Add template for SIP configuration
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templates/platon/sip.conf.j2
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templates/platon/sip.conf.j2
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; SIP Configuration for Asterisk
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[general]
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context=default ; Default context for incoming calls
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allowguest=yes ; Allow or reject guest calls (default is yes)
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; If your Asterisk is connected to the Internet
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; and you have allowguest=yes
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; you want to check which services you offer everyone
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; out there, by enabling them in the default context (see below).
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;match_auth_username=yes ; if available, match user entry using the
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; 'username' field from the authentication line
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; instead of the From: field.
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allowoverlap=no ; Disable overlap dialing support. (Default is yes)
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;allowoverlap=yes ; Enable RFC3578 overlap dialing support.
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; Can use the Incomplete application to collect the
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; needed digits from an ambiguous dialplan match.
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;allowoverlap=dtmf ; Enable overlap dialing support using DTMF delivery
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; methods (inband, RFC2833, SIP INFO) in the early
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; media phase. Uses the Incomplete application to
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; collect the needed digits.
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;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
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; Default is enabled. The Dial() options 't' and 'T' are not
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; related as to whether SIP transfers are allowed or not.
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;realm=mydomain.tld ; Realm for digest authentication
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; defaults to "asterisk". If you set a system name in
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; asterisk.conf, it defaults to that system name
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; Realms MUST be globally unique according to RFC 3261
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; Set this to your host name or domain name
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;domainsasrealm=no ; Use domains list as realms
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; You can serve multiple Realms specifying several
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; 'domain=...' directives (see below).
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; In this case Realm will be based on request 'From'/'To' header
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; and should match one of domain names.
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; Otherwise default 'realm=...' will be used.
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udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
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; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
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tcpenable=no ; Enable server for incoming TCP connections (default is no)
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tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
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; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
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;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
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;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
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; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
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; Remember that the IP address must match the common name (hostname) in the
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; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
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; For details how to construct a certificate for SIP see
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; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
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;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
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; of seconds a client has to authenticate. If
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; the client does not authenticate beofre this
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; timeout expires, the client will be
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; disconnected. (default: 30 seconds)
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;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
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; unauthenticated sessions that will be allowed
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; to connect at any given time. (default: 100)
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transport=udp ; Set the default transports. The order determines the primary default transport.
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; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
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srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; Note: Asterisk only uses the first host
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; in SRV records
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; Disabling DNS SRV lookups disables the
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; ability to place SIP calls based on domain
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; names to some other SIP users on the Internet
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; Specifying a port in a SIP peer definition or
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; when dialing outbound calls will supress SRV
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; lookups for that peer or call.
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;pedantic=yes ; Enable checking of tags in headers,
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; international character conversions in URIs
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; and multiline formatted headers for strict
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; SIP compatibility (defaults to "yes")
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; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
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;tos_sip=cs3 ; Sets TOS for SIP packets.
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;tos_audio=ef ; Sets TOS for RTP audio packets.
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;tos_video=af41 ; Sets TOS for RTP video packets.
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;tos_text=af41 ; Sets TOS for RTP text packets.
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;cos_sip=3 ; Sets 802.1p priority for SIP packets.
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;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
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;cos_video=4 ; Sets 802.1p priority for RTP video packets.
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;cos_text=3 ; Sets 802.1p priority for RTP text packets.
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;maxexpiry=3600 ; Maximum allowed time of incoming registrations
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; and subscriptions (seconds)
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;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
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;defaultexpiry=120 ; Default length of incoming/outgoing registration
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;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
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;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
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; Default value is 70
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;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
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; and reported in milliseconds with sip show settings.
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; Set to low value if you use low timeout for NAT of UDP sessions
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; Default: 60
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;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
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; Default: 100
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;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
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; Default: 1
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;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
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;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
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; fully. Enable this option to not get error messages
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; when sending MWI to phones with this bug.
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;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
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; the From: header as the "name" portion. Also fill the
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; "user" portion of the URI in the From: header with this
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; value if no fromuser is set
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; Default: empty
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;vmexten=voicemail ; dialplan extension to reach mailbox sets the
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; Message-Account in the MWI notify message
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; defaults to "asterisk"
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; Codec negotiation
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;
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; When Asterisk is receiving a call, the codec will initially be set to the
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; first codec in the allowed codecs defined for the user receiving the call
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; that the caller also indicates that it supports. But, after the caller
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; starts sending RTP, Asterisk will switch to using whatever codec the caller
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; is sending.
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;
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; When Asterisk is placing a call, the codec used will be the first codec in
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; the allowed codecs that the callee indicates that it supports. Asterisk will
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; *not* switch to whatever codec the callee is sending.
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;
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preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
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; rather than advertising all joint codec capabilities. This
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; limits the other side's codec choice to exactly what we prefer.
