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

; The following sections show example configurations for various scenarios.
; Most require a couple or more configuration types configured in concert.

;===============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 "localnet=".
;
; 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
;localnet=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
;privkey_file=/path/mykey.key
;cipher=ALL
;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'.

;[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

;[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.
;
; Section Naming
;
; Sections can have the same name as long as their "type="
; options are set to different values.

;[mytrunk]
;type=endpoint
;transport=transport-udp
;context=from-external
;disallow=all
;allow=ulaw
;outbound_auth=mytrunk
;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=yes"
; 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=yes
;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
;mwifromuser=6001
;
; Extension and Device state options
;
;devicestate_busy_at=1
;allowsubscribe=yes
;subminexpiry=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


;============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
;contactacl=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
;contactdeny=0.0.0.0/0.0.0.0
;contactpermit=209.16.236.0/24
;contactpermit=209.16.236.1
;contactpermit=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
;contactdeny=0.0.0.0/0.0.0.0
;contactpermit=209.16.236.0
;permit=209.16.236.0/24, !209.16.236.12/32



; 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=      ; Default privacy level (default: "")
;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")
;external_media_address=        ; IP used for External Media handling (default:
                                ; "")
;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")
;mailboxes=     ; Mailbox es to be associated with (default: "")
;moh_suggest=default     ; Default Music On Hold class (default: "default")
;outbound_auth= ; Authentication object used for outbound requests (default:
                ; "")
;outbound_proxy=        ; Proxy through which to send requests (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")
;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")
;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: "")
;fromdomain=    ; Domain to user in From header for requests to this endpoint
                ; (default: "")
;dtls_verify=    ; Verify that the provided peer certificate is valid (default:
                 ; "")
;dtls_rekey=     ; Interval at which to renegotiate the TLS session and rekey
                 ; the SRTP session (default: "")
;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: "")
;srtp_tag_32=no ; Determines whether 32 byte tags should be used instead of 80
                ; byte tags (default: "no")


;==========================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=asterisk ; SIP realm for endpoint (default: "asterisk")
;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: "")
;cert_file=     ; Certificate file for endpoint TLS ONLY (default: "")
;cipher=        ; Preferred Cryptography Cipher TLS ONLY (default: "")
;domain=        ; Domain the transport comes from (default: "")
;external_media_address=        ; External 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")


;==========================CONTACT SECTION OPTIONS=========================
;[contact]
;  SYNOPSIS: A way of creating an aliased name to a SIP URI
;type=  ; Must be of type contact (default: "")
;uri=   ; SIP URI to contact peer (default: "")
;expiration_time=       ; Time to keep alive a contact (default: "")
;qualify_frequency=0    ; Interval at which to qualify a contact (default: "0")


;==========================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")
;authenticate_qualify=no        ; Authenticates a qualify request if needed
                                ; (default: "no")


;==========================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")
;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=     ; Value used in User Agent header for SIP requests and Server
                 ; header for SIP responses (default: Populated by Asterisk
                 ; Version)
;default_outbound_endpoint= ; Endpoint to use when sending an outbound request
                            ; to a URI without a specified endpoint.
                            ; (default: "default_outbound_endpoint")


; MODULE PROVIDING BELOW SECTION(S): res_pjsip_acl
;==========================ACL SECTION OPTIONS=========================
;[acl]
;  SYNOPSIS: Access Control List
;acl=   ; Name of IP ACL (default: "")
;contact_acl=    ; Name of Contact ACL (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 domains to deny access from (default: "")
;permit=        ; List of IP domains to allow access from (default: "")
;type=  ; Must be of type security (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")
;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: NEEDS A SYNOPSIS
;endpoint=      ; Name of Endpoint (default: "")
;match= ; IP addresses or networks to match against (default: "")
;type=  ; Must be of type identify (default: "")