[general]
port = 5060                 ; port to bind for sip connections
bindaddr = 0.0.0.0          ; ip to bind for sip connections
context = from-sip          ; default context for incoming sip calls
;
disallow = all              ; disallow all codecs, we want to enable, 
allow    = gsm              ; what we deem is necessary
allow    = ilbc
allow    = speex
allow    = g729             ; g729 only works for pass-thru, if you haven't bought a license
allow    = g726
allow    = ulaw             ; i would normally not allow ulaw, because it's high bandwidth,
                            ; but if you want to use free world dialup services, it's the
                            ; only codec they support
;
register => 12345:not_my_password@fwd.pulver.com/marlow-fwd                    ; these make it possible to get calls
register => 17471234567:not_my_password_either@proxy01.sipphone.com/marlow-sip ; in from these services.
;
;
; Dial out to fwdnet
;
[fwd]
  type=peer
  secret=not_my_password
  host=fwd.pulver.com
;
;
; Dial out to sipphone.com
;
[sipphone]
  type=peer
  secret=not_my_password_either
  host=proxy01.sipphone.com
;
;
; This is a local extension running kphone, firefly or likewise
;
[marlow]
  callerid=("Martin List-Petersen" <3986>)
  type=friend
  secret=not_my_password_anyway
  host=dynamic
  context=intern
; canreinvite=no ; "canreinvite=no" is needed, so that all calls allways go through asterisk. This is
;                  needed, when you have the SIP client on the LAN and want to use asterisk as a proxy.
;
; qualify=200
; nat=yes        ; these two are needed, if you asterisk box sits on a public ip
;                  and your sip client/ata box/ip phone behind a nat firewall
;                  usually you would not need a stun server then.