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    1 ; PJSIP Configuration Samples and Quick Reference
    2 ;
    3 ; This file has several very basic configuration examples, to serve as a quick
    4 ; reference to jog your memory when you need to write up a new configuration.
    5 ; It is not intended to teach PJSIP configuration or serve as an exhaustive
    6 ; reference of options and potential scenarios.
    7 ;
    8 ; This file has two main sections.
    9 ; First, manually written examples to serve as a handy reference.
   10 ; Second, a list of all possible PJSIP config options by section. This is
   11 ; pulled from the XML config help. It only shows the synopsis for every item.
   12 ; If you want to see more detail please check the documentation sources
   13 ; mentioned at the top of this file.
   14 
   15 ; ============================================================================
   16 ; NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
   17 ;
   18 ; This file does not maintain the complete option documentation.
   19 ; ============================================================================
   20 
   21 ; Documentation
   22 ;
   23 ; The official documentation is at http://wiki.asterisk.org
   24 ; You can read the XML configuration help via Asterisk command line with
   25 ; "config show help res_pjsip", then you can drill down through the various
   26 ; sections and their options.
   27 ;
   28 
   29 ;========!!!!!!!!!!!!!!!!!!!  SECURITY NOTICE  !!!!!!!!!!!!!!!!!!!!===========
   30 ;
   31 ; At a minimum please read the file "README-SERIOUSLY.bestpractices.txt",
   32 ; located in the Asterisk source directory before starting Asterisk.
   33 ; Otherwise you risk allowing the security of the Asterisk system to be
   34 ; compromised. Beyond that please visit and read the security information on
   35 ; the wiki at: https://wiki.asterisk.org/wiki/x/EwFB
   36 ;
   37 ; A few basics to pay attention to:
   38 ;
   39 ; Anonymous Calls
   40 ;
   41 ; By default anonymous inbound calls via PJSIP are not allowed. If you want to
   42 ; route anonymous calls you'll need to define an endpoint named "anonymous".
   43 ; res_pjsip_endpoint_identifier_anonymous.so handles that functionality so it
   44 ; must be loaded. It is not recommended to accept anonymous calls.
   45 ;
   46 ; Access Control Lists
   47 ;
   48 ; See the example ACL configuration in this file. Read the configuration help
   49 ; for the section and all of its options. Look over the samples in acl.conf
   50 ; and documentation at https://wiki.asterisk.org/wiki/x/uA80AQ
   51 ; If possible, restrict access to only networks and addresses you trust.
   52 ;
   53 ; Dialplan Contexts
   54 ;
   55 ; When defining configuration (such as an endpoint) that links into
   56 ; dialplan configuration, be aware of what that dialplan does. It's easy to
   57 ; accidentally provide access to internal or outbound dialing extensions which
   58 ; could cost you severely. The "context=" line in endpoint configuration
   59 ; determines which dialplan context inbound calls will enter into.
   60 ;
   61 ;=============================================================================
   62 
   63 ; Overview of Configuration Section Types Used in the Examples
   64 ;
   65 ; * Transport "transport"
   66 ;   * Configures res_pjsip transport layer interaction.
   67 ; * Endpoint "endpoint"
   68 ;   * Configures core SIP functionality related to SIP endpoints.
   69 ; * Authentication "auth"
   70 ;   * Stores inbound or outbound authentication credentials for use by trunks,
   71 ;     endpoints, registrations.
   72 ; * Address of Record "aor"
   73 ;   * Stores contact information for use by endpoints.
   74 ; * Endpoint Identification "identify"
   75 ;   * Maps a host directly to an endpoint
   76 ; * Access Control List "acl"
   77 ;   * Defines a permission list or references one stored in acl.conf
   78 ; * Registration "registration"
   79 ;   * Contains information about an outbound SIP registration
   80 ; * Resource Lists
   81 ;   * Contains information for configuring resource lists.
   82 ; * Phone Provisioning "phoneprov"
   83 ;   * Contains information needed by res_phoneprov for autoprovisioning
   84 
   85 ; The following sections show example configurations for various scenarios.
   86 ; Most require a couple or more configuration types configured in concert.
   87 
   88 ;=============================================================================
   89 
   90 ; Naming of Configuration Sections
   91 ;
   92 ; Configuration section names are denoted with enclosing brackets,
   93 ; e.g. [6001]
   94 ; In most cases, you can name a section whatever makes sense to you. For example
   95 ; you might name a transport [transport-udp-nat] to help you remember how that
   96 ; section is being used. However, in some cases, ("endpoint" and "aor" types)
   97 ; the section name has a relationship to its function.
   98 ;
   99 ; Depending on the modules loaded, Asterisk can match SIP requests to an
  100 ; endpoint or aor in a few ways:
  101 ;
  102 ; 1) Match a section name for endpoint type sections to the username in the
  103 ;    "From" header of inbound SIP requests.
  104 ; 2) Match a section name for aor type sections to the username in the "To"
  105 ;    header of inbound SIP REGISTER requests.
  106 ; 3) With an identify type section configured, match an inbound SIP request of
  107 ;    any type to an endpoint or aor based on the IP source address of the
  108 ;    request.
  109 ;
  110 ; Note that sections can have the same name as long as their "type" options are
  111 ; set to different values. In most cases it makes sense to have associated
  112 ; configuration sections use the same name, as you'll see in the examples within
  113 ; this file.
  114 
  115 ;===============EXAMPLE TRANSPORTS============================================
  116 ;
  117 ; A few examples for potential transport options.
  118 ;
  119 ; For the NAT transport example, be aware that the options starting with
  120 ; the prefix "external_" will only apply to communication with addresses
  121 ; outside the range set with "local_net=".
  122 ;
  123 ; You can have more than one of any type of transport, as long as it doesn't
  124 ; use the same resources (bind address, port, etc) as the others.
  125 
  126 ; Basic UDP transport
  127 ;
  128 ;[transport-udp]
  129 ;type=transport
  130 ;protocol=udp    ;udp,tcp,tls,ws,wss,flow
  131 ;bind=0.0.0.0
  132 
  133 ; UDP transport behind NAT
  134 ;
  135 ;[transport-udp-nat]
  136 ;type=transport
  137 ;protocol=udp
  138 ;bind=0.0.0.0
  139 ;local_net=192.0.2.0/24
  140 ;external_media_address=203.0.113.1
  141 ;external_signaling_address=203.0.113.1
  142 
  143 ; Basic IPv6 UDP transport
  144 ;
  145 ;[transport-udp-ipv6]
  146 ;type=transport
  147 ;protocol=udp
  148 ;bind=::
  149 
  150 ; Example IPv4 TLS transport
  151 ;
  152 ;[transport-tls]
  153 ;type=transport
  154 ;protocol=tls
  155 ;bind=0.0.0.0
  156 ;cert_file=/path/mycert.crt
  157 ;priv_key_file=/path/mykey.key
  158 ;cipher=ADH-AES256-SHA,ADH-AES128-SHA
  159 ;method=tlsv1
  160 
  161 ; Example flow transport
  162 ;
  163 ; A flow transport is used to reference a flow of signaling with a specific
  164 ; target. All endpoints or other objects that reference the transport will use
  165 ; the same underlying transport and can share runtime discovered transport
  166 ; configuration (such as service routes). The protocol in use will be determined
  167 ; based on the URI used to establish the connection. Currently only TCP and TLS
  168 ; are supported.
  169 ;
  170 ;[transport-flow]
  171 ;type=transport
  172 ;protocol=flow
  173 
  174 ;===============OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION============
  175 ;
  176 ; This is a simple registration that works with some SIP trunking providers.
  177 ; You'll need to set up the auth example "mytrunk_auth" below to enable outbound
  178 ; authentication. Note that we "outbound_auth=" use for outbound authentication
  179 ; instead of "auth=", which is for inbound authentication.
  180 ;
  181 ; If you are registering to a server from behind NAT, be sure you assign a transport
  182 ; that is appropriately configured with NAT related settings. See the NAT transport example.
  183 ;
  184 ; "contact_user=" sets the SIP contact header's user portion of the SIP URI
  185 ; this will affect the extension reached in dialplan when the far end calls you at this
  186 ; registration. The default is 's'.
  187 ;
  188 ; If you would like to enable line support and have incoming calls related to this
  189 ; registration go to an endpoint automatically the "line" and "endpoint" options must
  190 ; be set. The "endpoint" option specifies what endpoint the incoming call should be
  191 ; associated with.
  192 
  193 ;[mytrunk]
  194 ;type=registration
  195 ;transport=transport-udp
  196 ;outbound_auth=mytrunk_auth
  197 ;server_uri=sip:sip.example.com
  198 ;client_uri=sip:1234567890@sip.example.com
  199 ;contact_user=1234567890
  200 ;retry_interval=60
  201 ;forbidden_retry_interval=600
  202 ;expiration=3600
  203 ;line=yes
  204 ;endpoint=mytrunk
  205 
  206 ;[mytrunk_auth]
  207 ;type=auth
  208 ;auth_type=userpass
  209 ;password=1234567890
  210 ;username=1234567890
  211 ;realm=sip.example.com
  212 
  213 ;===============ENDPOINT CONFIGURED AS A TRUNK, OUTBOUND AUTHENTICATION=======
  214 ;
  215 ; This is one way to configure an endpoint as a trunk. It is set up with
  216 ; "outbound_auth=" to enable authentication when dialing out through this
  217 ; endpoint. There is no inbound authentication set up since a provider will
  218 ; not normally authenticate when calling you.
  219 ;
  220 ; The identify configuration enables IP address matching against this endpoint.
  221 ; For calls from a trunking provider, the From user may be different every time,
  222 ; so we want to match against IP address instead of From user.
  223 ;
  224 ; If you want the provider of your trunk to know where to send your calls
  225 ; you'll need to use an outbound registration as in the example above this
  226 ; section.
