configs: Move sample config files into a subdirectory of configs

This moves all samples configs from configs/ to configs/samples. This allows
for additional sets of sample configuration files to be added in the future.

Review: https://reviewboard.asterisk.org/r/3804/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 0 insertions, 751 deletions

diff --git a/configs/pjsip.conf.sample b/configs/pjsip.conf.sample
deleted file mode 100644
index 58774dc7c..000000000
--- a/configs/pjsip.conf.sample
+++ /dev/null
@@ -1,751 +0,0 @@
-; 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.
-
-;=============================================================================
-
-; 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=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.
-;
-
-;[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
-;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
-
-
-
-; MODULE PROVIDING BELOW SECTION(S): res_pjsip
-;==========================ENDPOINT SECTION OPTIONS=========================
-;[endpoint]
-;  SYNOPSIS: Endpoint
-;100rel=yes     ; Allow support for RFC3262 provisional ACK tags (default:
-                ; "yes")
-;accountcode=foo        ; Set a default account code for channels created for
-                        ; this endpoint
-;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")
-;media_address=         ; IP address used in SDP for 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")
-;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")
-;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")
-;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: "")
-;from_domain=   ; 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")
-;set_var=       ; Variable set on a channel involving the endpoint. For multiple
-		; channel variables specify multiple 'set_var'(s)
-
-;==========================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: "")
-;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 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.
-
-;==========================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")
-;outbound_proxy=        ; Outbound proxy used when sending OPTIONS request
-                        ; (default: "")
-
-
-;==========================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")
-;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")
-;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")
-
-
-; 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")
-;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: "")