SIP.CONF - snip
====================
[grandstream1]
type=friend                     ; either "friend" (peer+user), "peer" or "user"
auth=plaintext          ;
context=local
fromuser=6245           ; overrides the callerid, e.g. required by FWD
callerid=John Doe <6245>
host=dynamic            ; we have a static but private IP address
nat=no                          ; there is not NAT between phone and Asterisk
canreinvite=yes         ; allow RTP voice traffic to bypass Asterisk
dtmfmode=info                   ; either RFC2833 or INFO for the BudgeTone
incominglimit=1         ; permit only 1 outgoing call at a time
                                ; from the phone to asterisk
mailbox=1234@default  ; mailbox 1234 in voicemail context "default"
disallow=all                    ; need to disallow=all before we can use allow=
allow=ulaw                      ; Note: In user sections the order of codecs
                                ; listed with allow= does NOT matter!
allow=alaw
allow=g723.1                    ; Asterisk only supports g723.1 pass-thru!
allow=g729                      ; Pass-thru only unless g729 license obtained