  227 ;
  228 ; NAT
  229 ;
  230 ; At a basic level configure the endpoint with a transport that is set up
  231 ; with the appropriate NAT settings. There may be some additional settings you
  232 ; need here based on your NAT/Firewall scenario. Look to the CLI config help
  233 ; "config show help res_pjsip endpoint" or on the wiki for other NAT related
  234 ; options and configuration. We've included a few below.
  235 ;
  236 ; AOR
  237 ;
  238 ; Endpoints use one or more AOR sections to store their contact details.
  239 ; You can define multiple contact addresses in SIP URI format in multiple
  240 ; "contact=" entries.
  241 ;
  242 
  243 ;[mytrunk]
  244 ;type=endpoint
  245 ;transport=transport-udp
  246 ;context=from-external
  247 ;disallow=all
  248 ;allow=ulaw
  249 ;outbound_auth=mytrunk_auth
  250 ;aors=mytrunk
  251 ;                   ;A few NAT relevant options that may come in handy.
  252 ;force_rport=yes    ;It's a good idea to read the configuration help for each
  253 ;direct_media=no    ;of these options.
  254 ;ice_support=yes
  255 
  256 ;[mytrunk]
  257 ;type=aor
  258 ;contact=sip:198.51.100.1:5060
  259 ;contact=sip:198.51.100.2:5060
  260 
  261 ;[mytrunk]
  262 ;type=identify
  263 ;endpoint=mytrunk
  264 ;match=198.51.100.1
  265 ;match=198.51.100.2
  266 ;match=192.168.10.0:5061/24
  267 
  268 
  269 ;=============ENDPOINT CONFIGURED AS A TRUNK, INBOUND AUTH AND REGISTRATION===
  270 ;
  271 ; Here we are allowing a remote device to register to Asterisk and requiring
  272 ; that they authenticate for registration and calls.
  273 ; You'll note that this configuration is essentially the same as configuring
  274 ; an endpoint for use with a SIP phone.
  275 
  276 
  277 ;[7000]
  278 ;type=endpoint
  279 ;context=from-external
  280 ;disallow=all
  281 ;allow=ulaw
  282 ;transport=transport-udp
  283 ;auth=7000
  284 ;aors=7000
  285 
  286 ;[7000]
  287 ;type=auth
  288 ;auth_type=userpass
  289 ;password=7000
  290 ;username=7000
  291 
  292 ;[7000]
  293 ;type=aor
  294 ;max_contacts=1
  295 
  296 
  297 ;===============ENDPOINT CONFIGURED FOR USE WITH A SIP PHONE==================
  298 ;
  299 ; This example includes the endpoint, auth and aor configurations. It
  300 ; requires inbound authentication and allows registration, as well as references
  301 ; a transport that you'll need to uncomment from the previous examples.
  302 ;
  303 ; Uncomment one of the transport lines to choose which transport you want. If
  304 ; not specified then the default transport chosen is the first compatible transport
  305 ; in the configuration file for the contact URL.
  306 ;
  307 ; Modify the "max_contacts=" line to change how many unique registrations to allow.
  308 ;
  309 ; Use the "contact=" line instead of max_contacts= if you want to statically
  310 ; define the location of the device.
  311 ;
  312 ; If using the TLS enabled transport, you may want the "media_encryption=sdes"
  313 ; option to additionally enable SRTP, though they are not mutually inclusive.
  314 ;
  315 ; If this endpoint were remote, and it was using a transport configured for NAT
  316 ; then you likely want to use "direct_media=no" to prevent audio issues.
  317 
  318 
  319 ;[6001]
  320 ;type=endpoint
  321 ;transport=transport-udp
  322 ;context=from-internal
  323 ;disallow=all
  324 ;allow=ulaw
  325 ;allow=gsm
  326 ;auth=6001
  327 ;aors=6001
  328 ;
  329 ; A few more transports to pick from, and some related options below them.
  330 ;
  331 ;transport=transport-tls
  332 ;media_encryption=sdes
  333 ;transport=transport-udp-ipv6
  334 ;transport=transport-udp-nat
  335 ;direct_media=no
  336 ;
  337 ; MWI related options
  338 
  339 ;aggregate_mwi=yes
  340 ;mailboxes=6001@default,7001@default
  341 ;mwi_from_user=6001
  342 ;
  343 ; Extension and Device state options
  344 ;
  345 ;device_state_busy_at=1
  346 ;allow_subscribe=yes
  347 ;sub_min_expiry=30
  348 ;
  349 ; STIR/SHAKEN support.
  350 ;
  351 ;stir_shaken=no
  352 
  353 ;[6001]
  354 ;type=auth
  355 ;auth_type=userpass
  356 ;password=6001
  357 ;username=6001
  358 
  359 ;[6001]
  360 ;type=aor
  361 ;max_contacts=1
  362 ;contact=sip:6001@192.0.2.1:5060
  363 
  364 ;===============ENDPOINT BEHIND NAT OR FIREWALL===============================
  365 ;
  366 ; This example assumes your transport is configured with a public IP and the
  367 ; endpoint itself is behind NAT and maybe a firewall, rather than having
  368 ; Asterisk behind NAT. For the sake of simplicity, we'll assume a typical
  369 ; VOIP phone. The most important settings to configure are:
  370 ;
  371 ;  * direct_media, to ensure Asterisk stays in the media path
  372 ;  * rtp_symmetric and force_rport options to help the far-end NAT/firewall
  373 ;
  374 ; Depending on the settings of your remote SIP device or NAT/firewall device
  375 ; you may have to experiment with a combination of these settings.
  376 ;
  377 ; If both Asterisk and the remote phones are a behind NAT/firewall then you'll
  378 ; have to make sure to use a transport with appropriate settings (as in the
  379 ; transport-udp-nat example).
  380 ;
  381 ;[6002]
  382 ;type=endpoint
  383 ;transport=transport-udp
  384 ;context=from-internal
  385 ;disallow=all
  386 ;allow=ulaw
  387 ;auth=6002
  388 ;aors=6002
  389 ;direct_media=no
  390 ;rtp_symmetric=yes
  391 ;force_rport=yes
  392 ;rewrite_contact=yes  ; necessary if endpoint does not know/register public ip:port
  393 ;ice_support=yes   ;This is specific to clients that support NAT traversal
  394                    ;for media via ICE,STUN,TURN. See the wiki at:
  395                    ;https://wiki.asterisk.org/wiki/x/D4FHAQ
  396                    ;for a deeper explanation of this topic.
  397 
  398 ;[6002]
  399 ;type=auth
  400 ;auth_type=userpass
  401 ;password=6002
  402 ;username=6002
  403 
  404 ;[6002]
  405 ;type=aor
  406 ;max_contacts=2
  407 
  408 
  409 ;============EXAMPLE ACL CONFIGURATION==========================================
  410 ;
  411 ; The ACL or Access Control List section defines a set of permissions to permit
  412 ; or deny access to various address or addresses. Alternatively it references an
  413 ; ACL configuration already set in acl.conf.
  414 ;
  415 ; The ACL configuration is independent of individual endpoint configuration and
  416 ; operates on all inbound SIP communication using res_pjsip.
  417 
  418 ; Reference an ACL defined in acl.conf.
  419 ;
  420 ;[acl]
  421 ;type=acl
  422 ;acl=example_named_acl1
  423 
  424 ; Reference a contactacl specifically.
  425 ;
  426 ;[acl]
  427 ;type=acl
  428 ;contact_acl=example_contact_acl1
  429 
  430 ; Define your own ACL here in pjsip.conf and
  431 ; permit or deny by IP address or range.
  432 ;
  433 ;[acl]
  434 ;type=acl
  435 ;deny=0.0.0.0/0.0.0.0
  436 ;permit=209.16.236.0/24
  437 ;deny=209.16.236.1
  438 
  439 ; Restrict based on Contact Headers rather than IP.
  440 ; Define options multiple times for various addresses or use a comma-delimited string.
  441 ;
  442 ;[acl]
  443 ;type=acl
  444 ;contact_deny=0.0.0.0/0.0.0.0
  445 ;contact_permit=209.16.236.0/24
  446 ;contact_permit=209.16.236.1
  447 ;contact_permit=209.16.236.2,209.16.236.3
  448 
  449 ; Restrict based on Contact Headers rather than IP and use
  450 ; advanced syntax. Note the bang symbol used for "NOT", so we can deny
  451 ; 209.16.236.12/32 within the permit= statement.
  452 ;
  453 ;[acl]
  454 ;type=acl
  455 ;contact_deny=0.0.0.0/0.0.0.0
  456 ;contact_permit=209.16.236.0
  457 ;permit=209.16.236.0/24, !209.16.236.12/32
  458 
  459 
  460 ;============EXAMPLE RLS CONFIGURATION==========================================
  461 ;
  462 ;Asterisk provides support for RFC 4662 Resource List Subscriptions. This allows
  463 ;for an endpoint to, through a single subscription, subscribe to the states of
  464 ;multiple resources. Resource lists are configured in pjsip.conf using the
  465 ;resource_list configuration object. Below is an example of a resource list that
  466 ;allows an endpoint to subscribe to the presence of alice, bob, and carol.
  467 
  468 ;[my_list]
  469 ;type=resource_list
  470 ;list_item=alice
  471 ;list_item=bob
  472 ;list_item=carol
  473 ;event=presence
  474 
  475 ;The "event" option in the resource list corresponds to the SIP event-package
  476 ;that the subscribed resources belong to. A resource list can only provide states
  477 ;for resources that belong to the same event-package. This means that you cannot
  478 ;create a list that is a combination of presence and message-summary resources,
  479 ;for instance. Any event-package that Asterisk supports can be used in a resource
  480 ;list (presence, dialog, and message-summary). Whenever support for a new event-
  481 ;package is added to Asterisk, support for that event-package in resource lists
  482 ;will automatically be supported.
