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conf included
2022-02-19 17:01:13 +01:00
; PJSIP Configuration Samples and Quick Reference
;
; This file has several very basic configuration examples, to serve as a quick
; reference to jog your memory when you need to write up a new configuration.
; It is not intended to teach PJSIP configuration or serve as an exhaustive
; reference of options and potential scenarios.
;
; This file has two main sections.
; First, manually written examples to serve as a handy reference.
; Second, a list of all possible PJSIP config options by section. This is
; pulled from the XML config help. It only shows the synopsis for every item.
; If you want to see more detail please check the documentation sources
; mentioned at the top of this file.
; ============================================================================
; NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
;
; This file does not maintain the complete option documentation.
; ============================================================================
; Documentation
;
; The official documentation is at http://wiki.asterisk.org
; You can read the XML configuration help via Asterisk command line with
; "config show help res_pjsip", then you can drill down through the various
; sections and their options.
;
;========!!!!!!!!!!!!!!!!!!! SECURITY NOTICE !!!!!!!!!!!!!!!!!!!!===========
;
; At a minimum please read the file "README-SERIOUSLY.bestpractices.txt",
; located in the Asterisk source directory before starting Asterisk.
; Otherwise you risk allowing the security of the Asterisk system to be
; compromised. Beyond that please visit and read the security information on
; the wiki at: https://wiki.asterisk.org/wiki/x/EwFB
;
; A few basics to pay attention to:
;
; Anonymous Calls
;
; By default anonymous inbound calls via PJSIP are not allowed. If you want to
; route anonymous calls you'll need to define an endpoint named "anonymous".
; res_pjsip_endpoint_identifier_anonymous.so handles that functionality so it
; must be loaded. It is not recommended to accept anonymous calls.
;
; Access Control Lists
;
; See the example ACL configuration in this file. Read the configuration help
; for the section and all of its options. Look over the samples in acl.conf
; and documentation at https://wiki.asterisk.org/wiki/x/uA80AQ
; If possible, restrict access to only networks and addresses you trust.
;
; Dialplan Contexts
;
; When defining configuration (such as an endpoint) that links into
; dialplan configuration, be aware of what that dialplan does. It's easy to
; accidentally provide access to internal or outbound dialing extensions which
; could cost you severely. The "context=" line in endpoint configuration
; determines which dialplan context inbound calls will enter into.
;
;=============================================================================
; Overview of Configuration Section Types Used in the Examples
;
; * Transport "transport"
; * Configures res_pjsip transport layer interaction.
; * Endpoint "endpoint"
; * Configures core SIP functionality related to SIP endpoints.
; * Authentication "auth"
; * Stores inbound or outbound authentication credentials for use by trunks,
; endpoints, registrations.
; * Address of Record "aor"
; * Stores contact information for use by endpoints.
; * Endpoint Identification "identify"
; * Maps a host directly to an endpoint
; * Access Control List "acl"
; * Defines a permission list or references one stored in acl.conf
; * Registration "registration"
; * Contains information about an outbound SIP registration
; * Resource Lists
; * Contains information for configuring resource lists.
; * Phone Provisioning "phoneprov"
; * Contains information needed by res_phoneprov for autoprovisioning
; The following sections show example configurations for various scenarios.
; Most require a couple or more configuration types configured in concert.
;=============================================================================
; Naming of Configuration Sections
;
; Configuration section names are denoted with enclosing brackets,
; e.g. [6001]
; In most cases, you can name a section whatever makes sense to you. For example
; you might name a transport [transport-udp-nat] to help you remember how that
; section is being used. However, in some cases, ("endpoint" and "aor" types)
; the section name has a relationship to its function.
;
; Depending on the modules loaded, Asterisk can match SIP requests to an
; endpoint or aor in a few ways:
;
; 1) Match a section name for endpoint type sections to the username in the
; "From" header of inbound SIP requests.
; 2) Match a section name for aor type sections to the username in the "To"
; header of inbound SIP REGISTER requests.
