SIP.CONF - snip====================[grandstream1]type=friend ; either "friend" (peer+user), "peer" or "user"auth=plaintext ;context=localfromuser=6245 ; overrides the callerid, e.g. required by FWDcallerid=John Doe <6245>host=dynamic ; we have a static but private IP addressnat=no ; there is not NAT between phone and Asteriskcanreinvite=yes ; allow RTP voice traffic to bypass Asteriskdtmfmode=info ; either RFC2833 or INFO for the BudgeToneincominglimit=1 ; permit only 1 outgoing call at a time ; from the phone to asteriskmailbox=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=alawallow=g723.1 ; Asterisk only supports g723.1 pass-thru!allow=g729 ; Pass-thru only unless g729 license obtained