Versions Compared

Key

  • This line was added.
  • This line was removed.
  • Formatting was changed.
Comment: Updated to GIT-13-13.15.0-rc1-299-gcad74cd

...

Option Name

Type

Default Value

Regular Expression

Description

100rel

 

yes

 

Allow support for RFC3262 provisional ACK tags

aggregate_mwi

 

yes

 

Condense MWI notifications into a single NOTIFY.

allow

 

 

 

Media Codec(s) to allow

allow_overlap

 

yes

 

Enable RFC3578 overlap dialing support.

aors

 

 

 

AoR(s) to be used with the endpoint

auth

 

 

 

Authentication Object(s) associated with the endpoint

callerid

 

 

 

CallerID information for the endpoint

callerid_privacy

 

 

 

Default privacy level

callerid_tag

 

 

 

Internal id_tag for the endpoint

context

 

 

 

Dialplan context for inbound sessions

direct_media_glare_mitigation

 

none

 

Mitigation of direct media (re)INVITE glare

direct_media_method

 

invite

 

Direct Media method type

connected_line_method

 

invite

 

Connected line method type

direct_media

 

yes

 

Determines whether media may flow directly between endpoints.

disable_direct_media_on_nat

 

no

 

Disable direct media session refreshes when NAT obstructs the media session

disallow

 

 

 

Media Codec(s) to disallow

dtmf_mode

 

rfc4733

 

DTMF mode

media_address

 

 

 

IP address used in SDP for media handling

bind_rtp_to_media_address

 

 

 

Bind the RTP instance to the media_address

force_rport

 

yes

 

Force use of return port

ice_support

 

no

 

Enable the ICE mechanism to help traverse NAT

identify_by

 

username,location

 

Way(s) for Endpoint to be identified

redirect_method

 

 

 

How redirects received from an endpoint are handled

mailboxes

 

 

 

NOTIFY the endpoint when state changes for any of the specified mailboxes

mwi_subscribe_replaces_unsolicited

 

 

 

An MWI subscribe will replace sending unsolicited NOTIFYs

voicemail_extension

 

 

 

The voicemail extension to send in the NOTIFY Message-Account header

moh_suggest

 

default

 

Default Music On Hold class

outbound_auth

 

 

 

Authentication object(s) used for outbound requests

outbound_proxy

 

 

 

Full SIP URI of the outbound proxy used to send requests

rewrite_contact

 

 

 

Allow Contact header to be rewritten with the source IP address-port

rtp_ipv6

 

no

 

Allow use of IPv6 for RTP traffic

rtp_symmetric

 

no

 

Enforce that RTP must be symmetric

send_diversion

 

yes

 

Send the Diversion header, conveying the diversion information to the called user agent

send_pai

 

no

 

Send the P-Asserted-Identity header

send_rpid

 

no

 

Send the Remote-Party-ID header

rpid_immediate

 

no

 

Immediately send connected line updates on unanswered incoming calls.

timers_min_se

 

90

 

Minimum session timers expiration period

timers

 

yes

 

Session timers for SIP packets

timers_sess_expires

 

1800

 

Maximum session timer expiration period

transport

 

 

 

Desired transport configuration

trust_id_inbound

 

no

 

Accept identification information received from this endpoint

trust_id_outbound

 

no

 

Send private identification details to the endpoint.

type

 

 

 

Must be of type 'endpoint'.

use_ptime

 

no

 

Use Endpoint's requested packetisation interval

use_avpf

 

no

 

Determines whether res_pjsip will use and enforce usage of AVPF for this endpoint.

force_avp

 

no

 

Determines whether res_pjsip will use and enforce usage of AVP, regardless of the RTP profile in use for this endpoint.

media_use_received_transport

 

no

 

Determines whether res_pjsip will use the media transport received in the offer SDP in the corresponding answer SDP.

media_encryption

 

no

 

Determines whether res_pjsip will use and enforce usage of media encryption for this endpoint.

media_encryption_optimistic

 

no

 

Determines whether encryption should be used if possible but does not terminate the session if not achieved.

g726_non_standard

 

no

 

Force g.726 to use AAL2 packing order when negotiating g.726 audio

inband_progress

 

no

 

Determines whether chan_pjsip will indicate ringing using inband progress.

call_group

 

 

 

