1542 linhas
72 KiB
Text
1542 linhas
72 KiB
Text
; PJSIP Configuration Samples and Quick Reference
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;
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; This file has several very basic configuration examples, to serve as a quick
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; reference to jog your memory when you need to write up a new configuration.
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; It is not intended to teach PJSIP configuration or serve as an exhaustive
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; reference of options and potential scenarios.
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;
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; This file has two main sections.
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; First, manually written examples to serve as a handy reference.
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; Second, a list of all possible PJSIP config options by section. This is
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; pulled from the XML config help. It only shows the synopsis for every item.
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; If you want to see more detail please check the documentation sources
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; mentioned at the top of this file.
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; ============================================================================
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; NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
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;
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; This file does not maintain the complete option documentation.
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; ============================================================================
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; Documentation
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;
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; The official documentation is at http://wiki.asterisk.org
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; You can read the XML configuration help via Asterisk command line with
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; "config show help res_pjsip", then you can drill down through the various
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; sections and their options.
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;
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;========!!!!!!!!!!!!!!!!!!! SECURITY NOTICE !!!!!!!!!!!!!!!!!!!!===========
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;
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; At a minimum please read the file "README-SERIOUSLY.bestpractices.txt",
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; located in the Asterisk source directory before starting Asterisk.
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; Otherwise you risk allowing the security of the Asterisk system to be
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; compromised. Beyond that please visit and read the security information on
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; the wiki at: https://wiki.asterisk.org/wiki/x/EwFB
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;
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; A few basics to pay attention to:
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;
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; Anonymous Calls
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;
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; By default anonymous inbound calls via PJSIP are not allowed. If you want to
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; route anonymous calls you'll need to define an endpoint named "anonymous".
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; res_pjsip_endpoint_identifier_anonymous.so handles that functionality so it
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; must be loaded. It is not recommended to accept anonymous calls.
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;
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; Access Control Lists
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;
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; See the example ACL configuration in this file. Read the configuration help
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; for the section and all of its options. Look over the samples in acl.conf
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; and documentation at https://wiki.asterisk.org/wiki/x/uA80AQ
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; If possible, restrict access to only networks and addresses you trust.
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;
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; Dialplan Contexts
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;
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; When defining configuration (such as an endpoint) that links into
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; dialplan configuration, be aware of what that dialplan does. It's easy to
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; accidentally provide access to internal or outbound dialing extensions which
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; could cost you severely. The "context=" line in endpoint configuration
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; determines which dialplan context inbound calls will enter into.
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;
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;=============================================================================
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; Overview of Configuration Section Types Used in the Examples
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;
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; * Transport "transport"
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; * Configures res_pjsip transport layer interaction.
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; * Endpoint "endpoint"
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; * Configures core SIP functionality related to SIP endpoints.
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; * Authentication "auth"
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; * Stores inbound or outbound authentication credentials for use by trunks,
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; endpoints, registrations.
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; * Address of Record "aor"
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; * Stores contact information for use by endpoints.
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; * Endpoint Identification "identify"
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; * Maps a host directly to an endpoint
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; * Access Control List "acl"
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; * Defines a permission list or references one stored in acl.conf
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; * Registration "registration"
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; * Contains information about an outbound SIP registration
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; * Resource Lists
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; * Contains information for configuring resource lists.
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; * Phone Provisioning "phoneprov"
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; * Contains information needed by res_phoneprov for autoprovisioning
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; The following sections show example configurations for various scenarios.
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; Most require a couple or more configuration types configured in concert.
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;=============================================================================
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; Naming of Configuration Sections
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;
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; Configuration section names are denoted with enclosing brackets,
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; e.g. [6001]
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; In most cases, you can name a section whatever makes sense to you. For example
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; you might name a transport [transport-udp-nat] to help you remember how that
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; section is being used. However, in some cases, ("endpoint" and "aor" types)
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; the section name has a relationship to its function.
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;
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; Depending on the modules loaded, Asterisk can match SIP requests to an
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; endpoint or aor in a few ways:
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;
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; 1) Match a section name for endpoint type sections to the username in the
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; "From" header of inbound SIP requests.
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; 2) Match a section name for aor type sections to the username in the "To"
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; header of inbound SIP REGISTER requests.
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; 3) With an identify type section configured, match an inbound SIP request of
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; any type to an endpoint or aor based on the IP source address of the
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; request.
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;
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; Note that sections can have the same name as long as their "type" options are
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; set to different values. In most cases it makes sense to have associated
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; configuration sections use the same name, as you'll see in the examples within
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; this file.
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;===============EXAMPLE TRANSPORTS============================================
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;
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; A few examples for potential transport options.
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;
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; For the NAT transport example, be aware that the options starting with
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; the prefix "external_" will only apply to communication with addresses
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; outside the range set with "local_net=".
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;
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; You can have more than one of any type of transport, as long as it doesn't
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; use the same resources (bind address, port, etc) as the others.
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; Basic UDP transport
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;
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;[transport-udp]
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;type=transport
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;protocol=udp ;udp,tcp,tls,ws,wss
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;bind=0.0.0.0
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; UDP transport behind NAT
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;
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;[transport-udp-nat]
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;type=transport
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;protocol=udp
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;bind=0.0.0.0
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;local_net=192.0.2.0/24
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;external_media_address=203.0.113.1
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;external_signaling_address=203.0.113.1
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; Basic IPv6 UDP transport
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;
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;[transport-udp-ipv6]
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;type=transport
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;protocol=udp
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;bind=::
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; Example IPv4 TLS transport
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;
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;[transport-tls]
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;type=transport
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;protocol=tls
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;bind=0.0.0.0
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;cert_file=/path/mycert.crt
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;priv_key_file=/path/mykey.key
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;cipher=ADH-AES256-SHA,ADH-AES128-SHA
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;method=tlsv1
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;===============OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION============
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;
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; This is a simple registration that works with some SIP trunking providers.
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; You'll need to set up the auth example "mytrunk_auth" below to enable outbound
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; authentication. Note that we "outbound_auth=" use for outbound authentication
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; instead of "auth=", which is for inbound authentication.
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;
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; If you are registering to a server from behind NAT, be sure you assign a transport
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; that is appropriately configured with NAT related settings. See the NAT transport example.
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;
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; "contact_user=" sets the SIP contact header's user portion of the SIP URI
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; this will affect the extension reached in dialplan when the far end calls you at this
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; registration. The default is 's'.
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;
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; If you would like to enable line support and have incoming calls related to this
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; registration go to an endpoint automatically the "line" and "endpoint" options must
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; be set. The "endpoint" option specifies what endpoint the incoming call should be
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; associated with.
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;[mytrunk]
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;type=registration
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;transport=transport-udp
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;outbound_auth=mytrunk_auth
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;server_uri=sip:sip.example.com
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;client_uri=sip:1234567890@sip.example.com
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;contact_user=1234567890
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;retry_interval=60
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;forbidden_retry_interval=600
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;expiration=3600
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;line=yes
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;endpoint=mytrunk
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;[mytrunk_auth]
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;type=auth
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;auth_type=userpass
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;password=1234567890
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;username=1234567890
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;realm=sip.example.com
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;===============ENDPOINT CONFIGURED AS A TRUNK, OUTBOUND AUTHENTICATION=======
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;
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; This is one way to configure an endpoint as a trunk. It is set up with
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; "outbound_auth=" to enable authentication when dialing out through this
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; endpoint. There is no inbound authentication set up since a provider will
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; not normally authenticate when calling you.
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;
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; The identify configuration enables IP address matching against this endpoint.
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; For calls from a trunking provider, the From user may be different every time,
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; so we want to match against IP address instead of From user.
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;
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; If you want the provider of your trunk to know where to send your calls
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; you'll need to use an outbound registration as in the example above this
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; section.
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;
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; NAT
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;
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; At a basic level configure the endpoint with a transport that is set up
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; with the appropriate NAT settings. There may be some additional settings you
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; need here based on your NAT/Firewall scenario. Look to the CLI config help
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; "config show help res_pjsip endpoint" or on the wiki for other NAT related
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; options and configuration. We've included a few below.
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;
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; AOR
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;
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; Endpoints use one or more AOR sections to store their contact details.
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; You can define multiple contact addresses in SIP URI format in multiple
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; "contact=" entries.
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;
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;[mytrunk]
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;type=endpoint
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;transport=transport-udp
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;context=from-external
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;disallow=all
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;allow=ulaw
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;outbound_auth=mytrunk_auth
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;aors=mytrunk
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; ;A few NAT relevant options that may come in handy.
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;force_rport=yes ;It's a good idea to read the configuration help for each
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;direct_media=no ;of these options.
