; "setvar" to set variables that can be used in the dialplan for various limits.
[general]
-context=default ; Default context for incoming calls
+context=public ; Default context for incoming calls. Defaults to 'default'
;allowguest=no ; Allow or reject guest calls (default is yes)
; If your Asterisk is connected to the Internet
; and you have allowguest=yes
; 'username' field from the authentication line
; instead of the From: field.
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
+;allowoverlap=yes ; Enable RFC3578 overlap dialing support.
+ ; Can use the Incomplete application to collect the
+ ; needed digits from an ambiguous dialplan match.
+;allowoverlap=dtmf ; Enable overlap dialing support using DTMF delivery
+ ; methods (inband, RFC2833, SIP INFO) in the early
+ ; media phase. Uses the Incomplete application to
+ ; collect the needed digits.
;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.
; 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
-;domainsasrealm=no ; Use domans list as realms
+;domainsasrealm=no ; Use domains list as realms
; You can serve multiple Realms specifying several
; 'domain=...' directives (see below).
; In this case Realm will be based on request 'From'/'To' header
;
; Note also that while Asterisk currently will parse an Allow header to learn
; what methods an endpoint supports, the only actual use for this currently
-; is for determining if Asterisk may send connected line UPDATE requests. Its
-; use may be expanded in the future.
+; is for determining if Asterisk may send connected line UPDATE requests and
+; MESSAGE requests. Its use may be expanded in the future.
;
; disallowed_methods = UPDATE
; For details how to construct a certificate for SIP see
; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
+;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
+ ; of seconds a client has to authenticate. If
+ ; the client does not authenticate beofre this
+ ; timeout expires, the client will be
+ ; disconnected. (default: 30 seconds)
+
+;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
+ ; unauthenticated sessions that will be allowed
+ ; to connect at any given time. (default: 100)
+
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; and multiline formatted headers for strict
; SIP compatibility (defaults to "yes")
-; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service 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.
;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
; Default value is 70
;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
+ ; and reported in milliseconds with sip show settings.
; Set to low value if you use low timeout for NAT of UDP sessions
; Default: 60
;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
-;allow=ilbc ; see doc/rtp-packetization for framing options
+;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/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
; Parkinglots are configured in features.conf
;language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
+;tonezone=se ; Default tonezone 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 (defaults to no)
;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
; INVITE requests are. By default this option is disabled.
+;accept_outofcall_message = no ; Disable this option to reject all MESSAGE requests outside of a
+ ; call. By default, this option is enabled. When enabled, MESSAGE
+ ; requests are passed in to the dialplan.
+
+;outofcall_message_context = messages ; Context all out of dialog msgs are sent to. When this
+ ; option is not set, the context used during peer matching
+ ; is used. This option can be defined at both the peer and
+ ; global level.
+
+;auth_message_requests = yes ; Enabling this option will authenticate MESSAGE requests.
+ ; By default this option is enabled. However, it can be disabled
+ ; should an application desire to not load the Asterisk server with
+ ; doing authentication and implement end to end security in the
+ ; message body.
+
;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
; If you have qualify on and the peer becomes unreachable
; this setting will enforce inactivation of the regexten
; extension for the peer
+;legacy_useroption_parsing=yes ; Default "no" ; If you have this option enabled and there are semicolons
+ ; in the user field of a sip URI, the field be truncated
+ ; at the first semicolon seen. This effectively makes
+ ; semicolon a non-usable character for peer names, extensions,
+ ; and maybe other, less tested things. This can be useful
+ ; for improving compatability with devices that like to use
+ ; user options for whatever reason. The behavior is similar to
+ ; how SIP URI's were typically handled in 1.6.2, hence the name.
; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
; in square brackets. For example, the caller id value 555.5555 becomes 5555555
; for their media streams is not actual port number that will be used on the nearer
; side of the NAT.
;
+; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
+; the nat setting in a peer definition, then the peer username will be discoverable
+; by outside parties as Asterisk will respond to different ports for defined and
+; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
+; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
+; other, then valid users with settings differing from those in the general section will
+; be discoverable.
+;
; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
; to receive them on.
;encryption=no ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.
+;encryption_taglen=80 ; Set the auth tag length offered in the INVITE either 32/80 default 80
;----------------------------------------- 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.
+; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
;
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
; just like friends added from the config file only on a
; but occasionally has spikes.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
+
+;----------------------------- SIP_CAUSE reporting ---------------------------------
+; storesipcause = no ; This option causes chan_sip to set the
+ ; HASH(SIP_CAUSE,<channel name>) channel variable
+ ; to the value of the last sip response.
+ ; WARNING: enabling this option carries a
+ ; significant performance burden. It should only
+ ; be used in low call volume situations. This
+ ; option defaults to "no".
+
;-----------------------------------------------------------------------------------
[authentication]
; Asterisk only matches on IP/port, not on names. This is mostly used for SIP
; trunks.
;
+; Use remotesecret for outbound authentication, and secret for authenticating
+; inbound requests. For historical reasons, if no remotesecret is supplied for an
+; outbound registration or call, the secret will be used.
+;
; For device names, we recommend using only a-z, numerics (0-9) and underscore
;
; For local phones, type=friend works most of the time
; use_q850_reason
; maxforwards
; encryption
+; description ; Used to provide a description of the peer in console output
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
type=friend
[natted-phone](!,basic-options) ; another template inheriting basic-options
- nat=yes
directmedia=no
host=dynamic
[public-phone](!,basic-options) ; another template inheriting basic-options
- nat=no
directmedia=yes
[my-codecs](!) ; a template for my preferred codecs
allow=gsm
allow=g723
allow=ulaw
+ ; Or, more simply:
+ ;allow=!all,ilbc,g729,gsm,g723,ulaw
[ulaw-phone](!) ; and another one for ulaw-only
disallow=all
allow=ulaw
+ ; Again, more simply:
+ ;allow=!all,ulaw
; and finally instantiate a few phones
;
;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
+;description=Courtesy Phone ; Description of the peer. Shown when doing 'sip show peers'.
;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
;directmedia=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
;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
;directmedia=no ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm ; GSM consumes far less bandwidth than ulaw
;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
;directmedia=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
projects / asterisk/asterisk.git / blobdiff