The numeric pickup groups for a channel.

pickup_group

 

 

 

The numeric pickup groups that a channel can pickup.

named_call_group

 

 

 

The named pickup groups for a channel.

named_pickup_group

 

 

 

The named pickup groups that a channel can pickup.

device_state_busy_at

 

0

 

The number of in-use channels which will cause busy to be returned as device state

t38_udptl

 

no

 

Whether T.38 UDPTL support is enabled or not

t38_udptl_ec

 

none

 

T.38 UDPTL error correction method

t38_udptl_maxdatagram

 

0

 

T.38 UDPTL maximum datagram size

fax_detect

 

no

 

Whether CNG tone detection is enabled

fax_detect_timeout

 

 

 

How long into a call before fax_detect is disabled for the call

t38_udptl_nat

 

no

 

Whether NAT support is enabled on UDPTL sessions

t38_udptl_ipv6

 

no

 

Whether IPv6 is used for UDPTL Sessions

tone_zone

 

 

 

Set which country's indications to use for channels created for this endpoint.

language

 

 

 

Set the default language to use for channels created for this endpoint.

one_touch_recording

 

no

 

Determines whether one-touch recording is allowed for this endpoint.

record_on_feature

 

automixmon

 

The feature to enact when one-touch recording is turned on.

record_off_feature

 

automixmon

 

The feature to enact when one-touch recording is turned off.

rtp_engine

 

asterisk

 

Name of the RTP engine to use for channels created for this endpoint

allow_transfer

 

yes

 

Determines whether SIP REFER transfers are allowed for this endpoint

user_eq_phone

 

no

 

Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number

sdp_owner

 

-

 

String placed as the username portion of an SDP origin (o=) line.

sdp_session

 

Asterisk

 

String used for the SDP session (s=) line.

tos_audio

 

 

 

DSCP TOS bits for audio streams

tos_video

 

 

 

DSCP TOS bits for video streams

cos_audio

 

 

 

Priority for audio streams

cos_video

 

 

 

Priority for video streams

allow_subscribe

 

yes

 

Determines if endpoint is allowed to initiate subscriptions with Asterisk.

sub_min_expiry

 

60

 

The minimum allowed expiry time for subscriptions initiated by the endpoint.

from_user

 

 

 

Username to use in From header for requests to this endpoint.

mwi_from_user

 

 

 

Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.

from_domain

 

 

 

Domain to user in From header for requests to this endpoint.

dtls_verify

 

 

 

Verify that the provided peer certificate is valid

dtls_rekey

 

 

 

Interval at which to renegotiate the TLS session and rekey the SRTP session

dtls_cert_file

 

 

 

Path to certificate file to present to peer

dtls_private_key

 

 

 

Path to private key for certificate file

dtls_cipher

 

 

 

Cipher to use for DTLS negotiation

dtls_ca_file

 

 

 

Path to certificate authority certificate

dtls_ca_path

 

 

 

Path to a directory containing certificate authority certificates

dtls_setup

 

 

 

Whether we are willing to accept connections, connect to the other party, or both.

dtls_fingerprint

 

 

 

Type of hash to use for the DTLS fingerprint in the SDP.

srtp_tag_32

 

 

 

Determines whether 32 byte tags should be used instead of 80 byte tags.

set_var

 

 

 

Variable set on a channel involving the endpoint.

message_context

 

 

 

Context to route incoming MESSAGE requests to.

accountcode

 

 

 

An accountcode to set automatically on any channels created for this endpoint.

rtp_keepalive

 

 

 

Number of seconds between RTP comfort noise keepalive packets.

rtp_timeout

 

0

 

Maximum number of seconds without receiving RTP (while off hold) before terminating call.

rtp_timeout_hold

 

0

 

Maximum number of seconds without receiving RTP (while on hold) before terminating call.

acl

 

 

 

List of IP ACL section names in acl.conf

deny

 

 

 

List of IP addresses to deny access from

permit

 

 

 

List of IP addresses to permit access from

contact_acl

 

 

 

List of Contact ACL section names in acl.conf

contact_deny

 

 

 

List of Contact header addresses to deny

contact_permit

 

 

 

List of Contact header addresses to permit

subscribe_context

 

 

 

Context for incoming MESSAGE requests.

contact_user

 

 

 

Force the user on the outgoing Contact header to this value.

asymmetric_rtp_codec

 

no

 

Allow the sending and receiving RTP codec to differ

rtcp_mux

 

no

 

Enable RFC 5761 RTCP multiplexing on the RTP port

refer_blind_progress

 

yes

 

Whether to notifies all the progress details on blind transfer

notify_early_inuse_ringing

 

no

 

Whether to notifies dialog-info 'early' on InUse&Ringing state

...