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;ice_support=yes
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;[mytrunk]
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;type=aor
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;contact=sip:198.51.100.1:5060
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;contact=sip:198.51.100.2:5060
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;[mytrunk]
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;type=identify
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;endpoint=mytrunk
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;match=198.51.100.1
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;match=198.51.100.2
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;match=192.168.10.0:5061/24
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;=============ENDPOINT CONFIGURED AS A TRUNK, INBOUND AUTH AND REGISTRATION===
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;
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; Here we are allowing a remote device to register to Asterisk and requiring
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; that they authenticate for registration and calls.
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; You'll note that this configuration is essentially the same as configuring
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; an endpoint for use with a SIP phone.
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;[7000]
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;type=endpoint
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;context=from-external
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;disallow=all
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;allow=ulaw
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;transport=transport-udp
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;auth=7000
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;aors=7000
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;[7000]
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;type=auth
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;auth_type=userpass
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;password=7000
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;username=7000
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;[7000]
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;type=aor
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;max_contacts=1
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;===============ENDPOINT CONFIGURED FOR USE WITH A SIP PHONE==================
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;
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; This example includes the endpoint, auth and aor configurations. It
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; requires inbound authentication and allows registration, as well as references
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; a transport that you'll need to uncomment from the previous examples.
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;
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; Uncomment one of the transport lines to choose which transport you want. If
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; not specified then the default transport chosen is the first compatible transport
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; in the configuration file for the contact URL.
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;
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; Modify the "max_contacts=" line to change how many unique registrations to allow.
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;
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; Use the "contact=" line instead of max_contacts= if you want to statically
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; define the location of the device.
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;
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; If using the TLS enabled transport, you may want the "media_encryption=sdes"
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; option to additionally enable SRTP, though they are not mutually inclusive.
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;
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; If this endpoint were remote, and it was using a transport configured for NAT
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; then you likely want to use "direct_media=no" to prevent audio issues.
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;[6001]
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;type=endpoint
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;transport=transport-udp
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;context=from-internal
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;disallow=all
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;allow=ulaw
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;allow=gsm
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;auth=6001
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;aors=6001
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;
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; A few more transports to pick from, and some related options below them.
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;
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;transport=transport-tls
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;media_encryption=sdes
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;transport=transport-udp-ipv6
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;transport=transport-udp-nat
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;direct_media=no
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;
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; MWI related options
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;aggregate_mwi=yes
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;mailboxes=6001@default,7001@default
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;mwi_from_user=6001
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;
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; Extension and Device state options
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;
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;device_state_busy_at=1
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;allow_subscribe=yes
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;sub_min_expiry=30
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;
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; STIR/SHAKEN support.
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;
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;stir_shaken=no
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;[6001]
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;type=auth
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;auth_type=userpass
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;password=6001
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;username=6001
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;[6001]
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;type=aor
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;max_contacts=1
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;contact=sip:6001@192.0.2.1:5060
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;===============ENDPOINT BEHIND NAT OR FIREWALL===============================
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;
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; This example assumes your transport is configured with a public IP and the
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; endpoint itself is behind NAT and maybe a firewall, rather than having
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; Asterisk behind NAT. For the sake of simplicity, we'll assume a typical
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; VOIP phone. The most important settings to configure are:
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;
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; * direct_media, to ensure Asterisk stays in the media path
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; * rtp_symmetric and force_rport options to help the far-end NAT/firewall
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;
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; Depending on the settings of your remote SIP device or NAT/firewall device
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; you may have to experiment with a combination of these settings.
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;
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; If both Asterisk and the remote phones are a behind NAT/firewall then you'll
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; have to make sure to use a transport with appropriate settings (as in the
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; transport-udp-nat example).
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;
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;[6002]
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;type=endpoint
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;transport=transport-udp
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;context=from-internal
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;disallow=all
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;allow=ulaw
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;auth=6002
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;aors=6002
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;direct_media=no
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;rtp_symmetric=yes
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;force_rport=yes
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;rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port
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;ice_support=yes ;This is specific to clients that support NAT traversal
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;for media via ICE,STUN,TURN. See the wiki at:
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;https://wiki.asterisk.org/wiki/x/D4FHAQ
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;for a deeper explanation of this topic.
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;[6002]
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;type=auth
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;auth_type=userpass
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;password=6002
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;username=6002
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;[6002]
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;type=aor
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;max_contacts=2
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;============EXAMPLE ACL CONFIGURATION==========================================
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;
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; The ACL or Access Control List section defines a set of permissions to permit
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; or deny access to various address or addresses. Alternatively it references an
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; ACL configuration already set in acl.conf.
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;
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; The ACL configuration is independent of individual endpoint configuration and
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; operates on all inbound SIP communication using res_pjsip.
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; Reference an ACL defined in acl.conf.
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;
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;[acl]
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;type=acl
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;acl=example_named_acl1
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; Reference a contactacl specifically.
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;
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;[acl]
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;type=acl
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;contact_acl=example_contact_acl1
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; Define your own ACL here in pjsip.conf and
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; permit or deny by IP address or range.
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;
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;[acl]
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;type=acl
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;deny=0.0.0.0/0.0.0.0
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;permit=209.16.236.0/24
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;deny=209.16.236.1
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; Restrict based on Contact Headers rather than IP.
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; Define options multiple times for various addresses or use a comma-delimited string.
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;
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;[acl]
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;type=acl
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;contact_deny=0.0.0.0/0.0.0.0
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;contact_permit=209.16.236.0/24
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;contact_permit=209.16.236.1
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;contact_permit=209.16.236.2,209.16.236.3
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; Restrict based on Contact Headers rather than IP and use
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; advanced syntax. Note the bang symbol used for "NOT", so we can deny
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; 209.16.236.12/32 within the permit= statement.
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;
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;[acl]
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;type=acl
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;contact_deny=0.0.0.0/0.0.0.0
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;contact_permit=209.16.236.0
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;permit=209.16.236.0/24, !209.16.236.12/32
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;============EXAMPLE RLS CONFIGURATION==========================================
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;
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;Asterisk provides support for RFC 4662 Resource List Subscriptions. This allows
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;for an endpoint to, through a single subscription, subscribe to the states of
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;multiple resources. Resource lists are configured in pjsip.conf using the
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;resource_list configuration object. Below is an example of a resource list that
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;allows an endpoint to subscribe to the presence of alice, bob, and carol.
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;[my_list]
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;type=resource_list
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;list_item=alice
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;list_item=bob
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;list_item=carol
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;event=presence
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;The "event" option in the resource list corresponds to the SIP event-package
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;that the subscribed resources belong to. A resource list can only provide states
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;for resources that belong to the same event-package. This means that you cannot
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;create a list that is a combination of presence and message-summary resources,
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;for instance. Any event-package that Asterisk supports can be used in a resource
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;list (presence, dialog, and message-summary). Whenever support for a new event-
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;package is added to Asterisk, support for that event-package in resource lists
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;will automatically be supported.
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;The "list_item" options indicate the names of resources to subscribe to. The
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;way these are interpreted is event-package specific. For instance, with presence
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;list_items, hints in the dialplan are looked up. With message-summary list_items,
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;mailboxes are looked up using your installed voicemail provider (app_voicemail
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;by default). Note that in the above example, the list_item options were given
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;one per line. However, it is also permissible to provide multiple list_item
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;options on a single line (e.g. list_item = alice,bob,carol).
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;In addition to the options presented in the above configuration, there are two
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;more configuration options that can be set.
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; * full_state: dictates whether Asterisk should always send the states of
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; all resources in the list at once. Defaults to "no". You should only set
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; this to "yes" if you are interoperating with an endpoint that does not
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; behave correctly when partial state notifications are sent to it.
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; * notification_batch_interval: By default, Asterisk will send a NOTIFY request
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; immediately when a resource changes state. This option causes Asterisk to
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; start batching resource state changes for the specified number of milliseconds
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; after a resource changes states. This way, if multiple resources change state
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; within a brief interval, Asterisk can send a single NOTIFY request with all
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; of the state changes reflected in it.
