;
; 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=default			; 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=mydomain.tld		; 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=5060			; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes			; 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
				
;domain=mydomain.tld		; Set default domain for this host
				; If configured, Asterisk will only allow
				; INVITE and REFER to non-local domains
				; Use "sip show domains" to list local domains
;domain=mydomain.tld,mydomain-incoming
				; Add domain and configure incoming context
				; for external calls to this domain
;domain=1.2.3.4			; Add IP address as local domain
				; You can have several "domain" settings
;allowexternalinvites=no	; Disable INVITE and REFER to non-local domains
				; Default is yes
;autodomain=yes			; Turn this on to have Asterisk add local host
				; name and local IP to domain list.
;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
;maxexpiry=3600			; Max length of incoming registration we allow
;defaultexpiry=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
;vmexten=voicemail      ; dialplan extension to reach mailbox sets the 
						; Message-Account in the MWI notify message 
						; defaults to "asterisk"
;videosupport=yes		; Turn on support for SIP video
;recordhistory=yes		; 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=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
;sendrpid = yes			; If Remote-Party-ID should be sent
;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
				; Valid values: yes, no, never Default: never
;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)
				; auto : Use rfc2833 if offered, inband otherwise

;compactheaders = yes		; send compact sip headers.
;sipdebug = yes			; Turn on SIP debugging by default, from
				; the moment the channel loads this configuration
;subscribecontext = default	; Set a specific context for SUBSCRIBE requests
				; Useful to limit subscriptions to local extensions
				; Settable per peer/user also
;notifyringing = yes		; Notify subscriptions on RINGING state

;
; 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 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 0 tries, continue forever
;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
; firewall's support of SIP+RTP ports.  You configure Asterisk choice of RTP
; ports for incoming audio in rtp.conf
;
;nat=no				; 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? (yes|no)

;rtupdate=yes			; Send registry updates to database using realtime? (yes|no)
				; If set to yes, when a SIP UA registers successfully, the ip address,
				; the origination port, the registration period, and the username of
				; the UA will be set to database via realtime. If not present, defaults to 'yes'.

;rtautoclear=yes		; Auto-Expire friends created on the fly on the same schedule
				; as if it had just registered? (yes|no|<seconds>)
				; If set to yes, 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
				; registration interval.

;ignoreregexpire=yes		; Enabling this setting has two functions:
				;
				; For non-realtime peers, when their registration expires, the information
				; will _not_ be removed from memory or the Asterisk database; if you attempt
				; to place a call to the peer, the existing information will be used in spite
				; of it having expired
				;
				; For realtime peers, when the peer is retrieved from realtime storage,
				; the registration information will be used regardless of whether
				; it has expired or not; if it expires while the realtime peer is still in
				; memory (due to caching or other reasons), the information will not be
				; removed from realtime storage

; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
; domains, each of which can direct the call to a specific context if desired.
; By default, all domains are accepted and sent to the default context or the
; context associated with the user/peer placing the call.
; Domains can be specified using:
; domain=<domain>[,<context>]
; Examples:
; domain=myasterisk.dom
; domain=customer.com,customer-context
;
; In addition, all the 'default' domains associated with a server should be
; added if incoming request filtering is desired.
; autodomain=yes
;
; To disallow requests for domains not serviced by this server:
; allowexternaldomains=no

; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
                          ; non-peers, use your primary domain "identity"
                          ; for From: headers instead of just your IP
                          ; address. This is to be polite and
                          ; it may be a mandatory requirement for some
                          ; destinations which do not have a prior
                          ; account relationship with your server. 

;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jb-enable = yes             ; Enables the use of a jitterbuffer on the receiving side of a
                              ; SIP channel. Defaults to "no". An enabled jitterbuffer will
                              ; be used only if the sending side can create and the receiving
                              ; side can not accept jitter. The SIP channel can accept jitter,
                              ; thus a jitterbuffer on the receive SIP side will be used only
                              ; if it is forced and enabled.

; jb-force = no               ; Forces the use of a jitterbuffer on the receive side of a SIP
                              ; channel. Defaults to "no".

; jb-max-size = 200           ; Max length of the jitterbuffer in milliseconds.

; jb-resynch-threshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
                              ; resynchronized. Useful to improve the quality of the voice, with
                              ; big jumps in/broken timestamps, usualy sent from exotic devices
                              ; and programs. Defaults to 1000.

; jb-impl = fixed             ; Jitterbuffer implementation, used on the receiving side of a SIP
                              ; channel. Two implementation are currenlty available - "fixed"
                              ; (with size always equals to jb-max-size) and "adaptive" (with
                              ; variable size, actually the new jb of IAX2). Defaults to fixed.

; jb-log = no                 ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

[authentication]
; Global credentials for outbound calls, i.e. when a proxy challenges your
; Asterisk server for authentication. These credentials override
; any credentials in peer/register definition if realm is matched.
;
; This way, Asterisk can authenticate for outbound calls to other
; realms. We match realm on the proxy challenge and pick an set of 
; credentials from this list
; Syntax:
;	auth = <user>:<secret>@<realm>
;	auth = <user>#<md5secret>@<realm>
; Example:
;auth=mark:topsecret@digium.com
; 
; You may also add auth= statements to [peer] definitions 
; Peer auth= override all other authentication settings if we match on realm

