From be219c9ec9bcb1e54b8eed5ffaf7bced2f34abc4 Mon Sep 17 00:00:00 2001 From: Rusty Newton Date: Fri, 30 Aug 2013 20:37:54 +0000 Subject: New pjsip.conf.sample (issue ASTERISK-22145) (closes issue ASTERISK-22145) Reported By: Matt Jordan Review: https://reviewboard.asterisk.org/r/2811/ ........ Merged revisions 398147 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398148 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/pjsip.conf.sample | 682 ++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 658 insertions(+), 24 deletions(-) (limited to 'configs') diff --git a/configs/pjsip.conf.sample b/configs/pjsip.conf.sample index dd4ff151c..dbeda4d62 100644 --- a/configs/pjsip.conf.sample +++ b/configs/pjsip.conf.sample @@ -1,26 +1,660 @@ -; This is an in-flux configuration file for the res_pjsip module, it will change as things progress +; 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. -;;; Transports -;[local] +; 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 ; Supported protocols are udp, tcp, and tls -;bind=0.0.0.0 ; This supports both IPv4 and IPv6, port is optional - -;;; Endpoints -[endpoint] -type=endpoint -context=default -disallow=all -allow=ulaw -dtmfmode=rfc4733 ; Supported DTMF modes are rfc4733, inband, info, and none -;transport=local ; Name of a specific transport to use when placing calls -;100rel=yes ; Enable or disable 100rel support - valid options are: yes, no, required -;timers=yes ; Enable or disable session timers support - valid options are: yes, no, required, always -;timers_min_se=90 ; Minimum session timers expiration period, in seconds -;timers_sess_expires=1800 ; Session timers expiration period, in seconds -;mohsuggest=example ; What musiconhold class to suggest that the peer channel use when this endpoint places them on hold -;rtp_ipv6=yes ; Force IPv6 for RTP transport -;rtp_symmetric=yes ; Enable symmetric RTP support -;use_ptime=yes ; Whether to use the ptime value received from the endpoint or not -;media_encryption=no ; Options for media encryption are no, and sdes -;use_avpf=no ; Whether to force usage of AVPF transport for this endpoint +;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 +;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: "") +;dtmfmode=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: "") +;mohsuggest=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") +;callgroup= ; The numeric pickup groups for a channel (default: "") +;pickupgroup= ; The numeric pickup groups that a channel can pickup (default: + ; "") +;namedcallgroup= ; The named pickup groups for a channel (default: "") +;namedpickupgroup= ; The named pickup groups that a channel can pickup + ; (default: "") +;devicestate_busy_at=0 ; The number of in use channels which will cause busy + ; to be returned as device state (default: "0") +;t38udptl=no ; Whether T 38 UDPTL support is enabled or not (default: "no") +;t38udptl_ec=none ; T 38 UDPTL error correction method (default: "none") +;t38udptl_maxdatagram=0 ; T 38 UDPTL maximum datagram size (default: "0") +;faxdetect=no ; Whether CNG tone detection is enabled (default: "no") +;t38udptl_nat=no ; Whether NAT support is enabled on UDPTL sessions + ; (default: "no") +;t38udptl_ipv6=no ; Whether IPv6 is used for UDPTL Sessions (default: + ; "no") +;tonezone= ; 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") +;recordonfeature=automixmon ; The feature to enact when one touch recording + ; is turned on (default: "automixmon") +;recordofffeature=automixmon ; The feature to enact when one touch recording + ; is turned off (default: "automixmon") +;rtpengine=asterisk ; Name of the RTP engine to use for channels created + ; for this endpoint (default: "asterisk") +;allowtransfer=yes ; Determines whether SIP REFER transfers are allowed + ; for this endpoint (default: "yes") +;sdpowner=- ; String placed as the username portion of an SDP origin o line + ; (default: "-") +;sdpsession=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") +;allowsubscribe=yes ; Determines if endpoint is allowed to initiate + ; subscriptions with Asterisk (default: "yes") +;subminexpiry=0 ; The minimum allowed expiry time for subscriptions initiated + ; by the endpoint (default: "0") +;fromuser= ; Username to use in From header for requests to this endpoint + ; (default: "") +;mwifromuser= ; 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: "") +;dtlsverify= ; Verify that the provided peer certificate is valid (default: + ; "") +;dtlsrekey= ; Interval at which to renegotiate the TLS session and rekey + ; the SRTP session (default: "") +;dtlscertfile= ; Path to certificate file to present to peer (default: "") +;dtlsprivatekey= ; Path to private key for certificate file (default: + ; "") +;dtlscipher= ; Cipher to use for DTLS negotiation (default: "") +;dtlscafile= ; Path to certificate authority certificate (default: "") +;dtlscapath= ; Path to a directory containing certificate authority + ; certificates (default: "") +;dtlssetup= ; 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: "") +;localnet= ; Network to consider local used for NAT purposes (default: "") +;password= ; Password required for transport (default: "") +;privkey_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 +;timert1=500 ; Set transaction timer T1 value milliseconds (default: "500") +;timerb=32000 ; Set transaction timer B value milliseconds (default: "32000") +;compactheaders=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 +;maxforwards=70 ; Value used in Max Forwards header for SIP requests (default: + ; "70") +;type= ; Must be of type global (default: "") +;useragent= ; Value used in User Agent header for SIP requests and Server + ; header for SIP responses (default: Populated by Asterisk + ; Version) + + + + +; MODULE PROVIDING BELOW SECTION(S): res_pjsip_acl +;==========================ACL SECTION OPTIONS========================= +;[acl] +; SYNOPSIS: Access Control List +;acl= ; Name of IP ACL (default: "") +;contactacl= ; Name of Contact ACL (default: "") +;contactdeny= ; List of Contact Header addresses to Deny (default: "") +;contactpermit= ; 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: "") + + + + -- cgit v1.2.3