sip.conf.sample

;
; 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]
port = 5060			; Port to bind to
bindaddr = 0.0.0.0		; Address to bind SIP channel to
context = default		; Default context for incoming calls
;srvlookup = yes		; Enable DNS SRV lookups on outbound calls
				; Asterisk only uses the first host in SRV records
;pedantic = yes			; Enable slow, pedantic checking for Pingtel
;tos=lowdelay			; IP QoS parameter, either keyword or value
				; like tos=184
;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 NOTIFY
;videosupport=yes		; Turn on support for SIP video

;disallow=all			; Disallow all codecs
;allow=ulaw			; Allow codecs in order of preference
;allow=ilbc

;register => 1234:password@mysipprovider.com	
;Register with a SIP provider

;register => 2345@mysipprovider.com/1234 	
;Register 2345 at sip provider. Calls from this provider connect to local extension 1234 in extensions.conf.

;externip = 200.201.202.203	; Address that we're going to put in outbound SIP messages
				; if we're behind a NAT
;localnet = 192.168.1.0		; Internal NETWORK address
;localmask = 255.255.255.0     	; Internal netmask
				; The externip, localnet and localmask is used
				; when registering and communication with other proxies
				; that we're registred with

;[snomsip]
;type=friend
;secret=blah
;host=dynamic
;dtmfmode=inband		; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59
;mailbox=1234,2345		; Mailbox for message waiting indicator
;restrictcid=yes		; To have the callerid restriced -> sent as ANI

;[pingtel]
;type=friend
;username=pingtel
;secret=blah
;host=dynamic
;qualify=1000			; Consider it down if it's 1 second to reply
				; Helps with NAT session
				; qualify=yes uses default value

;callgroup=1,3-4
;pickupgroup=1,3-4
;defaultip=192.168.0.60

;[cisco]
;type=friend
;username=cisco
;secret=blah
;nat=yes			; This phone may be natted
				; Use IP address that packet is received from
				; instead of trusting SIP headers
;host=dynamic
;canreinvite=no			; Asterisk by default tries to redirect the
				; RTP media stream (audio) to go directly from
				; the caller to the callee.  Some devices do not
				; support this (especially if one of them is 
				; behinda a NAT).
;qualify=200			; Qualify peer is no more than 200ms away
;defaultip=192.168.0.4

;[cisco1]
;type=friend
;username=cisco1
;fromuser=markster		; Specify user to put in "from" instead of callerid
;fromdomain=yourdomain.com	; Specify domain to put in "from" instead of callerid
				; fromuser and fromdomain are used when Asterisk
				; places calls to this account.  It is not used for
				; calls from this account.
;secret=blah
;host=dynamic
;defaultip=192.168.0.4
;amaflags=default		; Choices are default, omit, billing, documentation
;accountcode=markster		; Users may be associated with an accountcode to ease billing