;------------------------------------------------------------------------------
; Users and peers have different settings available. Friends have all settings,
; since a friend is both a peer and a user
;
; User config options:        Peer configuration:
; --------------------        -------------------
; context                     context
; permit                      permit
; deny                        deny
; secret                      secret
; md5secret                   md5secret
; dtmfmode                    dtmfmode
; canreinvite                 canreinvite
; nat                         nat
; callgroup                   callgroup
; pickupgroup                 pickupgroup
; language                    language
; allow                       allow
; disallow                    disallow
; insecure                    insecure
; trustrpid                   trustrpid
; progressinband              progressinband
; promiscredir                promiscredir
; useclientcode               useclientcode
; accountcode                 accountcode
; setvar                      setvar
; callerid		      callerid
; amaflags		      amaflags
; call-limit		      call-limit
; restrictcid		      restrictcid
; subscribecontext	      subscribecontext
;                             mailbox
;                             username
;                             template
;                             fromdomain
;                             regexten
;                             fromuser
;                             host
;                             port
;                             qualify
;                             defaultip
;                             rtptimeout
;                             rtpholdtimeout
;                             sendrpid

;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
; We match on IP address of the proxy for incoming calls 
; since we can not match on username (caller id)
;type=peer
;context=from-fwd
;host=fwd.pulver.com

;[sip_proxy-out]
;type=peer          		; we only want to call out, not be called
;secret=guessit
;username=yourusername		; Authentication user for outbound proxies
;fromuser=yourusername		; Many SIP providers require this!
;fromdomain=provider.sip.domain	
;host=box.provider.com
;usereqphone=yes		; This provider requires ";user=phone" on URI
;call-limit=5			; permit only 5 simultaneous outgoing calls to this peer

;------------------------------------------------------------------------------
; Definitions of locally connected SIP phones
;
; type = user	a device that authenticates to us by "from" field to place calls
; type = peer	a device we place calls to or that calls us and we match by host
; type = friend two configurations (peer+user) in one
;
; For local phones, type=friend works most of the time
;
; If you have one-way audio, you propably have NAT problems. 
; If Asterisk is on a public IP, and the phone is inside of a NAT device
; you will need to configure nat option for those phones.
; Also, turn on qualify=yes to keep the nat session open

;[grandstream1]
;type=friend 			
;context=from-sip		; Where to start in the dialplan when this phone calls
;callerid=John Doe <1234>	; Full caller ID, to override the phones config
;host=192.168.0.23		; we have a static but private IP address
				; No registration allowed
;nat=no				; there is not NAT between phone and Asterisk
;canreinvite=yes		; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info			; either RFC2833 or INFO for the BudgeTone
;call-limit=1			; permit only 1 outgoing call and 1 incoming call at a time
				; from the phone to asterisk
				; (1 for the explicit peer, 1 for the explicit user,
				; remember that a friend equals 1 peer and 1 user in
				; memory)
;mailbox=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=alaw
;allow=g723.1			; Asterisk only supports g723.1 pass-thru!
;allow=g729			; Pass-thru only unless g729 license obtained
;astdb=chan2ext/SIP/grandstream1=1234	; ensures an astDB entry exists


;[xlite1]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
;type=friend
;regexten=1234			; When they register, create extension 1234
;callerid="Jane Smith" <5678>
;host=dynamic			; This device needs to register
;nat=yes			; X-Lite is behind a NAT router
;canreinvite=no			; Typically set to NO if behind NAT
;disallow=all
;allow=gsm			; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
;mailbox=1234@default,1233@default	; Subscribe to status of multiple mailboxes


;[snom]
;type=friend			; Friends place calls and receive calls
;context=from-sip		; Context for incoming calls from this user
;secret=blah
;subscribecontext=localextensions	; Only allow SUBSCRIBE for local extensions
;language=de			; Use German prompts for this user 
;host=dynamic			; This peer register with us
;dtmfmode=inband		; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59		; IP used until peer registers
;mailbox=1234@context,2345      ; Mailbox(-es) for message waiting indicator
;vmexten=voicemail      ; dialplan extension to reach mailbox 
                        ; sets the Message-Account in the MWI notify message
                        ; defaults to global vmexten which defaults to "asterisk"
;restrictcid=yes		; To have the callerid restriced -> sent as ANI
;disallow=all
;allow=ulaw			; dtmfmode=inband only works with ulaw or alaw!


;[polycom]
;type=friend			; Friends place calls and receive calls
;context=from-sip		; Context for incoming calls from this user
;secret=blahpoly
;host=dynamic			; This peer register with us
;dtmfmode=rfc2833		; Choices are inband, rfc2833, or info
;username=polly			; Username to use in INVITE until peer registers
				; Normally you do NOT need to set this parameter
;disallow=all
;allow=ulaw                     ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no		; Polycom phones don't work properly with "never"


;[pingtel]
;type=friend
;secret=blah
;host=dynamic
;insecure=port			; Allow matching of peer by IP address without matching port number
;insecure=invite		; Do not require authentication of incoming INVITEs
;insecure=port,invite		; (both)
;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		; We are in caller groups 1,3,4
;pickupgroup=1,3-5		; We can do call pick-p for call group 1,3,4,5
;defaultip=192.168.0.60		; IP address to use if peer has not registred

;[cisco1]
;type=friend
;secret=blah
;qualify=200			; Qualify peer is no more than 200ms away
;nat=yes			; This phone may be natted
				; Send SIP and RTP to the IP address that packet is 
				; received from instead of trusting SIP headers 
;host=dynamic			; This device registers with us
;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 
				; behind a NAT).
;defaultip=192.168.0.4		; IP address to use until registration
;username=goran			; Username to use when calling this device before registration
				; Normally you do NOT need to set this parameter
;setvar=CUSTID=5678		; Channel variable to be set for all calls from this device

