;
; 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
;
;   module reload chan_sip.so     Reload configuration file
;                                 Active SIP peers will not be reconfigured
;
;------- Naming devices ------------------------------------------------------
;
; When naming devices, make sure you understand how Asterisk matches calls
; that come in.
;	1. Asterisk checks the SIP From: address username and matches against
;	   names of devices with type=user 
;	   The name is the text between square brackets [name]
;	2. Asterisk checks the IP address (and port number) that the INVITE
;	   was sent from and matches against any devices with type=peer
;
; Don't mix extensions with the names of the devices. Devices need a unique
; name. The device name is *not* used as phone numbers. Phone numbers are
; anything you declare as an extension in the dialplan (extensions.conf).
;
; Note: The parameter "username" is not the username and in most cases is
;       not needed at all. Check below. In later releases, it's renamed
;       to "defaultuser" which is a better name, since it is used in 
;       combination with the "defaultip" setting.
;-----------------------------------------------------------------------------

[general]
context=default                 ; Default context for incoming calls
;allowguest=no                  ; Allow or reject guest calls (default is yes)
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
                                ; Default is enabled. The Dial() options 't' and 'T' are not
                                ; related as to whether SIP transfers are allowed or not.
;realm=mydomain.tld             ; Realm for digest authentication
                                ; defaults to "asterisk". If you set a system name in
                                ; asterisk.conf, it defaults to that system name
                                ; 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)
                                ; bindport is the local UDP port that Asterisk will listen on
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
                                ; Specifying a port in a SIP peer definition or
                                ; when dialing outbound calls will supress SRV
                                ; lookups for that peer or call.
                                
;pedantic=yes                   ; Enable checking of tags in headers, 
                                ; international character conversions in URIs
                                ; and multiline formatted headers for strict
                                ; SIP compatibility (defaults to "no")

; See doc/ip-tos.txt for a description of these parameters.
;tos_sip=cs3                    ; Sets TOS for SIP packets.
;tos_audio=ef                   ; Sets TOS for RTP audio packets.
;tos_video=af41                 ; Sets TOS for RTP video packets.

;maxexpiry=3600                 ; Maximum allowed time of incoming registrations
                                ; and subscriptions (seconds)
;minexpiry=60                   ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120              ; Default length of incoming/outgoing registration
;t1min=100                      ; Minimum roundtrip time for messages to monitored hosts
                                ; Defaults to 100 ms
;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10                    ; Default time between mailbox checks for peers
;buggymwi=no                    ; Cisco SIP firmware doesn't support the MWI RFC
                                ; fully. Enable this option to not get error messages
                                ; when sending MWI to phones with this bug.
;vmexten=voicemail              ; dialplan extension to reach mailbox sets the 
                                ; Message-Account in the MWI notify message 
                                ; defaults to "asterisk"

; Codec negotiation
;
; When Asterisk is receiving a call, the codec will initially be set to the
; first codec in the allowed codecs defined for the user receiving the call
; that the caller also indicates that it supports. But, after the caller
; starts sending RTP, Asterisk will switch to using whatever codec the caller
; is sending.
;
; When Asterisk is placing a call, the codec used will be the first codec in
; the allowed codecs that the callee indicates that it supports. Asterisk will
; *not* switch to whatever codec the callee is sending.
;
;disallow=all                   ; First disallow all codecs
;allow=ulaw                     ; Allow codecs in order of preference
;allow=ilbc                     ; see doc/rtp-packetization for framing options