  483 
  484 ;The "list_item" options indicate the names of resources to subscribe to. The
  485 ;way these are interpreted is event-package specific. For instance, with presence
  486 ;list_items, hints in the dialplan are looked up. With message-summary list_items,
  487 ;mailboxes are looked up using your installed voicemail provider (app_voicemail
  488 ;by default). Note that in the above example, the list_item options were given
  489 ;one per line. However, it is also permissible to provide multiple list_item
  490 ;options on a single line (e.g. list_item = alice,bob,carol).
  491 
  492 ;In addition to the options presented in the above configuration, there are two
  493 ;more configuration options that can be set.
  494 ; * full_state: dictates whether Asterisk should always send the states of
  495 ;   all resources in the list at once. Defaults to "no". You should only set
  496 ;   this to "yes" if you are interoperating with an endpoint that does not
  497 ;   behave correctly when partial state notifications are sent to it.
  498 ; * notification_batch_interval: By default, Asterisk will send a NOTIFY request
  499 ;   immediately when a resource changes state. This option causes Asterisk to
  500 ;   start batching resource state changes for the specified number of milliseconds
  501 ;   after a resource changes states. This way, if multiple resources change state
  502 ;   within a brief interval, Asterisk can send a single NOTIFY request with all
  503 ;   of the state changes reflected in it.
  504 
  505 ;There is a limitation to the size of resource lists in Asterisk. If a constructed
  506 ;notification from Asterisk will exceed 64000 bytes, then the message is deemed
  507 ;too large to send. If you find that you are seeing error messages about SIP
  508 ;NOTIFY requests being too large to send, consider breaking your lists into
  509 ;sub-lists.
  510 
  511 ;============EXAMPLE PHONEPROV CONFIGURATION================================
  512 
  513 ; Before configuring provisioning here, see the documentation for res_phoneprov
  514 ; and configure phoneprov.conf appropriately.
  515 
  516 ; For each user to be autoprovisioned, a [phoneprov] configuration section
  517 ; must be created.  At a minimum, the 'type', 'PROFILE' and 'MAC' variables must
  518 ; be set.  All other variables are optional.
  519 ; Example:
  520 
  521 ;[1000]
  522 ;type=phoneprov               ; must be specified as 'phoneprov'
  523 ;endpoint=1000                ; Required only if automatic setting of
  524                               ; USERNAME, SECRET, DISPLAY_NAME and CALLERID
  525                               ; are needed.
  526 ;PROFILE=digium               ; required
  527 ;MAC=deadbeef4dad             ; required
  528 ;SERVER=myserver.example.com  ; A standard variable
  529 ;TIMEZONE=America/Denver      ; A standard variable
  530 ;MYVAR=somevalue              ; A user confdigured variable
  531 
  532 ; If the phoneprov sections have common variables, it is best to create a
  533 ; phoneprov template.  The example below will produce the same configuration
  534 ; as the one specified above except that MYVAR will be overridden for
  535 ; the specific user.
  536 ; Example:
  537 
  538 ;[phoneprov_defaults](!)
  539 ;type=phoneprov               ; must be specified as 'phoneprov'
  540 ;PROFILE=digium               ; required
  541 ;SERVER=myserver.example.com  ; A standard variable
  542 ;TIMEZONE=America/Denver      ; A standard variable
  543 ;MYVAR=somevalue              ; A user configured variable
  544 
  545 ;[1000](phoneprov_defaults)
  546 ;endpoint=1000                ; Required only if automatic setting of
  547                               ; USERNAME, SECRET, DISPLAY_NAME and CALLERID
  548                               ; are needed.
  549 ;MAC=deadbeef4dad             ; required
  550 ;MYVAR=someOTHERvalue         ; A user confdigured variable
  551 
  552 ; To have USERNAME and SECRET automatically set, the endpoint
  553 ; specified here must in turn have an outbound_auth section defined.
  554 
  555 ; Fuller example:
  556 
  557 ;[1000]
  558 ;type=endpoint
  559 ;outbound_auth=1000-auth
  560 ;callerid=My Name <8005551212>
  561 ;transport=transport-udp-nat
  562 
  563 ;[1000-auth]
  564 ;type=auth
  565 ;auth_type=userpass
  566 ;username=myname
  567 ;password=mysecret
  568 
  569 ;[phoneprov_defaults](!)
  570 ;type=phoneprov               ; must be specified as 'phoneprov'
  571 ;PROFILE=someprofile          ; required
  572 ;SERVER=myserver.example.com  ; A standard variable
  573 ;TIMEZONE=America/Denver      ; A standard variable
  574 ;MYVAR=somevalue              ; A user configured variable
  575 
  576 ;[1000](phoneprov_defaults)
  577 ;endpoint=1000                ; Required only if automatic setting of
  578                               ; USERNAME, SECRET, DISPLAY_NAME and CALLERID
  579                               ; are needed.
  580 ;MAC=deadbeef4dad             ; required
  581 ;MYVAR=someUSERvalue          ; A user confdigured variable
  582 ;LABEL=1000                   ; A standard variable
  583 
  584 ; The previous sections would produce a template substitution map as follows:
  585 
  586 ;MAC=deadbeef4dad               ;added by pp1000
  587 ;USERNAME=myname                ;automatically added by 1000-auth username
  588 ;SECRET=mysecret                ;automatically added by 1000-auth password
  589 ;PROFILE=someprofile            ;added by defaults
  590 ;SERVER=myserver.example.com    ;added by defaults
  591 ;SERVER_PORT=5060               ;added by defaults
  592 ;MYVAR=someUSERvalue            ;added by defaults but overdidden by user
  593 ;CALLERID=8005551212            ;automatically added by 1000 callerid
  594 ;DISPLAY_NAME=My Name           ;automatically added by 1000 callerid
  595 ;TIMEZONE=America/Denver        ;added by defaults
  596 ;TZOFFSET=252100                ;automatically calculated by res_phoneprov
  597 ;DST_ENABLE=1                   ;automatically calculated by res_phoneprov
  598 ;DST_START_MONTH=3              ;automatically calculated by res_phoneprov
  599 ;DST_START_MDAY=9               ;automatically calculated by res_phoneprov
  600 ;DST_START_HOUR=3               ;automatically calculated by res_phoneprov
  601 ;DST_END_MONTH=11               ;automatically calculated by res_phoneprov
  602 ;DST_END_MDAY=2                 ;automatically calculated by res_phoneprov
  603 ;DST_END_HOUR=1                 ;automatically calculated by res_phoneprov
  604 ;ENDPOINT_ID=1000               ;automatically added by this module
  605 ;AUTH_ID=1000-auth              ;automatically added by this module
  606 ;TRANSPORT_ID=transport-udp-nat ;automatically added by this module
  607 ;LABEL=1000                     ;added by user
  608 
  609 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip
  610 ;==========================ENDPOINT SECTION OPTIONS=========================
  611 ;[endpoint]
  612 ;  SYNOPSIS: Endpoint
  613 ;100rel=yes     ; Allow support for RFC3262 provisional ACK tags (default:
  614                 ; "yes")
  615 ;aggregate_mwi=yes      ;  (default: "yes")
  616 ;allow= ; Media Codec s to allow (default: "")
  617 ;allow_overlap=yes ; Enable RFC3578 overlap dialing support. (default: "yes")
  618 ;aors=  ; AoR s to be used with the endpoint (default: "")
  619 ;auth=  ; Authentication Object s associated with the endpoint (default: "")
  620 ;callerid=      ; CallerID information for the endpoint (default: "")
  621 ;callerid_privacy=allowed_not_screened      ; Default privacy level (default: "allowed_not_screened")
  622 ;callerid_tag=  ; Internal id_tag for the endpoint (default: "")
  623 ;context=default        ; Dialplan context for inbound sessions (default:
  624                         ; "default")
  625 ;direct_media_glare_mitigation=none     ; Mitigation of direct media re INVITE
  626                                         ; glare (default: "none")
  627 ;direct_media_method=invite     ; Direct Media method type (default: "invite")
  628 ;trust_connected_line=yes       ; Accept Connected Line updates from this endpoint
  629                                 ; (default: "yes")
  630 ;send_connected_line=yes        ; Send Connected Line updates to this endpoint
  631                                 ; (default: "yes")
  632 ;connected_line_method=invite   ; Connected line method type.
  633                                 ; When set to "invite", check the remote's
  634                                 ; Allow header and if UPDATE is allowed, send
  635                                 ; UPDATE instead of INVITE to avoid SDP
  636                                 ; renegotiation.  If UPDATE is not Allowed,
  637                                 ; send INVITE.
  638                                 ; If set to "update", send UPDATE regardless
  639                                 ; of what the remote Allows.
  640                                 ; (default: "invite")
  641 ;direct_media=yes       ; Determines whether media may flow directly between
  642                         ; endpoints (default: "yes")
  643 ;disable_direct_media_on_nat=no ; Disable direct media session refreshes when
  644                                 ; NAT obstructs the media session (default:
  645                                 ; "no")
  646 ;disallow=      ; Media Codec s to disallow (default: "")
  647 ;dtmf_mode=rfc4733      ; DTMF mode (default: "rfc4733")
  648 ;media_address=         ; IP address used in SDP for media handling (default: "")
  649 ;bind_rtp_to_media_address=     ; Bind the RTP session to the media_address.
  650                                 ; This causes all RTP packets to be sent from
  651                                 ; the specified address. (default: "no")
  652 ;force_rport=yes        ; Force use of return port (default: "yes")
  653 ;ice_support=no ; Enable the ICE mechanism to help traverse NAT (default: "no")
  654 ;identify_by=username   ; A comma-separated list of ways the Endpoint or AoR can be
  655                         ; identified.