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;There is a limitation to the size of resource lists in Asterisk. If a constructed
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;notification from Asterisk will exceed 64000 bytes, then the message is deemed
|
|
;too large to send. If you find that you are seeing error messages about SIP
|
|
;NOTIFY requests being too large to send, consider breaking your lists into
|
|
;sub-lists.
|
|
|
|
;============EXAMPLE PHONEPROV CONFIGURATION================================
|
|
|
|
; Before configuring provisioning here, see the documentation for res_phoneprov
|
|
; and configure phoneprov.conf appropriately.
|
|
|
|
; For each user to be autoprovisioned, a [phoneprov] configuration section
|
|
; must be created. At a minimum, the 'type', 'PROFILE' and 'MAC' variables must
|
|
; be set. All other variables are optional.
|
|
; Example:
|
|
|
|
;[1000]
|
|
;type=phoneprov ; must be specified as 'phoneprov'
|
|
;endpoint=1000 ; Required only if automatic setting of
|
|
; USERNAME, SECRET, DISPLAY_NAME and CALLERID
|
|
; are needed.
|
|
;PROFILE=digium ; required
|
|
;MAC=deadbeef4dad ; required
|
|
;SERVER=myserver.example.com ; A standard variable
|
|
;TIMEZONE=America/Denver ; A standard variable
|
|
;MYVAR=somevalue ; A user confdigured variable
|
|
|
|
; If the phoneprov sections have common variables, it is best to create a
|
|
; phoneprov template. The example below will produce the same configuration
|
|
; as the one specified above except that MYVAR will be overridden for
|
|
; the specific user.
|
|
; Example:
|
|
|
|
;[phoneprov_defaults](!)
|
|
;type=phoneprov ; must be specified as 'phoneprov'
|
|
;PROFILE=digium ; required
|
|
;SERVER=myserver.example.com ; A standard variable
|
|
;TIMEZONE=America/Denver ; A standard variable
|
|
;MYVAR=somevalue ; A user configured variable
|
|
|
|
;[1000](phoneprov_defaults)
|
|
;endpoint=1000 ; Required only if automatic setting of
|
|
; USERNAME, SECRET, DISPLAY_NAME and CALLERID
|
|
; are needed.
|
|
;MAC=deadbeef4dad ; required
|
|
;MYVAR=someOTHERvalue ; A user confdigured variable
|
|
|
|
; To have USERNAME and SECRET automatically set, the endpoint
|
|
; specified here must in turn have an outbound_auth section defined.
|
|
|
|
; Fuller example:
|
|
|
|
;[1000]
|
|
;type=endpoint
|
|
;outbound_auth=1000-auth
|
|
;callerid=My Name <8005551212>
|
|
;transport=transport-udp-nat
|
|
|
|
;[1000-auth]
|
|
;type=auth
|
|
;auth_type=userpass
|
|
;username=myname
|
|
;password=mysecret
|
|
|
|
;[phoneprov_defaults](!)
|
|
;type=phoneprov ; must be specified as 'phoneprov'
|
|
;PROFILE=someprofile ; required
|
|
;SERVER=myserver.example.com ; A standard variable
|
|
;TIMEZONE=America/Denver ; A standard variable
|
|
;MYVAR=somevalue ; A user configured variable
|
|
|
|
;[1000](phoneprov_defaults)
|
|
;endpoint=1000 ; Required only if automatic setting of
|
|
; USERNAME, SECRET, DISPLAY_NAME and CALLERID
|
|
; are needed.
|
|
;MAC=deadbeef4dad ; required
|
|
;MYVAR=someUSERvalue ; A user confdigured variable
|
|
;LABEL=1000 ; A standard variable
|
|
|
|
; The previous sections would produce a template substitution map as follows:
|
|
|
|
;MAC=deadbeef4dad ;added by pp1000
|
|
;USERNAME=myname ;automatically added by 1000-auth username
|
|
;SECRET=mysecret ;automatically added by 1000-auth password
|
|
;PROFILE=someprofile ;added by defaults
|
|
;SERVER=myserver.example.com ;added by defaults
|
|
;SERVER_PORT=5060 ;added by defaults
|
|
;MYVAR=someUSERvalue ;added by defaults but overdidden by user
|
|
;CALLERID=8005551212 ;automatically added by 1000 callerid
|
|
;DISPLAY_NAME=My Name ;automatically added by 1000 callerid
|
|
;TIMEZONE=America/Denver ;added by defaults
|
|
;TZOFFSET=252100 ;automatically calculated by res_phoneprov
|
|
;DST_ENABLE=1 ;automatically calculated by res_phoneprov
|
|
;DST_START_MONTH=3 ;automatically calculated by res_phoneprov
|
|
;DST_START_MDAY=9 ;automatically calculated by res_phoneprov
|
|
;DST_START_HOUR=3 ;automatically calculated by res_phoneprov
|
|
;DST_END_MONTH=11 ;automatically calculated by res_phoneprov
|
|
;DST_END_MDAY=2 ;automatically calculated by res_phoneprov
|
|
;DST_END_HOUR=1 ;automatically calculated by res_phoneprov
|
|
;ENDPOINT_ID=1000 ;automatically added by this module
|
|
;AUTH_ID=1000-auth ;automatically added by this module
|
|
;TRANSPORT_ID=transport-udp-nat ;automatically added by this module
|
|
;LABEL=1000 ;added by user
|
|
|
|
; MODULE PROVIDING BELOW SECTION(S): res_pjsip
|
|
;==========================ENDPOINT SECTION OPTIONS=========================
|
|
;[endpoint]
|
|
; SYNOPSIS: Endpoint
|
|
;100rel=yes ; Allow support for RFC3262 provisional ACK tags (default:
|
|
; "yes")
|
|
;aggregate_mwi=yes ; (default: "yes")
|
|
;allow= ; Media Codec s to allow (default: "")
|
|
;allow_overlap=yes ; Enable RFC3578 overlap dialing support. (default: "yes")
|
|
;aors= ; AoR s to be used with the endpoint (default: "")
|
|
;auth= ; Authentication Object s associated with the endpoint (default: "")
|
|
;callerid= ; CallerID information for the endpoint (default: "")
|
|
;callerid_privacy=allowed_not_screened ; Default privacy level (default: "allowed_not_screened")
|
|
;callerid_tag= ; Internal id_tag for the endpoint (default: "")
|
|
;context=default ; Dialplan context for inbound sessions (default:
|
|
; "default")
|
|
;direct_media_glare_mitigation=none ; Mitigation of direct media re INVITE
|
|
; glare (default: "none")
|
|
;direct_media_method=invite ; Direct Media method type (default: "invite")
|
|
;trust_connected_line=yes ; Accept Connected Line updates from this endpoint
|
|
; (default: "yes")
|
|
;send_connected_line=yes ; Send Connected Line updates to this endpoint
|
|
; (default: "yes")
|
|
;connected_line_method=invite ; Connected line method type.
|
|
; When set to "invite", check the remote's
|
|
; Allow header and if UPDATE is allowed, send
|
|
; UPDATE instead of INVITE to avoid SDP
|
|
; renegotiation. If UPDATE is not Allowed,
|
|
; send INVITE.
|
|
; If set to "update", send UPDATE regardless
|
|
; of what the remote Allows.
|
|
; (default: "invite")
|
|
;direct_media=yes ; Determines whether media may flow directly between
|
|
; endpoints (default: "yes")
|
|
;disable_direct_media_on_nat=no ; Disable direct media session refreshes when
|
|
; NAT obstructs the media session (default:
|
|
; "no")
|
|
;disallow= ; Media Codec s to disallow (default: "")
|
|
;dtmf_mode=rfc4733 ; DTMF mode (default: "rfc4733")
|
|
;media_address= ; IP address used in SDP for media handling (default: "")
|
|
;bind_rtp_to_media_address= ; Bind the RTP session to the media_address.
|
|
; This causes all RTP packets to be sent from
|
|
; the specified address. (default: "no")
|
|
;force_rport=yes ; Force use of return port (default: "yes")
|
|
;ice_support=no ; Enable the ICE mechanism to help traverse NAT (default: "no")
|
|
;identify_by=username ; A comma-separated list of ways the Endpoint or AoR can be
|
|
; identified.
|
|
; "username": Identify by the From or To username and domain
|
|
; "auth_username": Identify by the Authorization username and realm
|
|
; "ip": Identify by the source IP address
|
|
; "header": Identify by a configured SIP header value.
|
|
; In the username and auth_username cases, if an exact match
|
|
; on both username and domain/realm fails, the match is
|
|
; retried with just the username.
|
|
; (default: "username,ip")
|
|
;redirect_method=user ; How redirects received from an endpoint are handled
|
|
; (default: "user")
|
|
;mailboxes= ; NOTIFY the endpoint when state changes for any of the specified mailboxes.
