asterisk/configs/samples/pjsip.conf.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.

; 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
; * 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=".
;
; IPv6: For endpoints using IPv6, remember to set "rtp_ipv6=yes" so that the RTP
; engine will also be able to bind to an IPv6 address.
;
; 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 defined transport
; in the configuration file.
;
; 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.
;
; Use the "rtp_ipv6=yes" option if you want to utilize RTP over an ipv6 transport.
;
; 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
;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

;===============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: "")
;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")
;connected_line_method=invite   ; Connected line method type (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   ; Way s for Endpoint to be identified (default:
                        ; "username")
;redirect_method=user   ; How redirects received from an endpoint are handled
                        ; (default: "user")
;mailboxes=     ; Mailbox es to be associated with (default: "")
;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_ipv6=no    ; Allow use of IPv6 for RTP traffic (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=     ; Desired transport configuration (default: "")
;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")
;t38_udptl_nat=no       ; Whether NAT support is enabled on UDPTL sessions
                        ; (default: "no")
;t38_udptl_ipv6=no      ; Whether IPv6 is used for 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_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)

;==========================AUTH SECTION OPTIONS=========================
;[auth]
;  SYNOPSIS: Authentication type
;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:
        ; "")
;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: "")
;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")
;method=        ; Method of SSL transport TLS ONLY (default: "")
;local_net=     ; Network to consider local used for NAT purposes (default: "")
;password=      ; Password required for transport (default: "")
;priv_key_file= ; Private key file TLS ONLY (default: "")
;protocol=udp   ; Protocol to use for SIP traffic (default: "udp")
;require_client_cert=   ; Require client certificate TLS ONLY (default: "")
;type=  ; Must be of type transport (default: "")
;verify_client= ; Require verification of client certificate TLS ONLY (default:
                ; "")
;verify_server= ; Require verification of server certificate TLS ONLY (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.

;==========================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=     ; Mailbox es to be associated with (default: "")
;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     ; Determines whether new contacts replace existing ones
                        ; (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=        ; Outbound proxy used when sending OPTIONS request
                        ; (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")
;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 SVN-branch-12-r404375  ; Value used in User Agent
                                                ; header for SIP requests and
                                                ; Server header for SIP
                                                ; responses (default: "Asterisk
                                                ; PBX SVN-branch-12-r404375")
;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")
;endpoint_identifier_order=ip,username,anonymous
            ; The order by which endpoint identifiers are given priority.
            ; Identifier names are derived from res_pjsip_endpoint_identifier_*
            ; modules. (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.
; 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.
;regcontext=sipregistrations

; 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=        ; Outbound Proxy used to send registrations (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: "")
;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 source IP address
;endpoint=      ; Name of Endpoint (default: "")
;match= ; IP addresses or networks 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: "")