[general]
context=default			; Default context for incoming calls
allowoverlap=yes		; Disable overlap dialing support. (Default is yes)
bindport=5062			; 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)

srvlookup=no			; Enable DNS SRV lookups on outbound calls

disallow=all			; First disallow all codecs
allow=ulaw			; Allow codecs in order of preference
allow=alaw			; see doc/rtp-packetization for framing options
externip=xxx.xxx.xxx.xxx		; Address that we're going to put in outbound SIP
				; messages if we're behind a NAT
localnet=192.168.1.0/24

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)


notifyringing=yes
notifyhold=yes
limitonpeers=yes                ;This setting tells Asterisk to apply call limits to peers only. 
				;This will improve call limits and status notification for devices 
				;set to type=friend because the peer limit will be checked, 
				;and not create a separate limit for the user and peer portions 
				;of a friend
callcounter = yes
canreinvite=no


[50]
type=friend
defaultuser=50
secret=secret50
context=4every1
callerid="Lukasz Gigaset" <50>
host=dynamic
dtmfmode=rfc2833 ; inband is not supported in compressed codecs like gsm, so we better set it to rfc2833 
qualify=yes ; Default 2000ms
mailbox=50
callgroup=1
pickupgroup=1
call-limit=2
subscribecontext=blf_context
nat=no

[51]
type=friend
defaultuser=51
secret=secret51
context=4every1
callerid="Lukasz Acer" <51>
host=dynamic
dtmfmode=rfc2833
qualify=yes
mailbox=51
callgroup=1
call-limit=1
subscribecontext=blf_context