|
|
; Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state
|
|
; changes happen for any of the specified mailboxes. (default: "")
|
|
;voicemail_extension= ; The voicemail extension to send in the NOTIFY Message-Account header
|
|
; (default: global/default_voicemail_extension)
|
|
;mwi_subscribe_replaces_unsolicited=no
|
|
; An MWI subscribe will replace unsoliticed NOTIFYs
|
|
; (default: "no")
|
|
;moh_suggest=default ; Default Music On Hold class (default: "default")
|
|
;moh_passthrough=yes ; Pass Music On Hold through using SIP re-invites with sendonly
|
|
; when placing on hold and sendrecv when taking off hold
|
|
;outbound_auth= ; Authentication object used for outbound requests (default:
|
|
; "")
|
|
;outbound_proxy= ; Proxy through which to send requests, a full SIP URI
|
|
; must be provided (default: "")
|
|
;rewrite_contact=no ; Allow Contact header to be rewritten with the source
|
|
; IP address port (default: "no")
|
|
;rtp_symmetric=no ; Enforce that RTP must be symmetric (default: "no")
|
|
;send_diversion=yes ; Send the Diversion header conveying the diversion
|
|
; information to the called user agent (default: "yes")
|
|
;send_pai=no ; Send the P Asserted Identity header (default: "no")
|
|
;send_rpid=no ; Send the Remote Party ID header (default: "no")
|
|
;rpid_immediate=no ; Send connected line updates on unanswered incoming calls immediately. (default: "no")
|
|
;timers_min_se=90 ; Minimum session timers expiration period (default:
|
|
; "90")
|
|
;timers=yes ; Session timers for SIP packets (default: "yes")
|
|
;timers_sess_expires=1800 ; Maximum session timer expiration period
|
|
; (default: "1800")
|
|
;transport= ; Explicit transport configuration to use (default: "")
|
|
; This will force the endpoint to use the specified transport
|
|
; configuration to send SIP messages. You need to already know
|
|
; what kind of transport (UDP/TCP/IPv4/etc) the endpoint device
|
|
; will use.
|
|
|
|
;trust_id_inbound=no ; Accept identification information received from this
|
|
; endpoint (default: "no")
|
|
;trust_id_outbound=no ; Send private identification details to the endpoint
|
|
; (default: "no")
|
|
;type= ; Must be of type endpoint (default: "")
|
|
;use_ptime=no ; Use Endpoint s requested packetisation interval (default:
|
|
; "no")
|
|
;use_avpf=no ; Determines whether res_pjsip will use and enforce usage of
|
|
; AVPF for this endpoint (default: "no")
|
|
;media_encryption=no ; Determines whether res_pjsip will use and enforce
|
|
; usage of media encryption for this endpoint (default:
|
|
; "no")
|
|
;media_encryption_optimistic=no ; Use encryption if possible but don't fail the call
|
|
; if not possible.
|
|
;g726_non_standard=no ; When set to "yes" and an endpoint negotiates g.726
|
|
; audio then g.726 for AAL2 packing order is used contrary
|
|
; to what is recommended in RFC3551. Note, 'g726aal2' also
|
|
; needs to be specified in the codec allow list
|
|
; (default: "no")
|
|
;inband_progress=no ; Determines whether chan_pjsip will indicate ringing
|
|
; using inband progress (default: "no")
|
|
;call_group= ; The numeric pickup groups for a channel (default: "")
|
|
;pickup_group= ; The numeric pickup groups that a channel can pickup (default:
|
|
; "")
|
|
;named_call_group= ; The named pickup groups for a channel (default: "")
|
|
;named_pickup_group= ; The named pickup groups that a channel can pickup
|
|
; (default: "")
|
|
;device_state_busy_at=0 ; The number of in use channels which will cause busy
|
|
; to be returned as device state (default: "0")
|
|
;t38_udptl=no ; Whether T 38 UDPTL support is enabled or not (default: "no")
|
|
;t38_udptl_ec=none ; T 38 UDPTL error correction method (default: "none")
|
|
;t38_udptl_maxdatagram=0 ; T 38 UDPTL maximum datagram size (default:
|
|
; "0")
|
|
;fax_detect=no ; Whether CNG tone detection is enabled (default: "no")
|
|
;fax_detect_timeout=30 ; How many seconds into a call before fax_detect is
|
|
; disabled for the call.
|
|
; Zero disables the timeout.
|
|
; (default: "0")
|
|
;t38_udptl_nat=no ; Whether NAT support is enabled on UDPTL sessions
|
|
; (default: "no")
|
|
;tone_zone= ; Set which country s indications to use for channels created
|
|
; for this endpoint (default: "")
|
|
;language= ; Set the default language to use for channels created for this
|
|
; endpoint (default: "")
|
|
;one_touch_recording=no ; Determines whether one touch recording is allowed for
|
|
; this endpoint (default: "no")
|
|
;record_on_feature=automixmon ; The feature to enact when one touch recording
|
|
; is turned on (default: "automixmon")
|
|
;record_off_feature=automixmon ; The feature to enact when one touch recording
|
|
; is turned off (default: "automixmon")
|
|
;rtp_engine=asterisk ; Name of the RTP engine to use for channels created
|
|
; for this endpoint (default: "asterisk")
|
|
;allow_transfer=yes ; Determines whether SIP REFER transfers are allowed
|
|
; for this endpoint (default: "yes")
|
|
;sdp_owner=- ; String placed as the username portion of an SDP origin o line
|
|
; (default: "-")
|
|
;sdp_session=Asterisk ; String used for the SDP session s line (default:
|
|
; "Asterisk")
|
|
;tos_audio=0 ; DSCP TOS bits for audio streams (default: "0")
|
|
;tos_video=0 ; DSCP TOS bits for video streams (default: "0")
|
|
;cos_audio=0 ; Priority for audio streams (default: "0")
|
|
;cos_video=0 ; Priority for video streams (default: "0")
|
|
;allow_subscribe=yes ; Determines if endpoint is allowed to initiate
|
|
; subscriptions with Asterisk (default: "yes")
|
|
;sub_min_expiry=0 ; The minimum allowed expiry time for subscriptions
|
|
; initiated by the endpoint (default: "0")
|
|
;from_user= ; Username to use in From header for requests to this endpoint
|
|
; (default: "")
|
|
;mwi_from_user= ; Username to use in From header for unsolicited MWI NOTIFYs to
|
|
; this endpoint (default: "")
|
|
;from_domain= ; Domain to user in From header for requests to this endpoint
|
|
; (default: "")
|
|
;dtls_verify=no ; Verify that the provided peer certificate is valid (default:
|
|
; "no")
|
|
;dtls_rekey=0 ; Interval at which to renegotiate the TLS session and rekey
|
|
; the SRTP session (default: "0")
|
|
;dtls_auto_generate_cert= ; Enable ephemeral DTLS certificate generation (default:
|
|
; "no")
|
|
;dtls_cert_file= ; Path to certificate file to present to peer (default:
|
|
; "")
|
|
;dtls_private_key= ; Path to private key for certificate file (default:
|
|
; "")
|
|
;dtls_cipher= ; Cipher to use for DTLS negotiation (default: "")
|
|
;dtls_ca_file= ; Path to certificate authority certificate (default: "")
|
|
;dtls_ca_path= ; Path to a directory containing certificate authority
|
|
; certificates (default: "")
|
|
;dtls_setup= ; Whether we are willing to accept connections connect to the
|
|
; other party or both (default: "")
|
|
;dtls_fingerprint= ; Hash to use for the fingerprint placed into SDP
|
|
; (default: "SHA-256")
|
|
;srtp_tag_32=no ; Determines whether 32 byte tags should be used instead of 80
|
|
; byte tags (default: "no")
|
|
;set_var= ; Variable set on a channel involving the endpoint. For multiple
|
|
; channel variables specify multiple 'set_var'(s)
|
|
;rtp_keepalive= ; Interval, in seconds, between comfort noise RTP packets if
|
|
; RTP is not flowing. This setting is useful for ensuring that
|
|
; holes in NATs and firewalls are kept open throughout a call.
|
|
;rtp_timeout= ; Hang up channel if RTP is not received for the specified
|
|
; number of seconds when the channel is off hold (default:
|
|
; "0" or not enabled)
|
|
;rtp_timeout_hold= ; Hang up channel if RTP is not received for the specified
|
|
; number of seconds when the channel is on hold (default:
|
|
; "0" or not enabled)
|
|
;contact_user= ; On outgoing requests, force the user portion of the Contact
|
|
; header to this value (default: "")
|
|
;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
|
|
; rather than advertising all joint codec capabilities. This
|
|
; limits the other side's codec choice to exactly what we prefer.
|
|
; default is no.
|
|
;asymmetric_rtp_codec= ; Allow the sending and receiving codec to differ and
|
|
; not be automatically matched (default: "no")
|
|
;refer_blind_progress= ; Whether to notifies all the progress details on blind
|
|
; transfer (default: "yes"). The value "no" is useful
|
|
; for some SIP phones (Mitel/Aastra, Snom) which expect
|
|
; a sip/frag "200 OK" after REFER has been accepted.
