;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/username@domain to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/user@proxyhostname
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
;   sip show peers              Show all SIP peers (including friends)
;   sip show users              Show all SIP users (including friends)
;   sip show registry           Show status of hosts we register with
;
;   sip debug                   Show all SIP messages
;

[general]
context=default                 ; Default context for incoming calls
recordhistory=yes               ; Record SIP history by default
                                ; (see sip history / sip no history)
realm=asterisk.localhost.localdomain            ; Realm for digest authentication
                                ; defaults to "asterisk"
                                ; Realms MUST be globally unique according to RFC 3261
                                ; Set this to your host name or domain name
port=5060                       ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=192.168.0.200          ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=no            ; Enable DNS SRV lookups on outbound calls
                                ; Note: Asterisk only uses the first host
                                ; in SRV records
                                ; Disabling DNS SRV lookups disables the
                                ; ability to place SIP calls based on domain
                                ; names to some other SIP users on the Internet

;pedantic=yes                   ; Enable slow, pedantic checking for Pingtel
                                ; and multiline formatted headers for strict
                                ; SIP compatibility (defaults to "no")
;tos=184                        ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay                   ; lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600                ; Max length of incoming registration we allow
;defaultexpirey=120             ; Default length of incoming/outoing registration
;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
;videosupport=yes               ; Turn on support for SIP video

allow=all                       ; First disallow all codecs
;allow=ulaw                     ; Allow codecs in order of preference
;allow=ilbc                     ; Note: codec order is respected only in [general]
;musicclass=default             ; Sets the default music on hold class for all SIP calls
                                ; This may also be set for individual users/peers
;language=en                    ; Default language setting for all users/peers
                                ; This may also be set for individual users/peers
;relaxdtmf=yes                  ; Relax dtmf handling
;rtptimeout=60                  ; Terminate call if 60 seconds of no RTP activity
                                ; when we're not on hold
;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP activity
                                ; when we're on hold (must be > rtptimeout)
;trustrpid = no                 ; If Remote-Party-ID should be trusted
;progressinband=no              ; If we should generate in-band ringing always
;useragent=Asterisk PBX         ; Allows you to change the user agent string
;nat=no                         ; NAT settings
                                ; yes = Always ignore info and assume NAT
                                ; no = Use NAT mode only according to RFC3581
                                ; never = Never attempt NAT mode or RFC3581 support