OpenPKG Specs: asterisk/sip.conf@acad5c9dea5f (annotated)

asterisk/sip.conf@acad5c9dea5f (annotated)

asterisk/sip.conf

Fri, 22 Oct 2010 19:54:57 +0200

author: Michael Schloh von Bennewitz <michael@schloh.com>
date: Fri, 22 Oct 2010 19:54:57 +0200
changeset 281: acad5c9dea5f
permissions: -rw-r--r--

Correct dependencies and use a canonical package name.

 ;
 ; 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)
 ;allowoverlap=no			; Disable overlap dialing support. (Default is yes)
 ;allowtransfer=no		; Disable all transfers (unless enabled in peers or users)
 				; Default is enabled
 ;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
 ;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
 ;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/README.tos 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"
 ;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
 ;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 '401 Unauthorized'
  				; instead of letting the requester know whether there was
  				; a matching user or peer for their request
 ;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.
 ;
 ; 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
 ;
 ;--------------------------- 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		; Notify subscriptions on RINGING state (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).
 ;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'.
 ;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.
 ; 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
 ;                             username
 ;                             template
 ;                             fromdomain
 ;                             regexten
 ;                             fromuser
 ;                             host
 ;                             port
 ;                             qualify
 ;                             defaultip
 ;                             rtptimeout
 ;                             rtpholdtimeout
 ;                             sendrpid
 ;                             outboundproxy
 ;                             rfc2833compensate
 ;[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 README.callingpres 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
 ;[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.

OpenPKG Specs / annotate

asterisk/sip.conf@acad5c9dea5f (annotated)

asterisk/sip.conf