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;disallow=all ; First disallow all codecs
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allow=alaw
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allow=ulaw ; Allow codecs in order of preference
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allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
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; for framing options
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;
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; This option specifies a preference for which music on hold class this channel
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; should listen to when put on hold if the music class has not been set on the
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; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
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; channel putting this one on hold did not suggest a music class.
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;
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; This option may be specified globally, or on a per-user or per-peer basis.
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;
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;mohinterpret=default
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;
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; This option specifies which music on hold class to suggest to the peer channel
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; when this channel places the peer on hold. It may be specified globally or on
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; a per-user or per-peer basis.
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;
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;mohsuggest=default
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;
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;parkinglot=plaza ; Sets the default parking lot for call parking
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; This may also be set for individual users/peers
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; Parkinglots are configured in features.conf
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;language=en ; Default language setting for all users/peers
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; This may also be set for individual users/peers
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;relaxdtmf=yes ; Relax dtmf handling
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;trustrpid = no ; If Remote-Party-ID should be trusted
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;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
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;sendrpid = rpid ; Use the "Remote-Party-ID" header
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; to send the identity of the remote party
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; This is identical to sendrpid=yes
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;sendrpid = pai ; Use the "P-Asserted-Identity" header
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; to send the identity of the remote party
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;rpid_update = no ; In certain cases, the only method by which a connected line
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; change may be immediately transmitted is with a SIP UPDATE request.
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; If communicating with another Asterisk server, and you wish to be able
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; transmit such UPDATE messages to it, then you must enable this option.
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; Otherwise, we will have to wait until we can send a reinvite to
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; transmit the information.
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;prematuremedia=no ; Some ISDN links send empty media frames before
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; the call is in ringing or progress state. The SIP
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; channel will then send 183 indicating early media
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; which will be empty - thus users get no ring signal.
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; Setting this to "yes" will stop any media before we have
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; call progress (meaning the SIP channel will not send 183 Session
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; Progress for early media). Default is "yes". Also make sure that
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; the SIP peer is configured with progressinband=never.
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;
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; In order for "noanswer" applications to work, you need to run
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; the progress() application in the priority before the app.
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;progressinband=never ; If we should generate in-band ringing always
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; use 'never' to never use in-band signalling, even in cases
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; where some buggy devices might not render it
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; Valid values: yes, no, never Default: never
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useragent=N39 Door PBX ; Allows you to change the user agent string
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; The default user agent string also contains the Asterisk
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; version. If you don't want to expose this, change the
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; useragent string.
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;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
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; Note that promiscredir when redirects are made to the
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; local system will cause loops since Asterisk is incapable
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; of performing a "hairpin" call.
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;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
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; a valid phone number
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;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
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; Other options:
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; info : SIP INFO messages (application/dtmf-relay)
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; shortinfo : SIP INFO messages (application/dtmf)
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; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
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; auto : Use rfc2833 if offered, inband otherwise
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;compactheaders = yes ; send compact sip headers.
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;
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;videosupport=yes ; Turn on support for SIP video. You need to turn this
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; on in this section to get any video support at all.
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; You can turn it off on a per peer basis if the general
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; video support is enabled, but you can't enable it for
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; one peer only without enabling in the general section.
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; If you set videosupport to "always", then RTP ports will
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; always be set up for video, even on clients that don't
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; support it. This assists callfile-derived calls and
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; certain transferred calls to use always use video when
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; available. [yes|NO|always]
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;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
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; Videosupport and maxcallbitrate is settable
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; for peers and users as well
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;callevents=no ; generate manager events when sip ua
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; performs events (e.g. hold)
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;authfailureevents=no ; generate manager "peerstatus" events when peer can't
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; authenticate with Asterisk. Peerstatus will be "rejected".
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alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
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; for any reason, always reject with an identical response
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; equivalent to valid username and invalid password/hash
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; instead of letting the requester know whether there was
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; a matching user or peer for their request. This reduces
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; the ability of an attacker to scan for valid SIP usernames.
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; This option is set to "yes" by default.
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;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
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; INVITE requests are. By default this option is disabled.
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;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
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; order instead of RFC3551 packing order (this is required
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; for Sipura and Grandstream ATAs, among others). This is
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; contrary to the RFC3551 specification, the peer _should_
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; be negotiating AAL2-G726-32 instead :-(
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;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
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; as any IP address used for staticly defined
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; hosts. This helps avoid the configuration
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; error of allowing your users to register at
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; the same address as a SIP provider.
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;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
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;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
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; register their phones.
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contactpermit=172.23.48.0/255.255.252.0
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;engine=asterisk ; RTP engine to use when communicating with the device
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;
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; If regcontext is specified, Asterisk will dynamically create and destroy a
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; NoOp priority 1 extension for a given peer who registers or unregisters with
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; us and have a "regexten=" configuration item.
