;
; SIP Configuration example 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
;
;   reload chan_sip.so		Reload configuration file
;				Active SIP peers will not be reconfigured
;

[general]
context=sip 			; Default context for incoming calls
;allowguest=no			; Allow or reject guest calls (default is yes, this can also be set to 'osp'
				; if asterisk was compiled with OSP support.
;realm=         		; 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
bindport=5080			; UDP Port to bind to (SIP standard port is 5060)
;bindaddr=127.0.0.1		; IP address to bind to (0.0.0.0 binds to all)
bindaddr=0.0.0.0
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
checkmwi=10			    ; Default time between mailbox checks for peers
videosupport=no 		; Turn on support for SIP video
recordhistory=no		; Record SIP history by default 
				; (see sip history / sip no history)

disallow=all			; First disallow all codecs
allow=ulaw			; Allow codecs in order of preference
;allow=ilbc			; 
;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=120			; Terminate call if 60 seconds of no RTP activity
				; when we're not on hold
rtpholdtimeout=240		; 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=never		; If we should generate in-band ringing always
				; use 'never' to never use in-band signalling, even in cases
				; where some buggy devices might not render it
;useragent=Asterisk PBX		; Allows you to change the user agent string
;promiscredir = no      	; If yes, allows 302 or REDIR to non-local SIP address
	                       	; Note that promiscredir when redirects are made to the
       	                	; local system will cause loops since SIP is incapable
       	                	; of performing a "hairpin" call.
;usereqphone = no		; If yes, ";user=phone" is added to uri that contains
				; a valid phone number
dtmfmode = rfc2833		; Set default dtmfmode for sending DTMF. Default: rfc2833
				; Other options: 
				; info : SIP INFO messages
				; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)

;compactheaders = yes		; send compact sip headers.
sipdebug = yes			; Turn on SIP debugging by default, from
				; the moment the channel loads this configuration

;
; If regcontext is specified, Asterisk will dynamically 
; create and destroy a NoOp priority 1 extension for a given
; peer who registers or unregisters with us.  The actual extension
; is the 'regexten' parameter of the registering peer or its
; name if 'regexten' is not provided.  More than one regexten may be supplied
; if they are separated by '&'.  Patterns may be used in regexten.
;
;regcontext=sipregistrations
;
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
;       register => user[:secret[:authuser]]@host[:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension
; needs to be defined in extensions.conf to be able to accept calls
; from this SIP proxy (provider)
;
; host is either a host name defined in DNS or the name of a 
; section defined below.
;
; Examples:
;
;register => 1234:password@mysipprovider.com	
;
;     This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;
;    Register 2345 at sip provider 'sip_proxy'.  Calls from this provider connect to local 
;    extension 1234 in extensions.conf default context, unless you define 
;    unless you configure a [sip_proxy] section below, and configure a context.
;	 Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
;        Tip 2: Use separate type=peer and type=user sections for SIP providers
;               (instead of type=friend) if you have calls in both directions
  
;registertimeout=20		; retry registration calls every 20 seconds (default)
;registerattempts=10		; Number of registration attempts before we give up
				; 0 = continue forever, hammering the other server until it 
				; accepts the registration
				; Default is 10 tries
;callevents=no			; generate manager events when sip ua performs events (e.g. hold)

;---------------------------------------------- NAT SUPPORT ------------------------
; The externip, externhost and localnet settings are used if you use Asterisk behind
; a NAT device to communicate with services on the outside.

;externip = 200.201.202.203	; Address that we're going to put in outbound SIP messages
				; if we're behind a NAT

				; The externip and localnet is used
				; when registering and communicating with other proxies
				; that we're registered with
;externhost=foo.dyndns.net	; Alternatively you can specify an 
				; external host, and Asterisk will 
				; perform DNS queries periodically.  Not
				; recommended for production 
				; environments!  Use externip instead
;externrefresh=10		; How often to refresh externhost if 
				; used
				; You may add multiple local networks.  A reasonable set of defaults
				; are:
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
;localnet=10.0.0.0/255.0.0.0	; Also RFC1918
;localnet=172.16.0.0/12		; Another RFC1918 with CIDR notation
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network

; The nat= setting is used when Asterisk is on a public IP, communicating with
; devices hidden behind a NAT device (broadband router).
; If you have one-way audio problems, you usually have problems with your NAT 
; configuration or your firewalls support of SIP+RTP ports.
; You configure Asterisk choice of RTP ports for incoming audio in rtp.conf
;
nat=never				; Global NAT settings  (Affects all peers and users)
                                ; yes = Always ignore info and assume NAT
                                ; no = Use NAT mode only according to RFC3581 
                                ; never = Never attempt NAT mode or RFC3581 support
				; route = Assume NAT, don't send rport 
				; (work around more UNIDEN bugs)

rtcachefriends=yes  ; Cache realtime friends by adding them to the internal list
			        ; just like friends added from the config file only on a
                                ; as-needed basis.
;rtnoupdate=yes ; do not send the update request over realtime.
rtautoclear=no ; Auto-Expire friends created on the fly on the same schedule
                                ; as if it had just registered when the registration expires
                                ; the friend will vanish from the configuration until requested
                                ; again.  If set to an integer, friends expire
				; within this number of seconds instead of the
				; same as the registration interval
rtignoreexpire=yes		; when reading a peer from Realtime, if the peer's registration
				; has expired based on its registration interval, used the stored
				; address information regardless


[authentication]