Method used when updating connected line information.

  • invite - When set to invite, check the remote's Allow header and if UPDATE is allowed, send UPDATE instead of INVITE to avoid SDP renegotiation. If UPDATE is not Allowed, send INVITE.
  • reinvite - Alias for the invite value.
  • update - If set to update, send UPDATE regardless of what the remote Allows.

Anchor
endpoint_dtmf_mode
endpoint_dtmf_mode

...

Option Name

Type

Default Value

Regular Expression

Description

auth_type

 

userpass

 

Authentication type

nonce_lifetime

 

32

 

Lifetime of a nonce associated with this authentication config.

md5_cred

 

 

 

MD5 Hash used for authentication.

password

 

 

 

PlainText password used for authentication.

realm

 

 

 

SIP realm for endpoint

type

 

 

 

Must be 'auth'

username

 

 

 

Username to use for account

...

Option Name

Type

Default Value

Regular Expression

Description

async_operations

 

1

 

Number of simultaneous Asynchronous Operations

bind

 

 

 

IP Address and optional port to bind to for this transport

ca_list_file

 

 

 

File containing a list of certificates to read (TLS ONLY)

ca_list_path

 

 

 

Path to directory containing a list of certificates to read (TLS ONLY)

cert_file

 

 

 

Certificate file for endpoint (TLS ONLY)

cipher

 

 

 

Preferred cryptography cipher names (TLS ONLY)

domain

 

 

 

Domain the transport comes from

external_media_address

 

 

 

External IP address to use in RTP handling

external_signaling_address

 

 

 

External address for SIP signalling

external_signaling_port

 

0

 

External port for SIP signalling

method

 

 

 

Method of SSL transport (TLS ONLY)

local_net

 

 

 

Network to consider local (used for NAT purposes).

password

 

 

 

Password required for transport

priv_key_file

 

 

 

Private key file (TLS ONLY)

protocol

 

udp

 

Protocol to use for SIP traffic

require_client_cert

 

false

 

Require client certificate (TLS ONLY)

type

 

 

 

Must be of type 'transport'.

verify_client

 

false

 

Require verification of client certificate (TLS ONLY)

verify_server

 

false

 

Require verification of server certificate (TLS ONLY)

tos

 

false

 

Enable TOS for the signalling sent over this transport

cos

 

false

 

Enable COS for the signalling sent over this transport

websocket_write_timeout

 

 

 

The timeout (in milliseconds) to set on WebSocket connections.

allow_reload

 

no

 

Allow this transport to be reloaded.

symmetric_transport

 

no

 

Use the same transport for outgoing reqests as incoming ones.

...

Option Name

Type

Default Value

Regular Expression

Description

type

 

 

 

Must be of type 'contact'.

uri

 

 

 

SIP URI to contact peer

expiration_time

 

 

 

Time to keep alive a contact

qualify_frequency

 

0

 

Interval at which to qualify a contact

qualify_timeout

 

3.0

 

Timeout for qualify

authenticate_qualify

 

no

 

Authenticates a qualify request if needed

outbound_proxy

 

 

 

Outbound proxy used when sending OPTIONS request

path

 

 

 

Stored Path vector for use in Route headers on outgoing requests.

user_agent

 

 

 

User-Agent header from registration.

endpoint

 

 

 

Endpoint name

reg_server

 

 

 

Asterisk Server name

via_addr

 

 

 

IP-address of the last Via header from registration.

via_port

 

 

 

IP-port of the last Via header from registration.

call_id

 

 

 

Call-ID header from registration.

...