|
|
;notify_early_inuse_ringing = ; Whether to notifies dialog-info 'early'
|
|
; on INUSE && RINGING state (default: "no").
|
|
; The value "yes" is useful for some SIP phones
|
|
; (Cisco SPA) to be able to indicate and pick up
|
|
; ringing devices.
|
|
;max_audio_streams= ; The maximum number of allowed negotiated audio streams
|
|
; (default: 1)
|
|
;max_video_streams= ; The maximum number of allowed negotiated video streams
|
|
; (default: 1)
|
|
;webrtc= ; When set to "yes" this also enables the following values that are needed
|
|
; for webrtc: rtcp_mux, use_avpf, ice_support, and use_received_transport.
|
|
; The following configuration settings also get defaulted as follows:
|
|
; media_encryption=dtls
|
|
; dtls_verify=fingerprint
|
|
; dtls_setup=actpass
|
|
; A dtls_cert_file and a dtls_ca_file still need to be specified.
|
|
; Default for this option is "no"
|
|
;incoming_mwi_mailbox = ; Mailbox name to use when incoming MWI NOTIFYs are
|
|
; received.
|
|
; If an MWI NOTIFY is received FROM this endpoint,
|
|
; this mailbox will be used when notifying other modules
|
|
; of MWI status changes. If not set, incoming MWI
|
|
; NOTIFYs are ignored.
|
|
;follow_early_media_fork = ; On outgoing calls, if the UAS responds with
|
|
; different SDP attributes on subsequent 18X or 2XX
|
|
; responses (such as a port update) AND the To tag
|
|
; on the subsequent response is different than that
|
|
; on the previous one, follow it. This usually
|
|
; happens when the INVITE is forked to multiple UASs
|
|
; and more than 1 sends an SDP answer.
|
|
; This option must also be enabled in the system
|
|
; section.
|
|
; (default: yes)
|
|
;accept_multiple_sdp_answers =
|
|
; On outgoing calls, if the UAS responds with
|
|
; different SDP attributes on non-100rel 18X or 2XX
|
|
; responses (such as a port update) AND the To tag on
|
|
; the subsequent response is the same as that on the
|
|
; previous one, process it. This can happen when the
|
|
; UAS needs to change ports for some reason such as
|
|
; using a separate port for custom ringback.
|
|
; This option must also be enabled in the system
|
|
; section.
|
|
; (default: no)
|
|
;suppress_q850_reason_headers =
|
|
; Suppress Q.850 Reason headers for this endpoint.
|
|
; Some devices can't accept multiple Reason headers
|
|
; and get confused when both 'SIP' and 'Q.850' Reason
|
|
; headers are received. This option allows the
|
|
; 'Q.850' Reason header to be suppressed.
|
|
; (default: no)
|
|
;ignore_183_without_sdp =
|
|
; Do not forward 183 when it doesn't contain SDP.
|
|
; Certain SS7 internetworking scenarios can result in
|
|
; a 183 to be generated for reasons other than early
|
|
; media. Forwarding this 183 can cause loss of
|
|
; ringback tone. This flag emulates the behavior of
|
|
; chan_sip and prevents these 183 responses from
|
|
; being forwarded.
|
|
; (default: no)
|
|
;stir_shaken =
|
|
; If this is enabled, STIR/SHAKEN operations will be
|
|
; performed on this endpoint. This includes inbound
|
|
; and outbound INVITEs. On an inbound INVITE, Asterisk
|
|
; will check for an Identity header and attempt to
|
|
; verify the call. On an outbound INVITE, Asterisk will
|
|
; add an Identity header that others can use to verify
|
|
; calls from this endpoint. Additional configuration is
|
|
; done in stir_shaken.conf.
|
|
; The STIR_SHAKEN dialplan function must be used to get
|
|
; the verification results on inbound INVITEs. Nothing
|
|
; happens to the call if verification fails; it's up to
|
|
; you to determine what to do with the results.
|
|
; (default: no)
|
|
|
|
;==========================AUTH SECTION OPTIONS=========================
|
|
;[auth]
|
|
; SYNOPSIS: Authentication type
|
|
;
|
|
; Note: Using the same auth section for inbound and outbound
|
|
; authentication is not recommended. There is a difference in
|
|
; meaning for an empty realm setting between inbound and outbound
|
|
; authentication uses. Look to the CLI config help
|
|
; "config show help res_pjsip auth realm" or on the wiki for the
|
|
; difference.
|
|
;
|
|
;auth_type=userpass ; Authentication type (default: "userpass")
|
|
;nonce_lifetime=32 ; Lifetime of a nonce associated with this
|
|
; authentication config (default: "32")
|
|
;md5_cred= ; MD5 Hash used for authentication (default: "")
|
|
;password= ; PlainText password used for authentication (default: "")
|
|
;realm= ; SIP realm for endpoint (default: "")
|
|
;type= ; Must be auth (default: "")
|
|
;username= ; Username to use for account (default: "")
|
|
|
|
|
|
;==========================DOMAIN_ALIAS SECTION OPTIONS=========================
|
|
;[domain_alias]
|
|
; SYNOPSIS: Domain Alias
|
|
;type= ; Must be of type domain_alias (default: "")
|
|
;domain= ; Domain to be aliased (default: "")
|
|
|
|
|
|
;==========================TRANSPORT SECTION OPTIONS=========================
|
|
;[transport]
|
|
; SYNOPSIS: SIP Transport
|
|
;
|
|
;async_operations=1 ; Number of simultaneous Asynchronous Operations
|
|
; (default: "1")
|
|
;bind= ; IP Address and optional port to bind to for this transport (default:
|
|
; "")
|
|
; Note that for the Websocket transport the TLS configuration is configured
|
|
; in http.conf and is applied for all HTTPS traffic.
|
|
;ca_list_file= ; File containing a list of certificates to read TLS ONLY
|
|
; (default: "")
|
|
;ca_list_path= ; Path to directory containing certificates to read TLS ONLY.
|
|
; PJProject version 2.4 or higher is required for this option to
|
|
; be used.
|
|
; (default: "")
|
|
;cert_file= ; Certificate file for endpoint TLS ONLY
|
|
; Will read .crt or .pem file but only uses cert,
|
|
; a .key file must be specified via priv_key_file.
|
|
; Since PJProject version 2.5: If the file name ends in _rsa,
|
|
; for example "asterisk_rsa.pem", the files "asterisk_dsa.pem"
|
|
; and/or "asterisk_ecc.pem" are loaded (certificate, inter-
|
|
; mediates, private key), to support multiple algorithms for
|
|
; server authentication (RSA, DSA, ECDSA). If the chains are
|
|
; different, at least OpenSSL 1.0.2 is required.
|
|
; (default: "")
|
|
;cipher= ; Preferred cryptography cipher names TLS ONLY (default: "")
|
|
;method= ; Method of SSL transport TLS ONLY (default: "")
|
|
;priv_key_file= ; Private key file TLS ONLY (default: "")
|
|
;verify_client= ; Require verification of client certificate TLS ONLY (default:
|
|
; "")
|
|
;verify_server= ; Require verification of server certificate TLS ONLY (default:
|
|
; "")
|
|
;require_client_cert= ; Require client certificate TLS ONLY (default: "")
|
|
;domain= ; Domain the transport comes from (default: "")
|
|
;external_media_address= ; External IP address to use in RTP handling
|
|
; (default: "")
|
|
;external_signaling_address= ; External address for SIP signalling (default:
|
|
; "")
|
|
;external_signaling_port=0 ; External port for SIP signalling (default:
|
|
; "0")
|
|
;local_net= ; Network to consider local used for NAT purposes (default: "")
|
|
;password= ; Password required for transport (default: "")
|
|
;protocol=udp ; Protocol to use for SIP traffic (default: "udp")
|
|
;type= ; Must be of type transport (default: "")
|
|
;tos=0 ; Enable TOS for the signalling sent over this transport (default: "0")
|
|
;cos=0 ; Enable COS for the signalling sent over this transport (default: "0")
|
|
;websocket_write_timeout=100 ; Default write timeout to set on websocket
|
|
; transports. This value may need to be adjusted
|
|
; for connections where Asterisk must write a
|
|
; substantial amount of data and the receiving
|
|
; clients are slow to process the received
|
|
; information. Value is in milliseconds; default
|
|
; is 100 ms.
|
|
;allow_reload=no ; Although transports can now be reloaded, that may not be
|
|
; desirable because of the slight possibility of dropped
|
|
; calls. To make sure there are no unintentional drops, if
|
|
; this option is set to 'no' (the default) changes to the
|
|
; particular transport will be ignored. If set to 'yes',
|
|
; changes (if any) will be applied.