  656                         ; "username": Identify by the From or To username and domain
  657                         ; "auth_username": Identify by the Authorization username and realm
  658                         ; "ip": Identify by the source IP address
  659                         ; "header": Identify by a configured SIP header value.
  660                         ; In the username and auth_username cases, if an exact match
  661                         ; on both username and domain/realm fails, the match is
  662                         ; retried with just the username.
  663                         ; (default: "username,ip")
  664 ;redirect_method=user   ; How redirects received from an endpoint are handled
  665                         ; (default: "user")
  666 ;mailboxes=     ; NOTIFY the endpoint when state changes for any of the specified mailboxes.
  667                 ; Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state
  668                 ; changes happen for any of the specified mailboxes. (default: "")
  669 ;voicemail_extension= ; The voicemail extension to send in the NOTIFY Message-Account header
  670                       ; (default: global/default_voicemail_extension)
  671 ;mwi_subscribe_replaces_unsolicited=no
  672                       ; An MWI subscribe will replace unsoliticed NOTIFYs
  673                       ; (default: "no")
  674 ;moh_suggest=default    ; Default Music On Hold class (default: "default")
  675 ;moh_passthrough=yes    ; Pass Music On Hold through using SIP re-invites with sendonly
  676                         ; when placing on hold and sendrecv when taking off hold
  677 ;outbound_auth= ; Authentication object used for outbound requests (default:
  678                 ; "")
  679 ;outbound_proxy=        ; Proxy through which to send requests, a full SIP URI
  680                         ; must be provided (default: "")
  681 ;rewrite_contact=no     ; Allow Contact header to be rewritten with the source
  682                         ; IP address port (default: "no")
  683 ;rtp_symmetric=no       ; Enforce that RTP must be symmetric (default: "no")
  684 ;send_diversion=yes     ; Send the Diversion header conveying the diversion
  685                         ; information to the called user agent (default: "yes")
  686 ;send_pai=no    ; Send the P Asserted Identity header (default: "no")
  687 ;send_rpid=no   ; Send the Remote Party ID header (default: "no")
  688 ;rpid_immediate=no      ; Send connected line updates on unanswered incoming calls immediately. (default: "no")
  689 ;timers_min_se=90       ; Minimum session timers expiration period (default:
  690                         ; "90")
  691 ;timers=yes     ; Session timers for SIP packets (default: "yes")
  692 ;timers_sess_expires=1800       ; Maximum session timer expiration period
  693                                 ; (default: "1800")
  694 ;transport=     ; Explicit transport configuration to use (default: "")
  695                 ; This will force the endpoint to use the specified transport
  696                 ; configuration to send SIP messages.  You need to already know
  697                 ; what kind of transport (UDP/TCP/IPv4/etc) the endpoint device
  698                 ; will use.
  699 
  700 ;trust_id_inbound=no    ; Accept identification information received from this
  701                         ; endpoint (default: "no")
  702 ;trust_id_outbound=no   ; Send private identification details to the endpoint
  703                         ; (default: "no")
  704 ;type=  ; Must be of type endpoint (default: "")
  705 ;use_ptime=no   ; Use Endpoint s requested packetisation interval (default:
  706                 ; "no")
  707 ;use_avpf=no    ; Determines whether res_pjsip will use and enforce usage of
  708                 ; AVPF for this endpoint (default: "no")
  709 ;media_encryption=no    ; Determines whether res_pjsip will use and enforce
  710                         ; usage of media encryption for this endpoint (default:
  711                         ; "no")
  712 ;media_encryption_optimistic=no ; Use encryption if possible but don't fail the call
  713                                 ; if not possible.
  714 ;g726_non_standard=no   ; When set to "yes" and an endpoint negotiates g.726
  715                         ; audio then g.726 for AAL2 packing order is used contrary
  716                         ; to what is recommended in RFC3551. Note, 'g726aal2' also
  717                         ; needs to be specified in the codec allow list
  718                         ; (default: "no")
  719 ;inband_progress=no     ; Determines whether chan_pjsip will indicate ringing
  720                         ; using inband progress (default: "no")
  721 ;call_group=    ; The numeric pickup groups for a channel (default: "")
  722 ;pickup_group=  ; The numeric pickup groups that a channel can pickup (default:
  723                 ; "")
  724 ;named_call_group=      ; The named pickup groups for a channel (default: "")
  725 ;named_pickup_group=    ; The named pickup groups that a channel can pickup
  726                         ; (default: "")
  727 ;device_state_busy_at=0 ; The number of in use channels which will cause busy
  728                         ; to be returned as device state (default: "0")
  729 ;t38_udptl=no   ; Whether T 38 UDPTL support is enabled or not (default: "no")
  730 ;t38_udptl_ec=none      ; T 38 UDPTL error correction method (default: "none")
  731 ;t38_udptl_maxdatagram=0        ; T 38 UDPTL maximum datagram size (default:
  732                                 ; "0")
  733 ;fax_detect=no  ; Whether CNG tone detection is enabled (default: "no")
  734 ;fax_detect_timeout=30  ; How many seconds into a call before fax_detect is
  735                         ; disabled for the call.
  736                         ; Zero disables the timeout.
  737                         ; (default: "0")
  738 ;t38_udptl_nat=no       ; Whether NAT support is enabled on UDPTL sessions
  739                         ; (default: "no")
  740 ;t38_bind_rtp_to_media_address=     ; Bind the UDPTL session to the media_address.
  741                                     ; This causes all UDPTL packets to be sent from
  742                                     ; the specified address. (default: "no")
  743 ;tone_zone=     ; Set which country s indications to use for channels created
  744                 ; for this endpoint (default: "")
  745 ;language=      ; Set the default language to use for channels created for this
  746                 ; endpoint (default: "")
  747 ;one_touch_recording=no ; Determines whether one touch recording is allowed for
  748                         ; this endpoint (default: "no")
  749 ;record_on_feature=automixmon   ; The feature to enact when one touch recording
  750                                 ; is turned on (default: "automixmon")
  751 ;record_off_feature=automixmon  ; The feature to enact when one touch recording
  752                                 ; is turned off (default: "automixmon")
  753 ;rtp_engine=asterisk    ; Name of the RTP engine to use for channels created
  754                         ; for this endpoint (default: "asterisk")
  755 ;allow_transfer=yes     ; Determines whether SIP REFER transfers are allowed
  756                         ; for this endpoint (default: "yes")
  757 ;sdp_owner=-    ; String placed as the username portion of an SDP origin o line
  758                 ; (default: "-")
  759 ;sdp_session=Asterisk   ; String used for the SDP session s line (default:
  760                         ; "Asterisk")
  761 ;tos_audio=0    ; DSCP TOS bits for audio streams (default: "0")
  762 ;tos_video=0    ; DSCP TOS bits for video streams (default: "0")
  763 ;cos_audio=0    ; Priority for audio streams (default: "0")
  764 ;cos_video=0    ; Priority for video streams (default: "0")
  765 ;allow_subscribe=yes    ; Determines if endpoint is allowed to initiate
  766                         ; subscriptions with Asterisk (default: "yes")
  767 ;sub_min_expiry=0       ; The minimum allowed expiry time for subscriptions
  768                         ; initiated by the endpoint (default: "0")
  769 ;from_user=     ; Username to use in From header for requests to this endpoint
  770                 ; (default: "")
  771 ;mwi_from_user= ; Username to use in From header for unsolicited MWI NOTIFYs to
  772                 ; this endpoint (default: "")
  773 ;from_domain=   ; Domain to user in From header for requests to this endpoint
  774                 ; (default: "")
  775 ;dtls_verify=no ; Verify that the provided peer certificate is valid (default:
  776                 ; "no")
  777 ;dtls_rekey=0   ; Interval at which to renegotiate the TLS session and rekey
  778                 ; the SRTP session (default: "0")
  779 ;dtls_auto_generate_cert= ; Enable ephemeral DTLS certificate generation (default:
  780                           ; "no")
  781 ;dtls_cert_file=          ; Path to certificate file to present to peer (default:
  782                           ; "")
  783 ;dtls_private_key=        ; Path to private key for certificate file (default:
  784                           ; "")
  785 ;dtls_cipher=   ; Cipher to use for DTLS negotiation (default: "")
  786 ;dtls_ca_file=  ; Path to certificate authority certificate (default: "")
  787 ;dtls_ca_path=  ; Path to a directory containing certificate authority
  788                 ; certificates (default: "")
  789 ;dtls_setup=    ; Whether we are willing to accept connections connect to the
  790                 ; other party or both (default: "")
  791 ;dtls_fingerprint= ; Hash to use for the fingerprint placed into SDP
  792                    ; (default: "SHA-256")
  793 ;srtp_tag_32=no ; Determines whether 32 byte tags should be used instead of 80
  794                 ; byte tags (default: "no")
  795 ;set_var=       ; Variable set on a channel involving the endpoint. For multiple
  796                 ; channel variables specify multiple 'set_var'(s)
  797 ;rtp_keepalive= ; Interval, in seconds, between comfort noise RTP packets if
  798                 ; RTP is not flowing. This setting is useful for ensuring that
  799                 ; holes in NATs and firewalls are kept open throughout a call.
  800 ;rtp_timeout=      ; Hang up channel if RTP is not received for the specified
  801                    ; number of seconds when the channel is off hold (default:
  802                    ; "0" or not enabled)
  803 ;rtp_timeout_hold= ; Hang up channel if RTP is not received for the specified
  804                    ; number of seconds when the channel is on hold (default:
  805                    ; "0" or not enabled)
  806 ;contact_user= ; On outgoing requests, force the user portion of the Contact
  807                ; header to this value (default: "")
  808 ;incoming_call_offer_pref= ; Based on this setting, a joint list of
  809                            ; preferred codecs between those received in an
  810                            ; incoming SDP offer (remote), and those specified
  811                            ; in the endpoint's "allow" parameter (local)
  812                            ; is created and is passed to the Asterisk core.