; 3) With an identify type section configured, match an inbound SIP request of
; any type to an endpoint or aor based on the IP source address of the
; request.
;
; Note that sections can have the same name as long as their "type" options are
; set to different values. In most cases it makes sense to have associated
; configuration sections use the same name, as you'll see in the examples within
; this file.
;===============EXAMPLE TRANSPORTS============================================
;
; A few examples for potential transport options.
;
; For the NAT transport example, be aware that the options starting with
; the prefix "external_" will only apply to communication with addresses
; outside the range set with "local_net=".
;
; You can have more than one of any type of transport, as long as it doesn't
; use the same resources (bind address, port, etc) as the others.
; Basic UDP transport
;
;[transport-udp]
;type=transport
;protocol=udp ;udp,tcp,tls,ws,wss
;bind=0.0.0.0
; UDP transport behind NAT
;
;[transport-udp-nat]
;type=transport
;protocol=udp
;bind=0.0.0.0
;local_net=192.0.2.0/24
;external_media_address=203.0.113.1
;external_signaling_address=203.0.113.1
; Basic IPv6 UDP transport
;
;[transport-udp-ipv6]
;type=transport
;protocol=udp
;bind=::
; Example IPv4 TLS transport
;
;[transport-tls]
;type=transport
;protocol=tls
;bind=0.0.0.0
;cert_file=/path/mycert.crt
;priv_key_file=/path/mykey.key
;cipher=ADH-AES256-SHA,ADH-AES128-SHA
;method=tlsv1
;===============OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION============
;
; This is a simple registration that works with some SIP trunking providers.
; You'll need to set up the auth example "mytrunk_auth" below to enable outbound
; authentication. Note that we "outbound_auth=" use for outbound authentication
; instead of "auth=", which is for inbound authentication.
;
; If you are registering to a server from behind NAT, be sure you assign a transport
; that is appropriately configured with NAT related settings. See the NAT transport example.
;
; "contact_user=" sets the SIP contact header's user portion of the SIP URI
; this will affect the extension reached in dialplan when the far end calls you at this
; registration. The default is 's'.
;
; If you would like to enable line support and have incoming calls related to this
; registration go to an endpoint automatically the "line" and "endpoint" options must
; be set. The "endpoint" option specifies what endpoint the incoming call should be
; associated with.
;[mytrunk]
;type=registration
;transport=transport-udp
;outbound_auth=mytrunk_auth
;server_uri=sip:sip.example.com
;client_uri=sip:1234567890@sip.example.com
;contact_user=1234567890
;retry_interval=60
;forbidden_retry_interval=600
;expiration=3600
;line=yes
;endpoint=mytrunk
;[mytrunk_auth]
;type=auth
;auth_type=userpass
;password=1234567890
;username=1234567890
;realm=sip.example.com
;===============ENDPOINT CONFIGURED AS A TRUNK, OUTBOUND AUTHENTICATION=======
;
; This is one way to configure an endpoint as a trunk. It is set up with
; "outbound_auth=" to enable authentication when dialing out through this
; endpoint. There is no inbound authentication set up since a provider will
; not normally authenticate when calling you.
;
; The identify configuration enables IP address matching against this endpoint.
; For calls from a trunking provider, the From user may be different every time,
; so we want to match against IP address instead of From user.
;
; If you want the provider of your trunk to know where to send your calls
; you'll need to use an outbound registration as in the example above this
; section.
;
; NAT
;
; At a basic level configure the endpoint with a transport that is set up
; with the appropriate NAT settings. There may be some additional settings you
; need here based on your NAT/Firewall scenario. Look to the CLI config help
; "config show help res_pjsip endpoint" or on the wiki for other NAT related
; options and configuration. We've included a few below.
;
; AOR
;
; Endpoints use one or more AOR sections to store their contact details.
; You can define multiple contact addresses in SIP URI format in multiple
; "contact=" entries.
;
;[mytrunk]
;type=endpoint
;transport=transport-udp
;context=from-external
;disallow=all
;allow=ulaw
;outbound_auth=mytrunk_auth
;aors=mytrunk
; ;A few NAT relevant options that may come in handy.
;force_rport=yes ;It's a good idea to read the configuration help for each
;direct_media=no ;of these options.
;ice_support=yes
;[mytrunk]
;type=aor
;contact=sip:198.51.100.1:5060
;contact=sip:198.51.100.2:5060
;[mytrunk]
;type=identify
;endpoint=mytrunk
;match=198.51.100.1
;match=198.51.100.2
;match=192.168.10.0:5061/24
;=============ENDPOINT CONFIGURED AS A TRUNK, INBOUND AUTH AND REGISTRATION===
;
; Here we are allowing a remote device to register to Asterisk and requiring
; that they authenticate for registration and calls.
; You'll note that this configuration is essentially the same as configuring
; an endpoint for use with a SIP phone.
;[7000]
;type=endpoint
;context=from-external
;disallow=all
;allow=ulaw
;transport=transport-udp
;auth=7000
;aors=7000
;[7000]
;type=auth
;auth_type=userpass
;password=7000
;username=7000
;[7000]
;type=aor
;max_contacts=1
;===============ENDPOINT CONFIGURED FOR USE WITH A SIP PHONE==================
;
; This example includes the endpoint, auth and aor configurations. It
; requires inbound authentication and allows registration, as well as references
; a transport that you'll need to uncomment from the previous examples.
;
; Uncomment one of the transport lines to choose which transport you want. If
; not specified then the default transport chosen is the first compatible transport
; in the configuration file for the contact URL.
;
; Modify the "max_contacts=" line to change how many unique registrations to allow.
;
; Use the "contact=" line instead of max_contacts= if you want to statically
; define the location of the device.
;
; If using the TLS enabled transport, you may want the "media_encryption=sdes"
; option to additionally enable SRTP, though they are not mutually inclusive.
;
; If this endpoint were remote, and it was using a transport configured for NAT
; then you likely want to use "direct_media=no" to prevent audio issues.
;[6001]
;type=endpoint
;transport=transport-udp
;context=from-internal
;disallow=all
;allow=ulaw
;allow=gsm
;auth=6001
;aors=6001
;
; A few more transports to pick from, and some related options below them.
;
;transport=transport-tls
;media_encryption=sdes
;transport=transport-udp-ipv6
;transport=transport-udp-nat
;direct_media=no
;
; MWI related options
;aggregate_mwi=yes
;mailboxes=6001@default,7001@default
;mwi_from_user=6001
;
; Extension and Device state options
;
;device_state_busy_at=1
;allow_subscribe=yes
;sub_min_expiry=30
;
; STIR/SHAKEN support.
;
;stir_shaken=no
;[6001]
;type=auth
;auth_type=userpass
;password=6001
;username=6001
;[6001]
;type=aor
;max_contacts=1
;contact=sip:6001@192.0.2.1:5060
;===============ENDPOINT BEHIND NAT OR FIREWALL===============================
;
; This example assumes your transport is configured with a public IP and the
; endpoint itself is behind NAT and maybe a firewall, rather than having
; Asterisk behind NAT. For the sake of simplicity, we'll assume a typical
; VOIP phone. The most important settings to configure are:
;
; * direct_media, to ensure Asterisk stays in the media path
; * rtp_symmetric and force_rport options to help the far-end NAT/firewall
;
; Depending on the settings of your remote SIP device or NAT/firewall device
; you may have to experiment with a combination of these settings.
;
; If both Asterisk and the remote phones are a behind NAT/firewall then you'll
; have to make sure to use a transport with appropriate settings (as in the
; transport-udp-nat example).
;
;[6002]
;type=endpoint
;transport=transport-udp
;context=from-internal
;disallow=all
;allow=ulaw
;auth=6002
;aors=6002
;direct_media=no
;rtp_symmetric=yes
;force_rport=yes
;rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port
;ice_support=yes ;This is specific to clients that support NAT traversal
;for media via ICE,STUN,TURN. See the wiki at:
;https://wiki.asterisk.org/wiki/x/D4FHAQ
;for a deeper explanation of this topic.