|
|
;symmetric_transport=no ; When a request from a dynamic contact comes in on a
|
|
; transport with this option set to 'yes', the transport
|
|
; name will be saved and used for subsequent outgoing
|
|
; requests like OPTIONS, NOTIFY and INVITE. It's saved
|
|
; as a contact uri parameter named 'x-ast-txp' and will
|
|
; display with the contact uri in CLI, AMI, and ARI
|
|
; output. On the outgoing request, if a transport
|
|
; wasn't explicitly set on the endpoint AND the request
|
|
; URI is not a hostname, the saved transport will be
|
|
; used and the 'x-ast-txp' parameter stripped from the
|
|
; outgoing packet.
|
|
|
|
;==========================AOR SECTION OPTIONS=========================
|
|
;[aor]
|
|
; SYNOPSIS: The configuration for a location of an endpoint
|
|
;contact= ; Permanent contacts assigned to AoR (default: "")
|
|
;default_expiration=3600 ; Default expiration time in seconds for
|
|
; contacts that are dynamically bound to an AoR
|
|
; (default: "3600")
|
|
;mailboxes= ; Allow subscriptions for the specified mailbox(es)
|
|
; This option applies when an external entity subscribes to an AoR
|
|
; for Message Waiting Indications. (default: "")
|
|
;voicemail_extension= ; The voicemail extension to send in the NOTIFY Message-Account header
|
|
; (default: global/default_voicemail_extension)
|
|
;maximum_expiration=7200 ; Maximum time to keep an AoR (default: "7200")
|
|
;max_contacts=0 ; Maximum number of contacts that can bind to an AoR (default:
|
|
; "0")
|
|
;minimum_expiration=60 ; Minimum keep alive time for an AoR (default: "60")
|
|
;remove_existing=no ; Allow a registration to succeed by displacing any existing
|
|
; contacts that now exceed the max_contacts count. Any
|
|
; removed contacts are the next to expire. The behaviour is
|
|
; beneficial when rewrite_contact is enabled and max_contacts
|
|
; is greater than one. The removed contact is likely the old
|
|
; contact created by rewrite_contact that the device is
|
|
; refreshing.
|
|
; (default: "no")
|
|
;type= ; Must be of type aor (default: "")
|
|
;qualify_frequency=0 ; Interval at which to qualify an AoR via OPTIONS requests.
|
|
; (default: "0")
|
|
;qualify_timeout=3.0 ; Qualify timeout in fractional seconds (default: "3.0")
|
|
;authenticate_qualify=no ; Authenticates a qualify request if needed
|
|
; (default: "no")
|
|
;outbound_proxy= ; Proxy through which to send OPTIONS requests, a full SIP URI
|
|
; must be provided (default: "")
|
|
|
|
|
|
;==========================SYSTEM SECTION OPTIONS=========================
|
|
[system]
|
|
type=system
|
|
; SYNOPSIS: Options that apply to the SIP stack as well as other system-wide settings
|
|
;timer_t1=500 ; Set transaction timer T1 value milliseconds (default: "500")
|
|
;timer_b=32000 ; Set transaction timer B value milliseconds (default: "32000")
|
|
;compact_headers=no ; Use the short forms of common SIP header names
|
|
; (default: "no")
|
|
;threadpool_initial_size=0 ; Initial number of threads in the res_pjsip
|
|
; threadpool (default: "0")
|
|
;threadpool_auto_increment=5 ; The amount by which the number of threads is
|
|
; incremented when necessary (default: "5")
|
|
;threadpool_idle_timeout=60 ; Number of seconds before an idle thread
|
|
; should be disposed of (default: "60")
|
|
;threadpool_max_size=0 ; Maximum number of threads in the res_pjsip threadpool
|
|
; A value of 0 indicates no maximum (default: "0")
|
|
;disable_tcp_switch=yes ; Disable automatic switching from UDP to TCP transports
|
|
; if outgoing request is too large.
|
|
; See RFC 3261 section 18.1.1.
|
|
; Disabling this option has been known to cause interoperability
|
|
; issues, so disable at your own risk.
|
|
; (default: "yes")
|
|
;follow_early_media_fork = ; On outgoing calls, if the UAS responds with
|
|
; different SDP attributes on subsequent 18X or 2XX
|
|
; responses (such as a port update) AND the To tag
|
|
; on the subsequent response is different than that
|
|
; on the previous one, follow it. This usually
|
|
; happens when the INVITE is forked to multiple UASs
|
|
; and more than 1 sends an SDP answer.
|
|
; This option must also be enabled on endpoints that
|
|
; require this functionality.
|
|
; (default: yes)
|
|
;accept_multiple_sdp_answers =
|
|
; On outgoing calls, if the UAS responds with
|
|
; different SDP attributes on non-100rel 18X or 2XX
|
|
; responses (such as a port update) AND the To tag on
|
|
; the subsequent response is the same as that on the
|
|
; previous one, process it. This can happen when the
|
|
; UAS needs to change ports for some reason such as
|
|
; using a separate port for custom ringback.
|
|
; This option must also be enabled on endpoints that
|
|
; require this functionality.
|
|
; (default: no)
|
|
;disable_rport=no ; Disable the use of "rport" in outgoing requests.
|
|
;type= ; Must be of type system (default: "")
|
|
|
|
;==========================GLOBAL SECTION OPTIONS=========================
|
|
;[global]
|
|
; SYNOPSIS: Options that apply globally to all SIP communications
|
|
;max_forwards=70 ; Value used in Max Forwards header for SIP requests
|
|
; (default: "70")
|
|
;type= ; Must be of type global (default: "")
|
|
;user_agent=Asterisk PBX ; Allows you to change the user agent string
|
|
; The default user agent string also contains
|
|
; the Asterisk version. If you don't want to
|
|
; expose this, change the user_agent string.
|
|
;default_outbound_endpoint=default_outbound_endpoint ; Endpoint to use when
|
|
; sending an outbound
|
|
; request to a URI
|
|
; without a specified
|
|
; endpoint (default: "d
|
|
; efault_outbound_endpo
|
|
; int")
|
|
;debug=no ; Enable/Disable SIP debug logging. Valid options include yes|no
|
|
; or a host address (default: "no")
|
|
;keep_alive_interval=90 ; The interval (in seconds) at which to send (double CRLF)
|
|
; keep-alives on all active connection-oriented transports;
|
|
; for connection-less like UDP see qualify_frequency.
|
|
; (default: "90")
|
|
;contact_expiration_check_interval=30
|
|
; The interval (in seconds) to check for expired contacts.
|
|
;disable_multi_domain=no
|
|
; Disable Multi Domain support.
|
|
; If disabled it can improve realtime performace by reducing
|
|
; number of database requsts
|
|
; (default: "no")
|
|
;endpoint_identifier_order=ip,username,anonymous
|
|
; The order by which endpoint identifiers are given priority.
|
|
; Currently, "ip", "header", "username", "auth_username" and "anonymous"
|
|
; are valid identifiers as registered by the res_pjsip_endpoint_identifier_*
|
|
; modules. Some modules like res_pjsip_endpoint_identifier_user register
|
|
; more than one identifier. Use the CLI command "pjsip show identifiers"
|
|
; to see the identifiers currently available.
|
|
; (default: ip,username,anonymous)
|
|
;max_initial_qualify_time=4 ; The maximum amount of time (in seconds) from
|
|
; startup that qualifies should be attempted on all
|
|
; contacts. If greater than the qualify_frequency
|
|
; for an aor, qualify_frequency will be used instead.
|
|
;regcontext=sipregistrations ; If regcontext is specified, Asterisk will dynamically
|
|
; create and destroy a NoOp priority 1 extension for a
|
|
; given endpoint who registers or unregisters with us.
|
|
; The extension added is the name of the endpoint.
|
|
;default_voicemail_extension=asterisk
|
|
; The voicemail extension to send in the NOTIFY Message-Account header
|
|
; if not set on endpoint or aor.
|
|
; (default: "")
|
|
;
|
|
; The following unidentified_request options are only used when "auth_username"
|
|
; matching is enabled in "endpoint_identifier_order".
|
|
;
|
|
;unidentified_request_count=5 ; The number of unidentified requests that can be
|
|
; received from a single IP address in
|
|
; unidentified_request_period seconds before a security
|
|
; event is generated. (default: 5)
|
|
;unidentified_request_period=5 ; See above. (default: 5 seconds)
|
|
;unidentified_request_prune_interval=30
|
|
; The interval at which unidentified requests
|
|
; are check to see if they can be pruned. If they're
|
|
; older than twice the unidentified_request_period,
|
|
; they're pruned.
|
|
;
|
|
;default_from_user=asterisk ; When Asterisk generates an outgoing SIP request, the
|
|
; From header username will be set to this value if
|
|
; there is no better option (such as CallerID or
|
|
; endpoint/from_user) to be used
|
|
;default_realm=asterisk ; When Asterisk generates a challenge, the digest realm
|
|
; will be set to this value if there is no better option
|
|
; (such as auth/realm) to be used.
|
|
|
|
; Asterisk Task Processor Queue Size
|
|
; On heavy loaded system with DB storage you may need to increase
|
|
; taskprocessor queue.
|
|
; If the taskprocessor queue size reached high water level,
|
|
; the alert is triggered.
|
|
; If the alert is set the pjsip distibutor stops processing incoming
|
|
; requests until the alert is cleared.
|
|
; The alert is cleared when taskprocessor queue size drops to the
|
|
; low water clear level.
|
|
; The next options set taskprocessor queue levels for MWI.
|
|
;mwi_tps_queue_high=500 ; Taskprocessor high water alert trigger level.
|
|
;mwi_tps_queue_low=450 ; Taskprocessor low water clear alert level.
|
|
; The default is -1 for 90% of high water level.
|
|
|
|
; Unsolicited MWI
|
|
; If there are endpoints configured with unsolicited MWI
|
|
; then res_pjsip_mwi module tries to send MWI to all endpoints on startup.
|
|
;mwi_disable_initial_unsolicited=no ; Disable sending unsolicited mwi to all endpoints on startup.
|
|
; If disabled then unsolicited mwi will start processing
|
|
; on the endpoint's next contact update.
|
|
|
|
;ignore_uri_user_options=no ; Enable/Disable ignoring SIP URI user field options.
|
|
; If you have this option enabled and there are semicolons
|
|
; in the user field of a SIP URI then the field is truncated
|
|
; at the first semicolon. This effectively makes the semicolon
|
|
; a non-usable character for PJSIP endpoint names, extensions,
|
|
; and AORs. This can be useful for improving compatability with
|
|
; an ITSP that likes to use user options for whatever reason.
|
|
; Example:
|
|
; URI: "sip:1235557890;phone-context=national@x.x.x.x;user=phone"
|
|
; The user field is "1235557890;phone-context=national"
|
|
; Which becomes this: "1235557890"
|
|
;
|
|
; Note: The caller-id and redirecting number strings obtained
|
|
; from incoming SIP URI user fields are always truncated at the
|
|
; first semicolon.
|
|
|
|
;send_contact_status_on_update_registration=no ; Enable sending AMI ContactStatus
|
|
; event when a device refreshes its registration
|
|
; (default: "no")
|
|
|
|
;taskprocessor_overload_trigger=global
|
|
; Set the trigger the distributor will use to detect
|
|
; taskprocessor overloads. When triggered, the distributor
|
|
; will not accept any new requests until the overload has
|
|
; cleared.
|
|
; "global": (default) Any taskprocessor overload will trigger.
|
|
; "pjsip_only": Only pjsip taskprocessor overloads will trigger.
|
|
; "none": No overload detection will be performed.
|
|
; WARNING: The "none" and "pjsip_only" options should be used
|
|
; with extreme caution and only to mitigate specific issues.
|
|
; Under certain conditions they could make things worse.
|
|
|
|
;norefersub=yes ; Enable sending norefersub option tag in Supported header to advertise
|
|
; that the User Agent is capable of accepting a REFER request with
|
|
; creating an implicit subscription (see RFC 4488).
|
|
; (default: "yes")
|
|
|
|
; MODULE PROVIDING BELOW SECTION(S): res_pjsip_acl
|
|
;==========================ACL SECTION OPTIONS=========================
|
|
;[acl]
|
|
; SYNOPSIS: Access Control List
|
|
;acl= ; List of IP ACL section names in acl conf (default: "")
|
|
;contact_acl= ; List of Contact ACL section names in acl conf (default: "")
|
|
;contact_deny= ; List of Contact header addresses to deny (default: "")
|
|
;contact_permit= ; List of Contact header addresses to permit (default:
|
|
; "")
|
|
;deny= ; List of IP addresses to deny access from (default: "")
|
|
;permit= ; List of IP addresses to permit access from (default: "")
|
|
;type= ; Must be of type acl (default: "")
|
|
|
|
|
|
|
|
|
|
; MODULE PROVIDING BELOW SECTION(S): res_pjsip_outbound_registration
|
|
;==========================REGISTRATION SECTION OPTIONS=========================
|
|
;[registration]
|
|
; SYNOPSIS: The configuration for outbound registration
|
|
;auth_rejection_permanent=yes ; Determines whether failed authentication
|
|
; challenges are treated as permanent failures
|
|
; (default: "yes")
|
|
;client_uri= ; Client SIP URI used when attemping outbound registration
|
|
; (default: "")
|
|
;contact_user= ; Contact User to use in request (default: "")
|
|
;expiration=3600 ; Expiration time for registrations in seconds
|
|
; (default: "3600")
|
|
;max_retries=10 ; Maximum number of registration attempts (default: "10")
|
|
;outbound_auth= ; Authentication object to be used for outbound registrations
|
|
; (default: "")
|
|
;outbound_proxy= ; Proxy through which to send registrations, a full SIP URI
|
|
; must be provided (default: "")
|
|
;retry_interval=60 ; Interval in seconds between retries if outbound
|
|
; registration is unsuccessful (default: "60")
|
|
;forbidden_retry_interval=0 ; Interval used when receiving a 403 Forbidden
|
|
; response (default: "0")
|
|
;fatal_retry_interval=0 ; Interval used when receiving a fatal response.
|
|
; (default: "0") A fatal response is any permanent
|
|
; failure (non-temporary 4xx, 5xx, 6xx) response
|
|
; received from the registrar. NOTE - if also set
|
|
; the 'forbidden_retry_interval' takes precedence
|
|
; over this one when a 403 is received. Also, if
|
|
; 'auth_rejection_permanent' equals 'yes' a 401 and
|
|
; 407 become subject to this retry interval.
|
|
;server_uri= ; SIP URI of the server to register against (default: "")
|
|
;transport= ; Transport used for outbound authentication (default: "")
|
|
;line= ; When enabled this option will cause a 'line' parameter to be
|
|
; added to the Contact header placed into the outgoing
|
|
; registration request. If the remote server sends a call
|
|
; this line parameter will be used to establish a relationship
|
|
; to the outbound registration, ultimately causing the
|
|
; configured endpoint to be used (default: "no")
|
|
;endpoint= ; When line support is enabled this configured endpoint name
|
|
; is used for incoming calls that are related to the outbound
|
|
; registration (default: "")
|
|
;type= ; Must be of type registration (default: "")
|
|
|
|
|
|
|
|
|
|
; MODULE PROVIDING BELOW SECTION(S): res_pjsip_endpoint_identifier_ip
|
|
;==========================IDENTIFY SECTION OPTIONS=========================
|
|
;[identify]
|
|
; SYNOPSIS: Identifies endpoints via some criteria.
|
|
;
|
|
; NOTE: If multiple matching criteria are provided then an inbound request will
|
|
; be matched to the endpoint if it matches ANY of the criteria.
|
|
;endpoint= ; Name of endpoint identified (default: "")
|
|
;srv_lookups=yes ; Perform SRV lookups for provided hostnames. (default: yes)
|
|
;match= ; Comma separated list of IP addresses, networks, or hostnames to match
|
|
; against (default: "")
|
|
;match_header= ; SIP header with specified value to match against (default: "")
|
|
;type= ; Must be of type identify (default: "")
|
|
|
|
|
|
|
|
|
|
;========================PHONEPROV_USER SECTION OPTIONS=======================
|
|
;[phoneprov]
|
|
; SYNOPSIS: Contains variables for autoprovisioning each user
|
|
;endpoint= ; The endpoint from which to gather username, secret, etc. (default: "")
|
|
;PROFILE= ; The name of a profile configured in phoneprov.conf (default: "")
|
|
;MAC= ; The mac address for this user (default: "")
|
|
;OTHERVAR= ; Any other name value pair to be used in templates (default: "")
|
|
; Common variables include LINE, LINEKEYS, etc.
|
|
; See phoneprov.conf.sample for others.
|
|
;type= ; Must be of type phoneprov (default: "")
|
|
|
|
|
|
|
|
; MODULE PROVIDING BELOW SECTION(S): res_pjsip_outbound_publish
|
|
;======================OUTBOUND_PUBLISH SECTION OPTIONS=====================
|
|
; See https://wiki.asterisk.org/wiki/display/AST/Publishing+Extension+State
|
|
; for more information.