  813                            ;
  814                            ; local - Include all codecs in the local list that
  815                            ; are also in the remote list preserving the local
  816                            ; order. (default).
  817                            ; local_first - Include only the first codec in the
  818                            ; local list that is also in the remote list.
  819                            ; remote - Include all codecs in the remote list that
  820                            ; are also in the local list preserving remote list
  821                            ; order.
  822                            ; remote_first - Include only the first codec in
  823                            ; the remote list that is also in the local list.
  824 ;outgoing_call_offer_pref= ; Based on this setting, a joint list of
  825                            ; preferred codecs between those received from the
  826                            ; Asterisk core (remote), and those specified in
  827                            ; the endpoint's "allow" parameter (local) is
  828                            ; created and is used to create the outgoing SDP
  829                            ; offer.
  830                            ;
  831                            ; local - Include all codecs in the local list that
  832                            ; are also in the remote list preserving the local
  833                            ; order.
  834                            ; local_merge - Include all codecs in the local list
  835                            ; preserving the local order.
  836                            ; local_first - Include only the first codec in the
  837                            ; local list.
  838                            ; remote - Include all codecs in the remote list that
  839                            ; are also in the local list preserving remote list
  840                            ; order.
  841                            ; remote_merge - Include all codecs in the local list
  842                            ; preserving the remote list order. (default)
  843                            ; remote_first - Include only the first codec in the
  844                            ; remote list that is also in the local list.
  845 ;preferred_codec_only=no   ; Respond to a SIP invite with the single most
  846                            ; preferred codec rather than advertising all joint
  847                            ; codec capabilities. This limits the other side's
  848                            ; codec choice to exactly what we prefer.
  849                            ; default is no.
  850                            ; NOTE: This option is deprecated in favor
  851                            ; of incoming_call_offer_pref.  Setting both
  852                            ; options is unsupported.
  853 ;asymmetric_rtp_codec= ; Allow the sending and receiving codec to differ and
  854                        ; not be automatically matched (default: "no")
  855 ;refer_blind_progress= ; Whether to notifies all the progress details on blind
  856                        ; transfer (default: "yes"). The value "no" is useful
  857                        ; for some SIP phones (Mitel/Aastra, Snom) which expect
  858                        ; a sip/frag "200 OK" after REFER has been accepted.
  859 ;notify_early_inuse_ringing = ; Whether to notifies dialog-info 'early'
  860                               ; on INUSE && RINGING state (default: "no").
  861                               ; The value "yes" is useful for some SIP phones
  862                               ; (Cisco SPA) to be able to indicate and pick up
  863                               ; ringing devices.
  864 ;max_audio_streams= ; The maximum number of allowed negotiated audio streams
  865                     ; (default: 1)
  866 ;max_video_streams= ; The maximum number of allowed negotiated video streams
  867                     ; (default: 1)
  868 ;webrtc= ; When set to "yes" this also enables the following values that are needed
  869          ; for webrtc: rtcp_mux, use_avpf, ice_support, and use_received_transport.
  870          ; The following configuration settings also get defaulted as follows:
  871          ;     media_encryption=dtls
  872          ;     dtls_verify=fingerprint
  873          ;     dtls_setup=actpass
  874          ; A dtls_cert_file and a dtls_ca_file still need to be specified.
  875          ; Default for this option is "no"
  876 ;incoming_mwi_mailbox = ; Mailbox name to use when incoming MWI NOTIFYs are
  877                         ; received.
  878                         ; If an MWI NOTIFY is received FROM this endpoint,
  879                         ; this mailbox will be used when notifying other modules
  880                         ; of MWI status changes.  If not set, incoming MWI
  881                         ; NOTIFYs are ignored.
  882 ;follow_early_media_fork = ; On outgoing calls, if the UAS responds with
  883                            ; different SDP attributes on subsequent 18X or 2XX
  884                            ; responses (such as a port update) AND the To tag
  885                            ; on the subsequent response is different than that
  886                            ; on the previous one, follow it.  This usually
  887                            ; happens when the INVITE is forked to multiple UASs
  888                            ; and more than 1 sends an SDP answer.
  889                            ; This option must also be enabled in the system
  890                            ; section.
  891                            ; (default: yes)
  892 ;accept_multiple_sdp_answers =
  893                            ; On outgoing calls, if the UAS responds with
  894                            ; different SDP attributes on non-100rel 18X or 2XX
  895                            ; responses (such as a port update) AND the To tag on
  896                            ; the subsequent response is the same as that on the
  897                            ; previous one, process it. This can happen when the
  898                            ; UAS needs to change ports for some reason such as
  899                            ; using a separate port for custom ringback.
  900                            ; This option must also be enabled in the system
  901                            ; section.
  902                            ; (default: no)
  903 ;suppress_q850_reason_headers =
  904                            ; Suppress Q.850 Reason headers for this endpoint.
  905                            ; Some devices can't accept multiple Reason headers
  906                            ; and get confused when both 'SIP' and 'Q.850' Reason
  907                            ; headers are received.  This option allows the
  908                            ; 'Q.850' Reason header to be suppressed.
  909                            ; (default: no)
  910 ;ignore_183_without_sdp =
  911                            ; Do not forward 183 when it doesn't contain SDP.
  912                            ; Certain SS7 internetworking scenarios can result in
  913                            ; a 183 to be generated for reasons other than early
  914                            ; media.  Forwarding this 183 can cause loss of
  915                            ; ringback tone.  This flag emulates the behavior of
  916                            ; chan_sip and prevents these 183 responses from
  917                            ; being forwarded.
  918                            ; (default: no)
  919 ;stir_shaken =
  920                            ; If this is enabled, STIR/SHAKEN operations will be
  921                            ; performed on this endpoint. This includes inbound
  922                            ; and outbound INVITEs. On an inbound INVITE, Asterisk
  923                            ; will check for an Identity header and attempt to
  924                            ; verify the call. On an outbound INVITE, Asterisk will
  925                            ; add an Identity header that others can use to verify
  926                            ; calls from this endpoint. Additional configuration is
  927                            ; done in stir_shaken.conf.
  928                            ; The STIR_SHAKEN dialplan function must be used to get
  929                            ; the verification results on inbound INVITEs. Nothing
  930                            ; happens to the call if verification fails; it's up to
  931                            ; you to determine what to do with the results.
  932                            ; (default: no)
  933 ;allow_unauthenticated_options =
  934                            ; By default, chan_pjsip will challenge an incoming
  935                            ; OPTIONS request for authentication credentials just
  936                            ; as it would an INVITE request. This is consistent
  937                            ; with RFC 3261.
  938                            ; There are many UAs that use an OPTIONS request as a
  939                            ; "ping" and they expect a 200 response indicating that
  940                            ; the remote party is up and running without a need to
  941                            ; authenticate.
  942                            ; Setting allow_unauthenticated_options to 'yes' will
  943                            ; instruct chan_pjsip to skip the authentication step
  944                            ; when it receives an OPTIONS request for this
  945                            ; endpoint.
  946                            ; There are security implications to enabling this
  947                            ; setting as it can allow information disclosure to
  948                            ; occur - specifically, if enabled, an external party
  949                            ; could enumerate and find the endpoint name by
  950                            ; sending OPTIONS requests and examining the
  951                            ; responses.
  952                            ; (default: no)
  953 
  954 ;==========================AUTH SECTION OPTIONS=========================
  955 ;[auth]
  956 ;  SYNOPSIS: Authentication type
  957 ;
  958 ;  Note: Using the same auth section for inbound and outbound
  959 ;  authentication is not recommended.  There is a difference in
  960 ;  meaning for an empty realm setting between inbound and outbound
  961 ;  authentication uses.  Look to the CLI config help
  962 ;  "config show help res_pjsip auth realm" or on the wiki for the
  963 ;  difference.
  964 ;
  965 ;auth_type=userpass  ; Authentication type.  May be
  966                      ; "userpass" for plain text passwords or
  967                      ; "md5" for pre-hashed credentials.
  968                      ; (default: "userpass")
  969 ;nonce_lifetime=32   ; Lifetime of a nonce associated with this
  970                      ; authentication config (default: "32")
  971 ;md5_cred=     ; As an alternative to specifying a plain text password,
  972                ; you can hash the username, realm and password
  973                ; together one time and place the hash value here.
  974                ; The input to the hash function must be in the
  975                ; following format:
  976                ; <username>:<realm>:<password>
  977                ; For incoming authentication (asterisk is the UAS),
  978                ; the realm must match either the realm set in this object
  979                ; or the default set in in the "global" object.
  980                ;
  981                ; For outgoing authentication (asterisk is the UAC),
  982                ; the realm must match what the server will be sending
  983                ; in their WWW-Authenticate header.  It can't be blank
  984                ; unless you expect the server to be sending a blank
  985                ; realm in the header.
  986                ; You can generate the hash with the following shell
  987                ; command:
  988                ; $ echo -n "myname:myrealm:mypassword" | md5sum
  989                ; Note the '-n'.  You don't want a newline to be part
  990                ; of the hash.  (default: "")
  991 ;password=     ; PlainText password used for authentication (default: "")
  992 ;realm=        ; For incoming authentication (asterisk is the UAS),
  993                ; this is the realm to be sent on WWW-Authenticate
  994                ; headers.  If not specified, the global object's
  995                ; "default_realm" will be used.
  996                ;
  997                ; For outgoing authentication (asterisk is the UAS), this
  998                ; must either be the realm the server is expected to send,
  999                ; or left blank or contain a single '*' to automatically
 1000                ; use the realm sent by the server. If you have multiple
 1001                ; auth object for an endpoint, the realm is also used to
 1002                ; match the auth object to the realm the server sent.