; This option specifies a preference for which music on hold class this channel
; should listen to when put on hold if the music class has not been set on the
; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
; channel putting this one on hold did not suggest a music class.
;
; This option may be specified globally, or on a per-user or per-peer basis.
;
;mohinterpret=default
;
; This option specifies which music on hold class to suggest to the peer channel
; when this channel places the peer on hold. It may be specified globally or on
; a per-user or per-peer basis.
;
;mohsuggest=default
;
;language=en                     ; Default language setting for all users/peers
                                 ; This may also be set for individual users/peers
;relaxdtmf=yes                   ; Relax dtmf handling
;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
;prematuremedia=no               ; Some ISDN links send empty media frames before 
                                 ; the call is in ringing or progress state. The SIP 
                                 ; channel will then send 183 indicating early media
                                 ; which will be empty - thus users get no ring signal.
                                 ; Setting this to "yes" will stop any media before we have
                                 ; call progress (meaning the SIP channel will not send 183 Session
                                 ; Progress for early media). Default is "no". Also make sure that
                                 ; the SIP peer is configured with progressinband=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 Asterisk 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.
;
;videosupport=yes                ; Turn on support for SIP video. You need to turn this on
                                 ; in the this section to get any video support at all.
                                 ; You can turn it off on a per peer basis if the general
                                 ; video support is enabled, but you can't enable it for
                                 ; one peer only without enabling in the general section.
;maxcallbitrate=384              ; Maximum bitrate for video calls (default 384 kb/s)
                                 ; Videosupport and maxcallbitrate is settable
                                 ; for peers and users as well
;callevents=no                   ; generate manager events when sip ua 
                                 ; performs events (e.g. hold)
;alwaysauthreject = yes          ; When an incoming INVITE or REGISTER is to be rejected,
                                 ; for any reason, always reject with an identical response
                                 ; equivalent to valid username and invalid password/hash
                                 ; instead of letting the requester know whether there was
                                 ; a matching user or peer for their request.  This reduces
                                 ; the ability of an attacker to scan for valid SIP usernames.

;g726nonstandard = yes          ; If the peer negotiates G726-32 audio, use AAL2 packing
                                ; order instead of RFC3551 packing order (this is required
                                ; for Sipura and Grandstream ATAs, among others). This is
                                ; contrary to the RFC3551 specification, the peer _should_
                                ; be negotiating AAL2-G726-32 instead :-(

;matchexterniplocally = yes     ; Only substitute the externip or externhost setting if it matches
                                ; your localnet setting. Unless you have some sort of strange network
                                ; setup you will not need to enable this.

;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from registering
                                ; as any IP address used for staticly defined
                                ; hosts.  This helps avoid the configuration
                                ; error of allowing your users to register at
                                ; the same address as a SIP provider.

;contactdeny=0.0.0.0/0.0.0.0           ; Use contactpermit and contactdeny to
;contactpermit=172.16.0.0/255.255.0.0  ; restrict at what IPs your users may
                                       ; register their phones.
;forwardloopdetected=no         ; Attempt to forward a call locally if the
                                ; destination replies with 482 Loop Detected
                                ; default = yes

;
; 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 and have a "regexten=" configuration item.  
; Multiple contexts may be specified by separating them with '&'. The 
; actual extension is the 'regexten' parameter of the registering peer or its
; name if 'regexten' is not provided.  If more than one context is provided,
; the context must be specified within regexten by appending the desired
; context after '@'.  More than one regexten may be supplied if they are 
; separated by '&'.  Patterns may be used in regexten.
;
;regcontext=sipregistrations
;

; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
; in square brackets.  For example, the caller id value 555.5555 becomes 5555555
; when this option is enabled.  Disabling this option results in no modification
; of the caller id value, which is necessary when the caller id represents something
; that must be preserved.  This option can only be used in the [general] section.
; By default this option is on.
;
;shrinkcallerid=yes     ; on by default

;--------------------------- RTP timers ----------------------------------------------------
; These timers are currently used for both audio and video streams. The RTP timeouts
; are only applied to the audio channel.
; The settings are settable in the global section as well as per device
;
;rtptimeout=60                  ; Terminate call if 60 seconds of no RTP or RTCP activity
                                ; on the audio channel
                                ; when we're not on hold. This is to be able to hangup
                                ; a call in the case of a phone disappearing from the net,
                                ; like a powerloss or grandma tripping over a cable.
;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP or RTCP activity
                                ; on the audio channel
                                ; when we're on hold (must be > rtptimeout)
;rtpkeepalive=<secs>            ; Send keepalives in the RTP stream to keep NAT open
                                ; (default is off - zero)
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes                 ; Turn on SIP debugging by default, from
                                ; the moment the channel loads this configuration
;recordhistory=yes              ; Record SIP history by default 
                                ; (see sip history / sip no history)
;dumphistory=yes                ; Dump SIP history at end of SIP dialogue
                                ; SIP history is output to the DEBUG logging channel


