        ; X-Lite is behind a NAT router
canreinvite=no			; Typically set to NO if behind NAT
disallow=all
allow=gsm			; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw