Difference between revisions of "Asterisk-pjsip.conf"

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== endpoint type ==
 
== endpoint type ==
=== Example ===
+
=== Options ===
 +
* rtp_symmetric=no
 +
: Enforce that RTP must be symmetric. Send media to the address and port from which Asterisk receives it, regardless of where SDP indicates that it should be sent.  (default: "no")
 +
 
 +
* force_rport
 +
: Send responses to the source IP address and port as though port were present, even if it's not.
 +
 
 +
* rewrite_contact
 +
: Rewrite SIP Contact to the source address and port of the request so that subsequent requests go to that address and port.
 +
 
 
=== Example ===
 
=== Example ===
 
==== endpoint configured as a trunk, outbound authentication ====
 
==== endpoint configured as a trunk, outbound authentication ====
Line 244: Line 253:
 
type=aor
 
type=aor
 
contact=sip:203.0.113.1:5060
 
contact=sip:203.0.113.1:5060
 +
</pre>
 +
 +
== Sample ==
 +
<pre>
 +
; 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
 +
 +
 +
;=============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
 +
 +
;[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)
 +
 +
;==========================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 (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]
 +
;  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)
 +
;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=20 ; The interval (in seconds) at which to send keepalive
 +
                        ; messages on all active connection-oriented transports
 +
                        ; (default: "0")
 +
;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.
 
</pre>
 
</pre>
  

Latest revision as of 09:33, 10 September 2019

Overview

Asterisk 의 pjsip 모듈 설정파일 pjsip.conf 내용 정리.

Basic

; 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
; * Phone Provisioning "phoneprov"
;   * Contains information needed by res_phoneprov for autoprovisioning

ENDPOINT

transport type

transport type. TCP, UDP 혹은 WebSocket 과 같은 프로토콜이나 TLS/SSL 과 같은 암호화를 설정한다.

Example

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 tranport

[transport-udp-ipv6]
type=transport
protocol=udp
bind=::

Exmple 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

registration type

Example

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 ofthe 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 wo this registration to 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 type

Options

  • rtp_symmetric=no
Enforce that RTP must be symmetric. Send media to the address and port from which Asterisk receives it, regardless of where SDP indicates that it should be sent. (default: "no")
  • force_rport
Send responses to the source IP address and port as though port were present, even if it's not.
  • rewrite_contact
Rewrite SIP Contact to the source address and port of the request so that subsequent requests go to that address and port.

Example

endpoint configured as a trunk, outbound authentication

[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

endpoint configured as a trunk, inbound auth and registration

[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

[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
rtp_ipv6=yes
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

[6001]
type=auth
auth_type=userpass
password=6001
username=6001

[6001]
type=aor
max_contacts=1
contact=sip:6001@192.0.2.1:5060

AUTH

접속 인증 설정.

Example

An example with username and password authentication.

[auth6001]
type=auth
auth_type=userpass
password=6001
username=6001

MD5 authentication

[auth6001]
type=auth
auth_type=md5
md5_cred=51e63a3da6425a39aecc045ec45f1ae8
username=6001

AOR(Address of Record)

Example

Create automatic contact objects.

[6001]
type=aor
max_contacts=1

Create manual contact objects.

[6001]
type=aor
contact=sip:6001@192.0.2.1:5060

It's useful to note that you could define only the domain and omit the user portion of the SIP URI ifyou wanted. Then you could define the user portion dynamically in your dialplan when calling the Dial application. You'll likely do this when building an AOR/Endpoint combo to use for dialing out to an ITSP. For example: "Dial(PJSIP/${EXTEN}@mytrunk)"

[mytrunk]
type=aor
contact=sip:203.0.113.1:5060

Sample

; 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


;=============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

;[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)

;==========================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 (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]
;  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)
;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=20 ; The interval (in seconds) at which to send keepalive
                        ; messages on all active connection-oriented transports
                        ; (default: "0")
;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.

See also