 1003                ; Using the same auth section for inbound and outbound
 1004                ; authentication is not recommended.  There is a difference in
 1005                ; meaning for an empty realm setting between inbound and outbound
 1006                ; authentication uses.
 1007                ; (default: "")
 1008 ;type=         ; Must be auth (default: "")
 1009 ;username=     ; Username to use for account (default: "")
 1010 
 1011 
 1012 ;==========================DOMAIN_ALIAS SECTION OPTIONS=========================
 1013 ;[domain_alias]
 1014 ;  SYNOPSIS: Domain Alias
 1015 ;type=  ; Must be of type domain_alias (default: "")
 1016 ;domain=        ; Domain to be aliased (default: "")
 1017 
 1018 
 1019 ;==========================TRANSPORT SECTION OPTIONS=========================
 1020 ;[transport]
 1021 ;  SYNOPSIS: SIP Transport
 1022 ;
 1023 ;async_operations=1     ; Number of simultaneous Asynchronous Operations
 1024                         ; (default: "1")
 1025 ;bind=  ; IP Address and optional port to bind to for this transport (default:
 1026         ; "")
 1027 ; Note that for the Websocket transport the TLS configuration is configured
 1028 ; in http.conf and is applied for all HTTPS traffic.
 1029 ;ca_list_file=  ; File containing a list of certificates to read TLS ONLY
 1030                 ; (default: "")
 1031 ;ca_list_path=  ; Path to directory containing certificates to read TLS ONLY.
 1032                 ; PJProject version 2.4 or higher is required for this option to
 1033                 ; be used.
 1034                 ; (default: "")
 1035 ;cert_file=     ; Certificate file for endpoint TLS ONLY
 1036                 ; Will read .crt or .pem file but only uses cert,
 1037                 ; a .key file must be specified via priv_key_file.
 1038                 ; Since PJProject version 2.5: If the file name ends in _rsa,
 1039                 ; for example "asterisk_rsa.pem", the files "asterisk_dsa.pem"
 1040                 ; and/or "asterisk_ecc.pem" are loaded (certificate, inter-
 1041                 ; mediates, private key), to support multiple algorithms for
 1042                 ; server authentication (RSA, DSA, ECDSA). If the chains are
 1043                 ; different, at least OpenSSL 1.0.2 is required.
 1044                 ; (default: "")
 1045 ;cipher=        ; Preferred cryptography cipher names TLS ONLY (default: "")
 1046 ;method=        ; Method of SSL transport TLS ONLY (default: "")
 1047 ;priv_key_file= ; Private key file TLS ONLY (default: "")
 1048 ;verify_client= ; Require verification of client certificate TLS ONLY (default:
 1049                 ; "")
 1050 ;verify_server= ; Require verification of server certificate TLS ONLY (default:
 1051                 ; "")
 1052 ;require_client_cert=   ; Require client certificate TLS ONLY (default: "")
 1053 ;domain=        ; Domain the transport comes from (default: "")
 1054 ;external_media_address=        ; External IP address to use in RTP handling
 1055                                 ; (default: "")
 1056 ;external_signaling_address=    ; External address for SIP signalling (default:
 1057                                 ; "")
 1058 ;external_signaling_port=0      ; External port for SIP signalling (default:
 1059                                 ; "0")
 1060 ;local_net=     ; Network to consider local used for NAT purposes (default: "")
 1061 ;password=      ; Password required for transport (default: "")
 1062 ;protocol=udp   ; Protocol to use for SIP traffic (default: "udp")
 1063 ;type=  ; Must be of type transport (default: "")
 1064 ;tos=0  ; Enable TOS for the signalling sent over this transport (default: "0")
 1065 ;cos=0  ; Enable COS for the signalling sent over this transport (default: "0")
 1066 ;websocket_write_timeout=100    ; Default write timeout to set on websocket
 1067                                 ; transports. This value may need to be adjusted
 1068                                 ; for connections where Asterisk must write a
 1069                                 ; substantial amount of data and the receiving
 1070                                 ; clients are slow to process the received
 1071                                 ; information. Value is in milliseconds; default
 1072                                 ; is 100 ms.
 1073 ;allow_reload=no    ; Although transports can now be reloaded, that may not be
 1074                     ; desirable because of the slight possibility of dropped
 1075                     ; calls. To make sure there are no unintentional drops, if
 1076                     ; this option is set to 'no' (the default) changes to the
 1077                     ; particular transport will be ignored. If set to 'yes',
 1078                     ; changes (if any) will be applied.
 1079 ;symmetric_transport=no ; When a request from a dynamic contact comes in on a
 1080                         ; transport with this option set to 'yes', the transport
 1081                         ; name will be saved and used for subsequent outgoing
 1082                         ; requests like OPTIONS, NOTIFY and INVITE.  It's saved
 1083                         ; as a contact uri parameter named 'x-ast-txp' and will
 1084                         ; display with the contact uri in CLI, AMI, and ARI
 1085                         ; output.  On the outgoing request, if a transport
 1086                         ; wasn't explicitly set on the endpoint AND the request
 1087                         ; URI is not a hostname, the saved transport will be
 1088                         ; used and the 'x-ast-txp' parameter stripped from the
 1089                         ; outgoing packet.
 1090 
 1091 ;==========================AOR SECTION OPTIONS=========================
 1092 ;[aor]
 1093 ;  SYNOPSIS: The configuration for a location of an endpoint
 1094 ;contact=       ; Permanent contacts assigned to AoR (default: "")
 1095 ;default_expiration=3600        ; Default expiration time in seconds for
 1096                                 ; contacts that are dynamically bound to an AoR
 1097                                 ; (default: "3600")
 1098 ;mailboxes=           ; Allow subscriptions for the specified mailbox(es)
 1099                       ; This option applies when an external entity subscribes to an AoR
 1100                       ; for Message Waiting Indications. (default: "")
 1101 ;voicemail_extension= ; The voicemail extension to send in the NOTIFY Message-Account header
 1102                       ; (default: global/default_voicemail_extension)
 1103 ;maximum_expiration=7200        ; Maximum time to keep an AoR (default: "7200")
 1104 ;max_contacts=0 ; Maximum number of contacts that can bind to an AoR (default:
 1105                 ; "0")
 1106 ;minimum_expiration=60  ; Minimum keep alive time for an AoR (default: "60")
 1107 ;remove_existing=no     ; Allow a registration to succeed by displacing any existing
 1108                         ; contacts that now exceed the max_contacts count.  Any
 1109                         ; removed contacts are the next to expire.  The behaviour is
 1110                         ; beneficial when rewrite_contact is enabled and max_contacts
 1111                         ; is greater than one.  The removed contact is likely the old
 1112                         ; contact created by rewrite_contact that the device is
 1113                         ; refreshing.
 1114                         ; (default: "no")
 1115 ;remove_unavailable=no  ; If remove_existing is disabled, will allow a registration
 1116                         ; to succeed by removing only unavailable contacts when
 1117                         ; max_contacts is exceeded. This will reject a registration
 1118                         ; that exceeds max_contacts if no unavailable contacts are
 1119                         ; present to remove. If remove_existing is enabled, will
 1120                         ; prioritize removal of unavailable contacts before removing
 1121                         ; expiring soonest.  This tames the behavior of remove_existing
 1122                         ; to only remove an available contact if an unavailable one is
 1123                         ; not present.
 1124                         ; (default: "no")
 1125 ;type=  ; Must be of type aor (default: "")
 1126 ;qualify_frequency=0    ; Interval at which to qualify an AoR via OPTIONS requests.
 1127                         ; (default: "0")
 1128 ;qualify_timeout=3.0      ; Qualify timeout in fractional seconds (default: "3.0")
 1129 ;authenticate_qualify=no        ; Authenticates a qualify request if needed
 1130                                 ; (default: "no")
 1131 ;outbound_proxy=        ; Proxy through which to send OPTIONS requests, a full SIP URI
 1132                         ; must be provided (default: "")
 1133 
 1134 
 1135 ;==========================SYSTEM SECTION OPTIONS=========================
 1136 ;[system]
 1137 ;  SYNOPSIS: Options that apply to the SIP stack as well as other system-wide settings
 1138 ;timer_t1=500   ; Set transaction timer T1 value milliseconds (default: "500")
 1139 ;timer_b=32000  ; Set transaction timer B value milliseconds (default: "32000")
 1140 ;compact_headers=no     ; Use the short forms of common SIP header names
 1141                         ; (default: "no")
 1142 ;threadpool_initial_size=0      ; Initial number of threads in the res_pjsip
 1143                                 ; threadpool (default: "0")
 1144 ;threadpool_auto_increment=5    ; The amount by which the number of threads is
 1145                                 ; incremented when necessary (default: "5")
 1146 ;threadpool_idle_timeout=60     ; Number of seconds before an idle thread
 1147                                 ; should be disposed of (default: "60")
 1148 ;threadpool_max_size=0  ; Maximum number of threads in the res_pjsip threadpool
 1149                         ; A value of 0 indicates no maximum (default: "0")
 1150 ;disable_tcp_switch=yes ; Disable automatic switching from UDP to TCP transports
 1151                         ; if outgoing request is too large.
 1152                         ; See RFC 3261 section 18.1.1.
 1153                         ; Disabling this option has been known to cause interoperability
 1154                         ; issues, so disable at your own risk.
 1155                         ; (default: "yes")
 1156 ;follow_early_media_fork = ; On outgoing calls, if the UAS responds with
 1157                            ; different SDP attributes on subsequent 18X or 2XX
 1158                            ; responses (such as a port update) AND the To tag
 1159                            ; on the subsequent response is different than that
 1160                            ; on the previous one, follow it.  This usually
 1161                            ; happens when the INVITE is forked to multiple UASs
 1162                            ; and more than 1 sends an SDP answer.