;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
; You can subscribe to the status of extensions with a "hint" priority
; (See extensions.conf.sample for examples)
; chan_sip support two major formats for notifications: dialog-info and SIMPLE 
;
; You will get more detailed reports (busy etc) if you have a call limit set
; for a device. When the call limit is filled, we will indicate busy. Note that
; you need at least 2 in order to be able to do attended transfers.
;
; For queues, you will need this level of detail in status reporting, regardless
; if you use SIP subscriptions. Queues and manager use the same internal interface
; for reading status information.
;
; Note: Subscriptions does not work if you have a realtime dialplan and use the
; realtime switch.
;
;allowsubscribe=no              ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default     ; Set a specific context for SUBSCRIBE requests
                                ; Useful to limit subscriptions to local extensions
                                ; Settable per peer/user also
;notifyringing = yes            ; Control whether subscriptions already INUSE get sent
                                ; RINGING when another call is sent (default: no)
;notifyhold = yes               ; Notify subscriptions on HOLD state (default: no)
                                ; Turning on notifyringing and notifyhold will add a lot
                                ; more database transactions if you are using realtime.
;limitonpeers = yes             ; Apply call limits on peers only. This will improve 
                                ; status notification when you are using type=friend
                                ; Inbound calls, that really apply to the user part
                                ; of a friend will now be added to and compared with
                                ; the peer limit instead of applying two call limits,
                                ; one for the peer and one for the user.
                                ; "sip show inuse" will only show active calls on 
                                ; the peer side of a "type=friend" object if this
                                ; setting is turned on.

;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
;
; This setting is available in the [general] section as well as in device configurations.
; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
; both parties have T38 support enabled in their Asterisk configuration 
; This has to be enabled in the general section for all devices to work. You can then
; disable it on a per device basis. 
;
; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
;
; t38pt_udptl = yes            ; Default false
;
;----------------------------------------- OUTBOUND SIP REGISTRATIONS  ------------------------
; 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

;----------------------------------------- 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 (;rport)
                                ; never = Never attempt NAT mode or RFC3581 support
                                ; route = Assume NAT, don't send rport 
                                ; (work around more UNIDEN bugs)

;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
; no reason for Asterisk to stay in the media path, the media will be redirected.
; This does not really work with in the case where Asterisk is outside and have
; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
;
;canreinvite=yes                ; 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).
                                ; The default setting is YES. If you have all clients
                                ; behind a NAT, or for some other reason wants Asterisk to
                                ; stay in the audio path, you may want to turn this off.

                                ; In Asterisk 1.4 this setting also affect direct RTP
                                ; at call setup (a new feature in 1.4 - setting up the
                                ; call directly between the endpoints instead of sending
                                ; a re-INVITE).

                                ; Additionally this option does not disable all reINVITE operations.
                                ; It only controls Asterisk generating reINVITEs for the specific
                                ; purpose of setting up a direct media path. If a reINVITE is
                                ; needed to switch a media stream to inactive (when placed on
                                ; hold) or to T.38, it will still be done, regardless of this
                                ; setting.

;directrtpsetup=yes             ; Enable the new experimental direct RTP setup. This sets up
                                ; the call directly with media peer-2-peer without re-invites.
                                ; Will not work for video and cases where the callee sends 
                                ; RTP payloads and fmtp headers in the 200 OK that does not match the
                                ; callers INVITE. This will also fail if canreinvite is enabled when
                                ; the device is actually behind NAT.

