-
Rusty Newton authored
pjsip.conf: privkey_file should be priv_key_file, mediaencryption=yes should be mediaencryption=sdes privkey_file was missed in the snake case update. An example included an invalid value for the mediaencryption option. ........ Merged revisions 414780 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Rusty Newton authoredpjsip.conf: privkey_file should be priv_key_file, mediaencryption=yes should be mediaencryption=sdes privkey_file was missed in the snake case update. An example included an invalid value for the mediaencryption option. ........ Merged revisions 414780 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
pjsip.conf.sample 30.99 KiB
; 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")
;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")
;==========================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: "")