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Comment: Updated to SVN-branch-13-r433694

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Option Name

Type

Default Value

Regular Expression

Description

100rel

Custom

yes

false

Allow support for RFC3262 provisional ACK tags

aggregate_mwi

Boolean

yes

false

Condense MWI notifications into a single NOTIFY.

allow

Codec

 

false

Media Codec(s) to allow

aors

String

 

false

AoR(s) to be used with the endpoint

auth

Custom

 

false

Authentication Object(s) associated with the endpoint

callerid

Custom

 

false

CallerID information for the endpoint

callerid_privacy

Custom

allowed_not_screened

false

Default privacy level

callerid_tag

Custom

 

false

Internal id_tag for the endpoint

context

String

default

false

Dialplan context for inbound sessions

direct_media_glare_mitigation

Custom

none

false

Mitigation of direct media (re)INVITE glare

direct_media_method

Custom

invite

false

Direct Media method type

connected_line_method

Custom

invite

false

Connected line method type

direct_media

Boolean

yes

false

Determines whether media may flow directly between endpoints.

disable_direct_media_on_nat

Boolean

no

false

Disable direct media session refreshes when NAT obstructs the media session

disallow

 

 

 

Media Codec(s) to disallow

dtmf_mode

Custom

rfc4733

false

DTMF mode

media_address

String

 

false

IP address used in SDP for media handling

force_rport

Boolean

yes

false

Force use of return port

ice_support

Boolean

no

false

Enable the ICE mechanism to help traverse NAT

identify_by

Custom

username

false

Way(s) for Endpoint to be identified

redirect_method

Custom

user

false

How redirects received from an endpoint are handled

mailboxes

String

 

false

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

moh_suggest

String

default

false

Default Music On Hold class

outbound_auth

Custom

 

false

Authentication object used for outbound requests

outbound_proxy

String

 

false

Proxy through which to send requests, a full SIP URI must be provided

rewrite_contact

Boolean

no

false

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

rtp_ipv6

Boolean

no

false

Allow use of IPv6 for RTP traffic

rtp_symmetric

Boolean

no

false

Enforce that RTP must be symmetric

send_diversion

Boolean

yes

false

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

send_pai

Boolean

no

false

Send the P-Asserted-Identity header

send_rpid

Boolean

no

false

Send the Remote-Party-ID header

rpid_immediate

Boolean

no

false

Immediately send connected line updates on unanswered incoming calls.

timers_min_se

Unsigned Integer

90

false

Minimum session timers expiration period

timers

Custom

yes

false

Session timers for SIP packets

timers_sess_expires

Unsigned Integer

1800

false

Maximum session timer expiration period

transport

String

 

false

Desired transport configuration

trust_id_inbound

Boolean

no

false

Accept identification information received from this endpoint

trust_id_outbound

Boolean

no

false

Send private identification details to the endpoint.

type

None

 

false

Must be of type 'endpoint'.

use_ptime

Boolean

no

false

Use Endpoint's requested packetisation interval

use_avpf

Boolean

no

false

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

force_avp

Boolean

no

false

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

Boolean

no

false

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

media_encryption

Custom

no

false

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

media_encryption_optimistic

Boolean

no

false

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

inband_progress

Boolean

no

false

Determines whether chan_pjsip will indicate ringing using inband progress.

call_group

Custom

 

false

The numeric pickup groups for a channel.

pickup_group

Custom

 

false

The numeric pickup groups that a channel can pickup.

named_call_group

Custom

 

false

The named pickup groups for a channel.

named_pickup_group

Custom

 

false

The named pickup groups that a channel can pickup.

device_state_busy_at

Unsigned Integer

0

false

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

t38_udptl

Boolean

no

false

Whether T.38 UDPTL support is enabled or not

t38_udptl_ec

Custom

none

false

T.38 UDPTL error correction method

t38_udptl_maxdatagram

Unsigned Integer

0

false

T.38 UDPTL maximum datagram size

fax_detect

Boolean

no

false

Whether CNG tone detection is enabled

t38_udptl_nat

Boolean

no

false

Whether NAT support is enabled on UDPTL sessions

t38_udptl_ipv6

Boolean

no

false

Whether IPv6 is used for UDPTL Sessions

tone_zone

String

 

false

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

language

String

 

false

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

one_touch_recording

Boolean

no

false

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

record_on_feature

String

automixmon

false

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

record_off_feature

String

automixmon

false

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

rtp_engine

String

asterisk

false

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

allow_transfer

Boolean

yes

false

Determines whether SIP REFER transfers are allowed for this endpoint

user_eq_phone

Boolean

no

false

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

-

false

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

sdp_session

String

Asterisk

false

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

tos_audio

Custom

0

false

DSCP TOS bits for audio streams

tos_video

Custom

0

false

DSCP TOS bits for video streams

cos_audio

Unsigned Integer

0

false

Priority for audio streams

cos_video

Unsigned Integer

0

false

Priority for video streams

allow_subscribe

Boolean

yes

false

Determines if endpoint is allowed to initiate subscriptions with Asterisk.

sub_min_expiry

Unsigned Integer

0

false

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

from_user

String

 

false

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

mwi_from_user

String

 

false

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

from_domain

String

 

false

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

dtls_verify

Custom

no

false

Verify that the provided peer certificate is valid

dtls_rekey

Custom

0

false

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

dtls_cert_file

Custom

 

false

Path to certificate file to present to peer

dtls_private_key

Custom

 

false

Path to private key for certificate file

dtls_cipher

Custom

 

false

Cipher to use for DTLS negotiation

dtls_ca_file

Custom

 

false

Path to certificate authority certificate

dtls_ca_path

Custom

 

false

Path to a directory containing certificate authority certificates

dtls_setup

Custom

 

false

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

dtls_fingerprint

Custom

 

false

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

srtp_tag_32

Boolean

no

false

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

set_var

Custom

 

false

Variable set on a channel involving the endpoint.

message_context

String

 

false

Context to route incoming MESSAGE requests to.

accountcode

String

 

false

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

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On inbound SIP messages from this endpoint, the Contact header will be changed to have the source IP address and port. This option does not affect outbound messages send to this endpoint.

Anchor
endpoint_rpid_immediate
endpoint_rpid_immediate

rpid_immediate

When enabled, immediately send 180 Ringing or 183 Progress response messages to the caller if the connected line information is updated before the call is answered. This can send a 180 Ringing response before the call has even reached the far end. The caller can start hearing ringback before the far end even gets the call. Many phones tend to grab the first connected line information and refuse to update the display if it changes. The first information is not likely to be correct if the call goes to an endpoint not under the control of this Asterisk box.

When disabled, a connected line update must wait for another reason to send a message with the connected line information to the caller before the call is answered. You can trigger the sending of the information by using an appropriate dialplan application such as Ringing.

Anchor
endpoint_timers_min_se
endpoint_timers_min_se

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Option Name

Type

Default Value

Regular Expression

Description

max_forwards

Unsigned Integer

70

false

Value used in Max-Forwards header for SIP requests.

keep_alive_interval

Unsigned Integer

0

false

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

type

None

 

false

Must be of type 'global'.

user_agent

String

Asterisk PBX SVN-branch-13-r433222r433694

false

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

default_outbound_endpoint

String

default_outbound_endpoint

false

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

debug

String

no

false

Enable/Disable SIP debug logging. Valid options include yes

no or a host address

endpoint_identifier_order

String

ip,username,anonymous

false

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_*)

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This documentation was imported from Asterisk Version SVN-branch-13-r433222r433694