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; Multiple contexts may be specified by separating them with '&'. The
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; actual extension is the 'regexten' parameter of the registering peer or its
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; name if 'regexten' is not provided. If more than one context is provided,
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; the context must be specified within regexten by appending the desired
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; context after '@'. More than one regexten may be supplied if they are
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; separated by '&'. Patterns may be used in regexten.
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;
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;regcontext=sipregistrations
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;regextenonqualify=yes ; Default "no"
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; If you have qualify on and the peer becomes unreachable
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; this setting will enforce inactivation of the regexten
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; extension for the peer
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;legacy_useroption_parsing=yes ; Default "no" ; If you have this option enabled and there are semicolons
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; in the user field of a sip URI, the field be truncated
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; at the first semicolon seen. This effectively makes
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; semicolon a non-usable character for peer names, extensions,
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; and maybe other, less tested things. This can be useful
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; for improving compatability with devices that like to use
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; user options for whatever reason. The behavior is similar to
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; how SIP URI's were typically handled in 1.6.2, hence the name.
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; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
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; in square brackets. For example, the caller id value 555.5555 becomes 5555555
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; when this option is enabled. Disabling this option results in no modification
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; of the caller id value, which is necessary when the caller id represents something
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; that must be preserved. This option can only be used in the [general] section.
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; By default this option is on.
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;
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;shrinkcallerid=yes ; on by default
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;use_q850_reason = no ; Default "no"
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; Set to yes add Reason header and use Reason header if it is available.
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;--------------------------- SIP timers ----------------------------------------------------
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; These timers are used primarily in INVITE transactions.
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; The default for Timer T1 is 500 ms or the measured run-trip time between
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; Asterisk and the device if you have qualify=yes for the device.
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;
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;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
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; Defaults to 100 ms
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;timert1=500 ; Default T1 timer
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; Defaults to 500 ms or the measured round-trip
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; time to a peer (qualify=yes).
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;timerb=32000 ; Call setup timer. If a provisional response is not received
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; in this amount of time, the call will autocongest
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; Defaults to 64*timert1
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;--------------------------- RTP timers ----------------------------------------------------
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; These timers are currently used for both audio and video streams. The RTP timeouts
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; are only applied to the audio channel.
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; The settings are settable in the global section as well as per device
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;
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;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
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; on the audio channel
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; when we're not on hold. This is to be able to hangup
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; a call in the case of a phone disappearing from the net,
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; like a powerloss or grandma tripping over a cable.
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;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
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; on the audio channel
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; when we're on hold (must be > rtptimeout)
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;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
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; (default is off - zero)
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;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
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; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
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; This mechanism can detect and reclaim SIP channels that do not terminate through normal
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; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
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; The operation of Session-Timers is driven by the following configuration parameters:
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;
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; * session-timers - Session-Timers feature operates in the following three modes:
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; originate : Request and run session-timers always
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; accept : Run session-timers only when requested by other UA
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; refuse : Do not run session timers in any case
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; The default mode of operation is 'accept'.
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; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
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; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
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; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
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;
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;session-timers=originate
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;session-expires=600
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;session-minse=90
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;session-refresher=uas
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;
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;--------------------------- SIP DEBUGGING ---------------------------------------------------
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;sipdebug = yes ; Turn on SIP debugging by default, from
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; the moment the channel loads this configuration
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;recordhistory=yes ; Record SIP history by default
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; (see sip history / sip no history)
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;dumphistory=yes ; Dump SIP history at end of SIP dialogue
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; SIP history is output to the DEBUG logging channel
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;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
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; You can subscribe to the status of extensions with a "hint" priority
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; (See extensions.conf.sample for examples)
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; chan_sip support two major formats for notifications: dialog-info and SIMPLE
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;
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; You will get more detailed reports (busy etc) if you have a call counter enabled
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; for a device.
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;
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; If you set the busylevel, we will indicate busy when we have a number of calls that
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; matches the busylevel treshold.
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;
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; For queues, you will need this level of detail in status reporting, regardless
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; if you use SIP subscriptions. Queues and manager use the same internal interface
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; for reading status information.
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;
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; Note: Subscriptions does not work if you have a realtime dialplan and use the
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; realtime switch.
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;
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;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
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;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
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; Useful to limit subscriptions to local extensions
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; Settable per peer/user also
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;notifyringing = no ; Control whether subscriptions already INUSE get sent
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; RINGING when another call is sent (default: yes)
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;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
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; Turning on notifyringing and notifyhold will add a lot
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; more database transactions if you are using realtime.