 1163                            ; This option must also be enabled on endpoints that
 1164                            ; require this functionality.
 1165                            ; (default: yes)
 1166 ;accept_multiple_sdp_answers =
 1167                            ; On outgoing calls, if the UAS responds with
 1168                            ; different SDP attributes on non-100rel 18X or 2XX
 1169                            ; responses (such as a port update) AND the To tag on
 1170                            ; the subsequent response is the same as that on the
 1171                            ; previous one, process it. This can happen when the
 1172                            ; UAS needs to change ports for some reason such as
 1173                            ; using a separate port for custom ringback.
 1174                            ; This option must also be enabled on endpoints that
 1175                            ; require this functionality.
 1176                            ; (default: no)
 1177 ;disable_rport=no ; Disable the use of "rport" in outgoing requests.
 1178 ;type=  ; Must be of type system (default: "")
 1179 
 1180 ;==========================GLOBAL SECTION OPTIONS=========================
 1181 ;[global]
 1182 ;  SYNOPSIS: Options that apply globally to all SIP communications
 1183 ;max_forwards=70        ; Value used in Max Forwards header for SIP requests
 1184                         ; (default: "70")
 1185 ;type=  ; Must be of type global (default: "")
 1186 ;user_agent=Asterisk PBX        ; Allows you to change the user agent string
 1187                                 ; The default user agent string also contains
 1188                                 ; the Asterisk version. If you don't want to
 1189                                 ; expose this, change the user_agent string.
 1190 ;default_outbound_endpoint=default_outbound_endpoint    ; Endpoint to use when
 1191                                                         ; sending an outbound
 1192                                                         ; request to a URI
 1193                                                         ; without a specified
 1194                                                         ; endpoint (default: "d
 1195                                                         ; efault_outbound_endpo
 1196                                                         ; int")
 1197 ;debug=no ; Enable/Disable SIP debug logging.  Valid options include yes|no
 1198           ; or a host address (default: "no")
 1199 ;keep_alive_interval=90 ; The interval (in seconds) at which to send (double CRLF)
 1200                         ; keep-alives on all active connection-oriented transports;
 1201                         ; for connection-less like UDP see qualify_frequency.
 1202                         ; (default: "90")
 1203 ;contact_expiration_check_interval=30
 1204                         ; The interval (in seconds) to check for expired contacts.
 1205 ;disable_multi_domain=no
 1206             ; Disable Multi Domain support.
 1207             ; If disabled it can improve realtime performace by reducing
 1208             ; number of database requsts
 1209             ; (default: "no")
 1210 ;endpoint_identifier_order=ip,username,anonymous
 1211             ; The order by which endpoint identifiers are given priority.
 1212             ; Currently, "ip", "header", "username", "auth_username" and "anonymous"
 1213             ; are valid identifiers as registered by the res_pjsip_endpoint_identifier_*
 1214             ; modules.  Some modules like res_pjsip_endpoint_identifier_user register
 1215             ; more than one identifier.  Use the CLI command "pjsip show identifiers"
 1216             ; to see the identifiers currently available.
 1217             ; (default: ip,username,anonymous)
 1218 ;max_initial_qualify_time=4 ; The maximum amount of time (in seconds) from
 1219                             ; startup that qualifies should be attempted on all
 1220                             ; contacts.  If greater than the qualify_frequency
 1221                             ; for an aor, qualify_frequency will be used instead.
 1222 ;regcontext=sipregistrations  ; If regcontext is specified, Asterisk will dynamically
 1223                               ; create and destroy a NoOp priority 1 extension for a
 1224                               ; given endpoint who registers or unregisters with us.
 1225                               ; The extension added is the name of the endpoint.
 1226 ;default_voicemail_extension=asterisk
 1227                    ; The voicemail extension to send in the NOTIFY Message-Account header
 1228                    ; if not set on endpoint or aor.
 1229                    ; (default: "")
 1230 ;
 1231 ; The following unidentified_request options are only used when "auth_username"
 1232 ; matching is enabled in "endpoint_identifier_order".
 1233 ;
 1234 ;unidentified_request_count=5   ; The number of unidentified requests that can be
 1235                                 ; received from a single IP address in
 1236                                 ; unidentified_request_period seconds before a security
 1237                                 ; event is generated. (default: 5)
 1238 ;unidentified_request_period=5  ; See above.  (default: 5 seconds)
 1239 ;unidentified_request_prune_interval=30
 1240                                 ; The interval at which unidentified requests
 1241                                 ; are check to see if they can be pruned.  If they're
 1242                                 ; older than twice the unidentified_request_period,
 1243                                 ; they're pruned.
 1244 ;
 1245 ;default_from_user=asterisk     ; When Asterisk generates an outgoing SIP request, the
 1246                                 ; From header username will be set to this value if
 1247                                 ; there is no better option (such as CallerID or
 1248                                 ; endpoint/from_user) to be used
 1249 ;default_realm=asterisk         ; When Asterisk generates a challenge, the digest realm
 1250                                 ; will be set to this value if there is no better option
 1251                                 ; (such as auth/realm) to be used.
 1252 
 1253                     ; Asterisk Task Processor Queue Size
 1254                     ; On heavy loaded system with DB storage you may need to increase
 1255                     ; taskprocessor queue.
 1256                     ; If the taskprocessor queue size reached high water level,
 1257                     ; the alert is triggered.
 1258                     ; If the alert is set the pjsip distibutor stops processing incoming
 1259                     ; requests until the alert is cleared.
 1260                     ; The alert is cleared when taskprocessor queue size drops to the
 1261                     ; low water clear level.
 1262                     ; The next options set taskprocessor queue levels for MWI.
 1263 ;mwi_tps_queue_high=500 ; Taskprocessor high water alert trigger level.
 1264 ;mwi_tps_queue_low=450  ; Taskprocessor low water clear alert level.
 1265                     ; The default is -1 for 90% of high water level.
 1266 
 1267                     ; Unsolicited MWI
 1268                     ; If there are endpoints configured with unsolicited MWI
 1269                     ; then res_pjsip_mwi module tries to send MWI to all endpoints on startup.
 1270 ;mwi_disable_initial_unsolicited=no ; Disable sending unsolicited mwi to all endpoints on startup.
 1271                     ; If disabled then unsolicited mwi will start processing
 1272                     ; on the endpoint's next contact update.
 1273 
 1274 ;ignore_uri_user_options=no ; Enable/Disable ignoring SIP URI user field options.
 1275                     ; If you have this option enabled and there are semicolons
 1276                     ; in the user field of a SIP URI then the field is truncated
 1277                     ; at the first semicolon.  This effectively makes the semicolon
 1278                     ; a non-usable character for PJSIP endpoint names, extensions,
 1279                     ; and AORs.  This can be useful for improving compatability with
 1280                     ; an ITSP that likes to use user options for whatever reason.
 1281                     ; Example:
 1282                     ; URI: "sip:1235557890;phone-context=national@x.x.x.x;user=phone"
 1283                     ; The user field is "1235557890;phone-context=national"
 1284                     ; Which becomes this: "1235557890"
 1285                     ;
 1286                     ; Note: The caller-id and redirecting number strings obtained
 1287                     ; from incoming SIP URI user fields are always truncated at the
 1288                     ; first semicolon.
 1289 
 1290 ;send_contact_status_on_update_registration=no ; Enable sending AMI ContactStatus
 1291                     ; event when a device refreshes its registration
 1292                     ; (default: "no")
 1293 
 1294 ;taskprocessor_overload_trigger=global
 1295                 ; Set the trigger the distributor will use to detect
 1296                 ; taskprocessor overloads.  When triggered, the distributor
 1297                 ; will not accept any new requests until the overload has
 1298                 ; cleared.
 1299                 ; "global": (default) Any taskprocessor overload will trigger.
 1300                 ; "pjsip_only": Only pjsip taskprocessor overloads will trigger.
 1301                 ; "none":  No overload detection will be performed.
 1302                 ; WARNING: The "none" and "pjsip_only" options should be used
 1303                 ; with extreme caution and only to mitigate specific issues.
 1304                 ; Under certain conditions they could make things worse.
 1305 
 1306 ;norefersub=yes     ; Enable sending norefersub option tag in Supported header to advertise
 1307                     ; that the User Agent is capable of accepting a REFER request with
 1308                     ; creating an implicit subscription (see RFC 4488).
 1309                     ; (default: "yes")
 1310 
 1311 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_acl
 1312 ;==========================ACL SECTION OPTIONS=========================
 1313 ;[acl]
 1314 ;  SYNOPSIS: Access Control List
 1315 ;acl=   ; List of IP ACL section names in acl conf (default: "")
 1316 ;contact_acl=   ; List of Contact ACL section names in acl conf (default: "")
 1317 ;contact_deny=  ; List of Contact header addresses to deny (default: "")
 1318 ;contact_permit=        ; List of Contact header addresses to permit (default:
 1319                         ; "")
 1320 ;deny=  ; List of IP addresses to deny access from (default: "")
 1321 ;permit=        ; List of IP addresses to permit access from (default: "")
 1322 ;type=  ; Must be of type acl (default: "")
 1323 
 1324 
 1325 
 1326 
 1327 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_outbound_registration
 1328 ;==========================REGISTRATION SECTION OPTIONS=========================
 1329 ;[registration]
 1330 ;  SYNOPSIS: The configuration for outbound registration
 1331 ;auth_rejection_permanent=yes   ; Determines whether failed authentication
 1332                                 ; challenges are treated as permanent failures
 1333                                 ; (default: "yes")
 1334 ;client_uri=    ; Client SIP URI used when attemping outbound registration
 1335                 ; (default: "")
 1336 ;contact_user=  ; Contact User to use in request (default: "")
 1337 ;expiration=3600        ; Expiration time for registrations in seconds
 1338                         ; (default: "3600")
 1339 ;max_retries=10 ; Maximum number of registration attempts (default: "10")
 1340 ;outbound_auth= ; Authentication object to be used for outbound registrations
 1341                 ; (default: "")
 1342 ;outbound_proxy=        ; Proxy through which to send registrations, a full SIP URI
 1343                         ; must be provided (default: "")
 1344 ;retry_interval=60      ; Interval in seconds between retries if outbound
 1345                         ; registration is unsuccessful (default: "60")
 1346 ;forbidden_retry_interval=0     ; Interval used when receiving a 403 Forbidden
 1347                                 ; response (default: "0")
 1348 ;fatal_retry_interval=0 ; Interval used when receiving a fatal response.