;canreinvite=nonat              ; An additional option is to allow media path redirection
                                ; (reinvite) but only when the peer where the media is being
                                ; sent is known to not be behind a NAT (as the RTP core can
                                ; determine it based on the apparent IP address the media
                                ; arrives from).

;canreinvite=update             ; Yet a third option... use UPDATE for media path redirection,
                                ; instead of INVITE. This can be combined with 'nonat', as
                                ; 'canreinvite=update,nonat'. It implies 'yes'.

;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read realtime.txt and extconfig.txt in the /doc directory of the
; source code.
;
;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)

;rtsavesysname=yes              ; Save systemname in realtime database at registration
                                ; Default= 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'. Note: realtime peers will
                                ; probably not function across reloads in the way that you expect, if
                                ; you turn this option off.
;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

;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
; 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.
; REGISTER to non-local domains will be automatically denied if a domain
; list is configured.
;
; 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

;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
;allowexternaldomains=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.

; 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 --------------------------
; jbenable = 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.

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

; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.

; jbresyncthreshold = 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, usually sent from exotic devices
                              ; and programs. Defaults to 1000.

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

; jblog = 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
; callingpres                 callingpres
; 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
; allowoverlap                allowoverlap
; allowsubscribe              allowsubscribe
; allowtransfer               allowtransfer
; subscribecontext            subscribecontext
; videosupport                videosupport
; maxcallbitrate              maxcallbitrate
; rfc2833compensate           mailbox
; t38pt_usertpsource          username
;                             fromdomain
;                             regexten
;                             fromuser
;                             host
;                             port
;                             qualify
;                             defaultip
;                             rtptimeout
;                             rtpholdtimeout
;                             sendrpid
;                             outboundproxy
;                             rfc2833compensate
;                             t38pt_usertpsource
;                             contactpermit         ; Limit what a host may register as (a neat trick
;                             contactdeny           ; is to register at the same IP as a SIP provider,
;                                                   ; then call oneself, and get redirected to that
;                                                   ; same location).

;[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
;outboundproxy=proxy.provider.domain    ; send outbound signaling to this proxy, not directly to the peer
                                        ; Call-limits will not be enforced on real-time peers,
                                        ; since they are not stored in-memory
;port=80                                ; The port number we want to connect to on the remote side
                                        ; Also used as "defaultport" in combination with "defaultip" settings

;------------------------------------------------------------------------------
; Definitions of locally connected SIP devices
;
; 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 device names, we recommend using only a-z, numerics (0-9) and underscore
; 
; For local phones, type=friend works most of the time
;
; If you have one-way audio, you probably 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
                                ; on incoming calls to Asterisk
;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
                                ; This will affect your subscriptions as well.
                                ; There is no combined call counter for a "friend"
                                ; so there's currently no way in sip.conf to limit
                                ; to one inbound or outbound call per phone. Use
                                ; the group counters in the dial plan for that.
                                ;
;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
;callingpres=allowed_passed_screen        ; Set caller ID presentation
                                ; See doc/callingpres.txt for more information


;[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
;subscribemwi=yes               ; Only send notifications if this phone 
                                ; subscribes for mailbox notification
;vmexten=voicemail              ; dialplan extension to reach mailbox 
                                ; sets the Message-Account in the MWI notify message
                                ; defaults to global vmexten which defaults to "asterisk"
;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
;
; Call group and Pickup group should be in the range from 0 to 63
;
;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 registered
;deny=0.0.0.0/0.0.0.0           ; ACL: Control access to this account based on IP address
;permit=192.168.0.60/255.255.255.0
;permit=192.168.0.60/24          ; we can also use CIDR notation for subnet masks

;[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

;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
;rfc2833compensate=yes          ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
                                ; You must have this turned on or DTMF reception will work improperly.
;t38pt_usertpsource=yes         ; Use the source IP address of RTP as the destination IP address for UDPTL packets
                                ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
                                ; external IP address of the remote device. If port forwarding is done at the client side
                                ; then UDPTL will flow to the remote device.