Option Name

Type

Default Value

Regular Expression

Description

contact

 

 

 

Permanent contacts assigned to AoR

default_expiration

 

3600

 

Default expiration time in seconds for contacts that are dynamically bound to an AoR.

mailboxes

 

 

 

Allow subscriptions for the specified mailbox(es)

voicemail_extension

 

 

 

The voicemail extension to send in the NOTIFY Message-Account header

maximum_expiration

 

7200

 

Maximum time to keep an AoR

max_contacts

 

0

 

Maximum number of contacts that can bind to an AoR

minimum_expiration

 

60

 

Minimum keep alive time for an AoR

remove_existing

 

no

 

Determines whether new contacts replace existing ones.

type

 

 

 

Must be of type 'aor'.

qualify_frequency

 

0

 

Interval at which to qualify an AoR

qualify_timeout

 

3.0

 

Timeout for qualify

authenticate_qualify

 

no

 

Authenticates a qualify request if needed

outbound_proxy

 

 

 

Outbound proxy used when sending OPTIONS request

support_path

 

 

 

Enables Path support for REGISTER requests and Route support for other requests.

...

Option Name

Type

Default Value

Regular Expression

Description

timer_t1

 

500

 

Set transaction timer T1 value (milliseconds).

timer_b

 

32000

 

Set transaction timer B value (milliseconds).

compact_headers

 

no

 

Use the short forms of common SIP header names.

threadpool_initial_size

 

0

 

Initial number of threads in the res_pjsip threadpool.

threadpool_auto_increment

 

5

 

The amount by which the number of threads is incremented when necessary.

threadpool_idle_timeout

 

60

 

Number of seconds before an idle thread should be disposed of.

threadpool_max_size

 

0

 

Maximum number of threads in the res_pjsip threadpool. A value of 0 indicates no maximum.

disable_tcp_switch

 

yes

 

Disable automatic switching from UDP to TCP transports.

type

 

 

 

Must be of type 'system'.

...

Option Name

Type

Default Value

Regular Expression

Description

max_forwards

 

70

 

Value used in Max-Forwards header for SIP requests.

keep_alive_interval

 

0

 

The interval (in seconds) to send keepalives to active connection-oriented transports.

contact_expiration_check_interval

 

30

 

The interval (in seconds) to check for expired contacts.

disable_multi_domain

 

no

 

Disable Multi Domain support

max_initial_qualify_time

 

0

 

The maximum amount of time from startup that qualifies should be attempted on all contacts. If greater than the qualify_frequency for an aor, qualify_frequency will be used instead.

unidentified_request_period

 

5

 

The number of seconds over which to accumulate unidentified requests.

unidentified_request_count

 

5

 

The number of unidentified requests from a single IP to allow.

unidentified_request_prune_interval

 

30

 

The interval at which unidentified requests are older than twice the unidentified_request_period are pruned.

type

 

 

 

Must be of type 'global'.

user_agent

 

Asterisk <Asterisk Version>

 

Value used in User-Agent header for SIP requests and Server header for SIP responses.

regcontext

 

 

 

When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us.

default_outbound_endpoint

 

default_outbound_endpoint

 

Endpoint to use when sending an outbound request to a URI without a specified endpoint.

default_voicemail_extension

 

 

 

The voicemail extension to send in the NOTIFY Message-Account header if not specified on endpoint or aor

debug

 

no

 

Enable/Disable SIP debug logging. Valid options include yes

no or a host address

endpoint_identifier_order

 

ip,username,anonymous

 

The order by which endpoint identifiers are processed and checked. Identifier names are usually derived from and can be found in the endpoint identifier module itself (res_pjsip_endpoint_identifier_*). You can use the CLI command "pjsip show identifiers" to see the identifiers currently available.

default_from_user

 

asterisk

 

When Asterisk generates an outgoing SIP request, the From header username will be set to this value if there is no better option (such as CallerID) to be used.

default_realm

 

asterisk

 

When Asterisk generates a challenge, the digest realm will be set to this value if there is no better option (such as auth/realm) to be used.

mwi_tps_queue_high

 

500

 

MWI taskprocessor high water alert trigger level.

mwi_tps_queue_low

 

-1

 

MWI taskprocessor low water clear alert level.

mwi_disable_initial_unsolicited

 

no

 

Enable/Disable sending unsolicited MWI to all endpoints on startup.

ignore_uri_user_options

 

 

 

Enable/Disable ignoring SIP URI user field options.

...

This documentation was imported from Asterisk Version GIT-13-13.15.0-rc1-276299-g1f59d08gcad74cd