 1349                         ; (default: "0") A fatal response is any permanent
 1350                         ; failure (non-temporary 4xx, 5xx, 6xx) response
 1351                         ; received from the registrar. NOTE - if also set
 1352                         ; the 'forbidden_retry_interval' takes precedence
 1353                         ; over this one when a 403 is received. Also, if
 1354                         ; 'auth_rejection_permanent' equals 'yes' a 401 and
 1355                         ; 407 become subject to this retry interval.
 1356 ;server_uri=    ; SIP URI of the server to register against (default: "")
 1357 ;transport=     ; Transport used for outbound authentication (default: "")
 1358 ;line=          ; When enabled this option will cause a 'line' parameter to be
 1359                 ; added to the Contact header placed into the outgoing
 1360                 ; registration request. If the remote server sends a call
 1361                 ; this line parameter will be used to establish a relationship
 1362                 ; to the outbound registration, ultimately causing the
 1363                 ; configured endpoint to be used (default: "no")
 1364 ;endpoint=      ; When line support is enabled this configured endpoint name
 1365                 ; is used for incoming calls that are related to the outbound
 1366                 ; registration (default: "")
 1367 ;type=  ; Must be of type registration (default: "")
 1368 
 1369 
 1370 
 1371 
 1372 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_endpoint_identifier_ip
 1373 ;==========================IDENTIFY SECTION OPTIONS=========================
 1374 ;[identify]
 1375 ;  SYNOPSIS: Identifies endpoints via some criteria.
 1376 ;
 1377 ; NOTE: If multiple matching criteria are provided then an inbound request will
 1378 ; be matched to the endpoint if it matches ANY of the criteria.
 1379 ;endpoint=      ; Name of endpoint identified (default: "")
 1380 ;srv_lookups=yes        ; Perform SRV lookups for provided hostnames. (default: yes)
 1381 ;match= ; Comma separated list of IP addresses, networks, or hostnames to match
 1382         ; against (default: "")
 1383 ;match_header= ; SIP header with specified value to match against (default: "")
 1384 ;type=  ; Must be of type identify (default: "")
 1385 
 1386 
 1387 
 1388 
 1389 ;========================PHONEPROV_USER SECTION OPTIONS=======================
 1390 ;[phoneprov]
 1391 ;  SYNOPSIS: Contains variables for autoprovisioning each user
 1392 ;endpoint=      ; The endpoint from which to gather username, secret, etc. (default: "")
 1393 ;PROFILE=       ; The name of a profile configured in phoneprov.conf (default: "")
 1394 ;MAC=           ; The mac address for this user (default: "")
 1395 ;OTHERVAR=      ; Any other name value pair to be used in templates (default: "")
 1396                 ; Common variables include LINE, LINEKEYS, etc.
 1397                 ; See phoneprov.conf.sample for others.
 1398 ;type=          ; Must be of type phoneprov (default: "")
 1399 
 1400 
 1401 
 1402 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_outbound_publish
 1403 ;======================OUTBOUND_PUBLISH SECTION OPTIONS=====================
 1404 ; See https://wiki.asterisk.org/wiki/display/AST/Publishing+Extension+State
 1405 ; for more information.
 1406 ;[outbound-publish]
 1407 ;type=outbound-publish     ; Must be of type 'outbound-publish'.
 1408 
 1409 ;expiration=3600           ; Expiration time for publications in seconds
 1410 
 1411 ;outbound_auth=            ; Authentication object(s) to be used for outbound
 1412                            ; publishes.
 1413                            ; This is a comma-delimited list of auth sections
 1414                            ; defined in pjsip.conf used to respond to outbound
 1415                            ; authentication challenges.
 1416                            ; Using the same auth section for inbound and
 1417                            ; outbound authentication is not recommended.  There
 1418                            ; is a difference in meaning for an empty realm
 1419                            ; setting between inbound and outbound authentication
 1420                            ; uses. See the auth realm description for details.
 1421 
 1422 ;outbound_proxy=           ; SIP URI of the outbound proxy used to send
 1423                            ; publishes
 1424 
 1425 ;server_uri=               ; SIP URI of the server and entity to publish to.
 1426                            ; This is the URI at which to find the entity and
 1427                            ; server to send the outbound PUBLISH to.
 1428                            ; This URI is used as the request URI of the outbound
 1429                            ; PUBLISH request from Asterisk.
 1430 
 1431 ;from_uri=                 ; SIP URI to use in the From header.
 1432                            ; This is the URI that will be placed into the From
 1433                            ; header of outgoing PUBLISH messages. If no URI is
 1434                            ; specified then the URI provided in server_uri will
 1435                            ; be used.
 1436 
 1437 ;to_uri=                   ; SIP URI to use in the To header.
 1438                            ; This is the URI that will be placed into the To
 1439                            ; header of outgoing PUBLISH messages. If no URI is
 1440                            ; specified then the URI provided in server_uri will
 1441                            ; be used.
 1442 
 1443 ;event=                    ; Event type of the PUBLISH.
 1444 
 1445 ;max_auth_attempts=        ; Maximum number of authentication attempts before
 1446                            ; stopping the pub.
 1447 
 1448 ;transport=                ; Transport used for outbound publish.
 1449                            ; A transport configured in pjsip.conf. As with other
 1450                            ; res_pjsip modules, this will use the first
 1451                            ; available transport of the appropriate type if
 1452                            ; unconfigured.
 1453 
 1454 ;multi_user=no             ; Enable multi-user support (Asterisk 14+ only)
 1455 
 1456 
 1457 
 1458 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_pubsub
 1459 ;=============================RESOURCE-LIST===================================
 1460 ; See https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30278158
 1461 ; for more information.
 1462 ;[resource_list]
 1463 ;type=resource_list        ; Must be of type 'resource_list'.
 1464 
 1465 ;event=                    ; The SIP event package that the list resource.
 1466                            ; belongs to.  The SIP event package describes the
 1467                            ; types of resources that Asterisk reports the state
 1468                            ; of.
 1469 
 1470 ;list_item=                ; The name of a resource to report state on.
 1471                            ; In general Asterisk looks up list items in the
 1472                            ; following way:
 1473                            ;  1. Check if the list item refers to another
 1474                            ;     configured resource list.
 1475                            ;  2. Pass the name of the resource off to
 1476                            ;     event-package-specific handlers to find the
 1477                            ;     specified resource.
 1478                            ; The second part means that the way the list item
 1479                            ; is specified depends on what type of list this is.
 1480                            ; For instance, if you have the event set to
 1481                            ; presence, then list items should be in the form of
 1482                            ; dialplan_extension@dialplan_context. For
 1483                            ; message-summary, mailbox names should be listed.
 1484 
 1485 ;full_state=no             ; Indicates if the entire list's state should be
 1486                            ; sent out.
 1487                            ; If this option is enabled, and a resource changes
 1488                            ; state, then Asterisk will construct a notification
 1489                            ; that contains the state of all resources in the
 1490                            ; list. If the option is disabled, Asterisk will
 1491                            ; construct a notification that only contains the
 1492                            ; states of resources that have changed.
 1493                            ; NOTE: Even with this option disabled, there are
 1494                            ; certain situations where Asterisk is forced to send
 1495                            ; a notification with the states of all resources in
 1496                            ; the list. When a subscriber renews or terminates
 1497                            ; its subscription to the list, Asterisk MUST send
 1498                            ; a full state notification.
 1499 
 1500 ;notification_batch_interval=0
 1501                            ; Time Asterisk should wait, in milliseconds,
 1502                            ; before sending notifications.
 1503 
 1504 ;==========================INBOUND_PUBLICATION================================
 1505 ; See https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP
 1506 ; for more information.
 1507 ;[inbound-publication]
 1508 ;type=                     ; Must be of type 'inbound-publication'.
 1509 
 1510 ;endpoint=                 ; Optional name of an endpoint that is only allowed
 1511                            ; to publish to this resource.
 1512 
 1513 
 1514 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_publish_asterisk
 1515 ;==========================ASTERISK_PUBLICATION===============================
 1516 ; See https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP
 1517 ; for more information.
 1518 ;[asterisk-publication]
 1519 ;type=asterisk-publication ; Must be of type 'asterisk-publication'.
 1520 
 1521 ;devicestate_publish=      ; Optional name of a publish item that can be used
 1522                            ; to publish a req.
 1523 
 1524 ;mailboxstate_publish=     ; Optional name of a publish item that can be used
 1525                            ; to publish a req.
 1526 
 1527 ;device_state=no           ; Whether we should permit incoming device state
 1528                            ; events.
 1529 
 1530 ;device_state_filter=      ; Optional regular expression used to filter what
 1531                            ; devices we accept events for.
 1532 
 1533 ;mailbox_state=no          ; Whether we should permit incoming mailbox state
 1534                            ; events.
 1535 
 1536 ;mailbox_state_filter=     ; Optional regular expression used to filter what
 1537                            ; mailboxes we accept events for.