;[6002]
;type=auth
;auth_type=userpass
;password=6002
;username=6002
;[6002]
;type=aor
;max_contacts=2
;============EXAMPLE ACL CONFIGURATION==========================================
;
; The ACL or Access Control List section defines a set of permissions to permit
; or deny access to various address or addresses. Alternatively it references an
; ACL configuration already set in acl.conf.
;
; The ACL configuration is independent of individual endpoint configuration and
; operates on all inbound SIP communication using res_pjsip.
; Reference an ACL defined in acl.conf.
;
;[acl]
;type=acl
;acl=example_named_acl1
; Reference a contactacl specifically.
;
;[acl]
;type=acl
;contact_acl=example_contact_acl1
; Define your own ACL here in pjsip.conf and
; permit or deny by IP address or range.
;
;[acl]
;type=acl
;deny=0.0.0.0/0.0.0.0
;permit=209.16.236.0/24
;deny=209.16.236.1
; Restrict based on Contact Headers rather than IP.
; Define options multiple times for various addresses or use a comma-delimited string.
;
;[acl]
;type=acl
;contact_deny=0.0.0.0/0.0.0.0
;contact_permit=209.16.236.0/24
;contact_permit=209.16.236.1
;contact_permit=209.16.236.2,209.16.236.3
; Restrict based on Contact Headers rather than IP and use
; advanced syntax. Note the bang symbol used for "NOT", so we can deny
; 209.16.236.12/32 within the permit= statement.
;
;[acl]
;type=acl
;contact_deny=0.0.0.0/0.0.0.0
;contact_permit=209.16.236.0
;permit=209.16.236.0/24, !209.16.236.12/32
;============EXAMPLE RLS CONFIGURATION==========================================
;
;Asterisk provides support for RFC 4662 Resource List Subscriptions. This allows
;for an endpoint to, through a single subscription, subscribe to the states of
;multiple resources. Resource lists are configured in pjsip.conf using the
;resource_list configuration object. Below is an example of a resource list that
;allows an endpoint to subscribe to the presence of alice, bob, and carol.
;[my_list]
;type=resource_list
;list_item=alice
;list_item=bob
;list_item=carol
;event=presence
;The "event" option in the resource list corresponds to the SIP event-package
;that the subscribed resources belong to. A resource list can only provide states
;for resources that belong to the same event-package. This means that you cannot
;create a list that is a combination of presence and message-summary resources,
;for instance. Any event-package that Asterisk supports can be used in a resource
;list (presence, dialog, and message-summary). Whenever support for a new event-
;package is added to Asterisk, support for that event-package in resource lists
;will automatically be supported.
;The "list_item" options indicate the names of resources to subscribe to. The
;way these are interpreted is event-package specific. For instance, with presence
;list_items, hints in the dialplan are looked up. With message-summary list_items,
;mailboxes are looked up using your installed voicemail provider (app_voicemail
;by default). Note that in the above example, the list_item options were given
;one per line. However, it is also permissible to provide multiple list_item
;options on a single line (e.g. list_item = alice,bob,carol).
;In addition to the options presented in the above configuration, there are two
;more configuration options that can be set.
; * full_state: dictates whether Asterisk should always send the states of
; all resources in the list at once. Defaults to "no". You should only set
; this to "yes" if you are interoperating with an endpoint that does not
; behave correctly when partial state notifications are sent to it.
; * notification_batch_interval: By default, Asterisk will send a NOTIFY request
; immediately when a resource changes state. This option causes Asterisk to
; start batching resource state changes for the specified number of milliseconds
; after a resource changes states. This way, if multiple resources change state
; within a brief interval, Asterisk can send a single NOTIFY request with all
; of the state changes reflected in it.
;There is a limitation to the size of resource lists in Asterisk. If a constructed
;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
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