|
|
;[outbound-publish]
|
|
;type=outbound-publish ; Must be of type 'outbound-publish'.
|
|
|
|
;expiration=3600 ; Expiration time for publications in seconds
|
|
|
|
;outbound_auth= ; Authentication object(s) to be used for outbound
|
|
; publishes.
|
|
; This is a comma-delimited list of auth sections
|
|
; defined in pjsip.conf used to respond to outbound
|
|
; authentication challenges.
|
|
; Using the same auth section for inbound and
|
|
; outbound authentication is not recommended. There
|
|
; is a difference in meaning for an empty realm
|
|
; setting between inbound and outbound authentication
|
|
; uses. See the auth realm description for details.
|
|
|
|
;outbound_proxy= ; SIP URI of the outbound proxy used to send
|
|
; publishes
|
|
|
|
;server_uri= ; SIP URI of the server and entity to publish to.
|
|
; This is the URI at which to find the entity and
|
|
; server to send the outbound PUBLISH to.
|
|
; This URI is used as the request URI of the outbound
|
|
; PUBLISH request from Asterisk.
|
|
|
|
;from_uri= ; SIP URI to use in the From header.
|
|
; This is the URI that will be placed into the From
|
|
; header of outgoing PUBLISH messages. If no URI is
|
|
; specified then the URI provided in server_uri will
|
|
; be used.
|
|
|
|
;to_uri= ; SIP URI to use in the To header.
|
|
; This is the URI that will be placed into the To
|
|
; header of outgoing PUBLISH messages. If no URI is
|
|
; specified then the URI provided in server_uri will
|
|
; be used.
|
|
|
|
;event= ; Event type of the PUBLISH.
|
|
|
|
;max_auth_attempts= ; Maximum number of authentication attempts before
|
|
; stopping the pub.
|
|
|
|
;transport= ; Transport used for outbound publish.
|
|
; A transport configured in pjsip.conf. As with other
|
|
; res_pjsip modules, this will use the first
|
|
; available transport of the appropriate type if
|
|
; unconfigured.
|
|
|
|
;multi_user=no ; Enable multi-user support (Asterisk 14+ only)
|
|
|
|
|
|
|
|
; MODULE PROVIDING BELOW SECTION(S): res_pjsip_pubsub
|
|
;=============================RESOURCE-LIST===================================
|
|
; See https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30278158
|
|
; for more information.
|
|
;[resource_list]
|
|
;type=resource_list ; Must be of type 'resource_list'.
|
|
|
|
;event= ; The SIP event package that the list resource.
|
|
; belongs to. The SIP event package describes the
|
|
; types of resources that Asterisk reports the state
|
|
; of.
|
|
|
|
;list_item= ; The name of a resource to report state on.
|
|
; In general Asterisk looks up list items in the
|
|
; following way:
|
|
; 1. Check if the list item refers to another
|
|
; configured resource list.
|
|
; 2. Pass the name of the resource off to
|
|
; event-package-specific handlers to find the
|
|
; specified resource.
|
|
; The second part means that the way the list item
|
|
; is specified depends on what type of list this is.
|
|
; For instance, if you have the event set to
|
|
; presence, then list items should be in the form of
|
|
; dialplan_extension@dialplan_context. For
|
|
; message-summary, mailbox names should be listed.
|
|
|
|
;full_state=no ; Indicates if the entire list's state should be
|
|
; sent out.
|
|
; If this option is enabled, and a resource changes
|
|
; state, then Asterisk will construct a notification
|
|
; that contains the state of all resources in the
|
|
; list. If the option is disabled, Asterisk will
|
|
; construct a notification that only contains the
|
|
; states of resources that have changed.
|
|
; NOTE: Even with this option disabled, there are
|
|
; certain situations where Asterisk is forced to send
|
|
; a notification with the states of all resources in
|
|
; the list. When a subscriber renews or terminates
|
|
; its subscription to the list, Asterisk MUST send
|
|
; a full state notification.
|
|
|
|
;notification_batch_interval=0
|
|
; Time Asterisk should wait, in milliseconds,
|
|
; before sending notifications.
|
|
|
|
;==========================INBOUND_PUBLICATION================================
|
|
; See https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP
|
|
; for more information.
|
|
;[inbound-publication]
|
|
;type= ; Must be of type 'inbound-publication'.
|
|
|
|
;endpoint= ; Optional name of an endpoint that is only allowed
|
|
; to publish to this resource.
|
|
|
|
|
|
; MODULE PROVIDING BELOW SECTION(S): res_pjsip_publish_asterisk
|
|
;==========================ASTERISK_PUBLICATION===============================
|
|
; See https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP
|
|
; for more information.
|
|
;[asterisk-publication]
|
|
;type=asterisk-publication ; Must be of type 'asterisk-publication'.
|
|
|
|
;devicestate_publish= ; Optional name of a publish item that can be used
|
|
; to publish a req.
|
|
|
|
;mailboxstate_publish= ; Optional name of a publish item that can be used
|
|
; to publish a req.
|
|
|
|
;device_state=no ; Whether we should permit incoming device state
|
|
; events.
|
|
|
|
;device_state_filter= ; Optional regular expression used to filter what
|
|
; devices we accept events for.
|
|
|
|
;mailbox_state=no ; Whether we should permit incoming mailbox state
|
|
; events.
|
|
|
|
;mailbox_state_filter= ; Optional regular expression used to filter what
|
|
; mailboxes we accept events for.
|
|
|
|
[transport-udp]
|
|
type=transport
|
|
protocol=udp
|
|
bind=0.0.0.0
|
|
|
|
|
|
; questa acl si applica a tutto
|
|
[acl]
|
|
type=acl
|
|
deny=0.0.0.0/0.0.0.0
|
|
permit=192.168.1.0/255.255.255.0
|
|
|
|
[removeme]
|
|
type=endpoint
|
|
deny=0.0.0.0/0.0.0.0
|
|
permit=192.168.1.0/255.255.255.0
|
|
transport=transport-udp
|
|
language=it
|
|
aors=removeme
|
|
auth=removeme
|
|
|
|
[removeme]
|
|
type=aors
|
|
[removeme]
|
|
type=auth
|
|
username=removeme
|
|
password=removeme
|
|
|
|
[lan](!)
|
|
type=endpoint
|
|
deny=0.0.0.0/0.0.0.0
|
|
permit=192.168.1.0/255.255.255.0
|
|
transport=transport-udp
|
|
language=it
|
|
disallow=all
|
|
; allow=ilbc
|
|
; allow=g729
|
|
; allow=gsm
|
|
; allow=g723
|
|
; allow=silk
|
|
; allow=silk16
|
|
; allow=silk24
|
|
allow=ulaw
|
|
allow=speex16
|
|
; allow=speex32
|
|
|
|
[interno](!,lan)
|
|
context=from-interni
|
|
|
|
[ext](!,lan)
|
|
context=from-ext
|
|
|
|
|
|
[baseauth](!)
|
|
type=auth
|
|
auth_type=userpass
|
|
|
|
[baseaor](!)
|
|
type=aor
|
|
max_contacts=1
|
|
|
|
[400](interno)
|
|
context=from-mixer
|
|
aors = 400
|
|
auth = 400
|
|
|
|
[400](baseaor)
|
|
|
|
[400](baseauth)
|
|
username=400
|
|
password=mixerone
|
|
|
|
[401](interno)
|
|
context=from-regia
|
|
aors = 401
|
|
auth=401
|
|
|
|
[401](baseaor)
|
|
[401](baseauth)
|
|
username=401
|
|
password=pass401
|
|
|
|
|
|
[402](ext)
|
|
; XXX: togli questo commento e fallo tornare un (interno)
|
|
; context=from-regia
|
|
aors = 402
|
|
auth=402
|
|
|
|
[402](baseaor)
|
|
[402](baseauth)
|
|
username=402
|
|
password=pass402
|
|
|
|
|
|
|
|
[403](interno)
|
|
aors = 403
|
|
auth=403
|
|
|
|
[403](baseaor)
|
|
[403](baseauth)
|
|
username=403
|
|
password=pass403
|
|
|
|
|
|
|
|
[404](interno)
|
|
aors = 404
|
|
auth=404
|
|
|
|
[404](baseaor)
|
|
[404](baseauth)
|
|
username=404
|
|
password=pass404
|
|
|
|
|
|
|
|
[020202](ext)
|
|
aors = 020202
|
|
auth=020202
|
|
|
|
[020202](baseaor)
|
|
[020202](baseauth)
|
|
username=020202
|
|
password=pass020202
|