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;notifycid = yes ; Control whether caller ID information is sent along with
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; dialog-info+xml notifications (supported by snom phones).
|
||||
; Note that this feature will only work properly when the
|
||||
; incoming call is using the same extension and context that
|
||||
; is being used as the hint for the called extension. This means
|
||||
; that it won't work when using subscribecontext for your sip
|
||||
; user or peer (if subscribecontext is different than context).
|
||||
; This is also limited to a single caller, meaning that if an
|
||||
; extension is ringing because multiple calls are incoming,
|
||||
; only one will be used as the source of caller ID. Specify
|
||||
; 'ignore-context' to ignore the called context when looking
|
||||
; for the caller's channel. The default value is 'no.' Setting
|
||||
; notifycid to 'ignore-context' also causes call-pickups attempted
|
||||
; via SNOM's NOTIFY mechanism to set the context for the call pickup
|
||||
; to PICKUPMARK.
|
||||
;callcounter = yes ; Enable call counters on devices. This can be set per
|
||||
; device too.
|
||||
|
||||
;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
|
||||
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
|
||||
|
||||
register => {{ gatekeeper_sip_registration }}/s
|
||||
;
|
||||
; This will pass incoming calls to the 's' extension
|
||||
|
||||
;----------------------------------- MEDIA HANDLING --------------------------------
|
||||
; By default, Asterisk tries to re-invite media streams 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 well in the case where Asterisk is outside and the
|
||||
; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
|
||||
;
|
||||
;directmedia=yes ; Asterisk by default tries to redirect the
|
||||
; RTP media stream 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 want Asterisk to
|
||||
; stay in the audio path, you may want to turn this off.
|
||||
|
||||
; 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. Note that direct T.38 is not supported.
|
||||
|
||||
;directmedia=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).
|
||||
|
||||
;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
|
||||
; instead of INVITE. This can be combined with 'nonat', as
|
||||
; 'directmedia=update,nonat'. It implies 'yes'.
|
||||
|
||||
;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 directmedia is enabled when
|
||||
; the device is actually behind NAT.
|
||||
|
||||
;directmediadeny=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict
|
||||
;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other
|
||||
; (There is no default setting, this is just an example)
|
||||
; Use this if some of your phones are on IP addresses that
|
||||
; can not reach each other directly. This way you can force
|
||||
; RTP to always flow through asterisk in such cases.
|
||||
|
||||
;ignoresdpversion=yes ; By default, Asterisk will honor the session version
|
||||
; number in SDP packets and will only modify the SDP
|
||||
; session if the version number changes. This option will
|
||||
; force asterisk to ignore the SDP session version number
|
||||
; and treat all SDP data as new data. This is required
|
||||
; for devices that send us non standard SDP packets
|
||||
; (observed with Microsoft OCS). By default this option is
|
||||
; off.
|
||||
|
||||
;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
|
||||
; Like the useragent parameter, the default user agent string
|
||||
; also contains the Asterisk version.
|
||||
;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
|
||||
; This field MUST NOT contain spaces
|
||||
;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.
|
||||
|
||||
;----------------------------------------- REALTIME SUPPORT ------------------------
|
||||
; For additional information on ARA, the Asterisk Realtime Architecture,
|
||||
; 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
|
||||
; 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.
|
||||
|
||||
;------------------------------ Advice of Charge CONFIGURATION --------------------------
|
||||
; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and
|
||||
; AOC-E to snom endpoints. This option can be used both in the
|
||||
; peer and global scope. The default for this option is off.
|
||||
|
||||
|
||||
;------------------------------ 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.
|
||||
|
||||
; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
|
||||
; The option represents the number of milliseconds by which the new jitter buffer
|
||||
; will pad its size. the default is 40, so without modification, the new
|
||||
; jitter buffer will set its size to the jitter value plus 40 milliseconds.
|
||||
; increasing this value may help if your network normally has low jitter,
|
||||
; 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]
|
||||
; 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
|
||||
|
||||
|
||||
[basic-options](!) ; a template
|
||||
dtmfmode=rfc2833
|
||||
context=from-office
|
||||
type=friend
|
||||
|
||||
[natted-phone](!,basic-options) ; another template inheriting basic-options
|
||||
directmedia=no
|
||||
host=dynamic
|
||||
|
||||
[public-phone](!,basic-options) ; another template inheriting basic-options
|
||||
directmedia=yes
|
||||
|
||||
[my-codecs](!) ; a template for my preferred codecs
|
||||
disallow=all
|
||||
allow=ilbc
|
||||
allow=g729
|
||||
allow=gsm
|
||||
allow=g723
|
||||
allow=ulaw
|
||||
|
||||
[ulaw-phone](!) ; and another one for ulaw-only
|
||||
disallow=all
|
||||
allow=ulaw
|
||||
Loading…
Reference in a new issue