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Comment: Updated to GIT-13-acb614d

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Option Name

Type

Default Value

Regular Expression

Description

type

None

 

false

Must be of type 'contact'.

uri

String

 

false

SIP URI to contact peer

expiration_time

Custom

 

false

Time to keep alive a contact

qualify_frequency

Unsigned Integer

0

false

Interval at which to qualify a contact

qualify_timeout

Double

3.0

false

Timeout for qualify

authenticate_qualify

Boolean

no

false

Authenticates a qualify request if needed

outbound_proxy

String

 

false

Outbound proxy used when sending OPTIONS request

path

String

 

false

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

user_agent

String

 

false

User-Agent header from registration.

reg_server

String

 

false

Asterisk Server name

via_addr

String

 

false

IP-address of the last Via header from registration.

via_port

Unsigned Integer

0

false

IP-port of the last Via header from registration.

call_id

String

 

false

Call-ID header from registration.

Configuration Option Descriptions

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Asterisk Server name on which SIP endpoint registered.

Anchor
contact_via_addr
contact_via_addr

via_addr

The last Via header should contain the address of UA which sent the request. The IP-address of the last Via header is automatically stored based on data present in incoming SIP REGISTER requests and is not intended to be configured manually.

Anchor
contact_via_port
contact_via_port

via_port

The IP-port of the last Via header is automatically stored based on data present in incoming SIP REGISTER requests and is not intended to be configured manually.

Anchor
contact_call_id
contact_call_id

call_id

The Call-ID header is automatically stored based on data present in incoming SIP REGISTER requests and is not intended to be configured manually.

aor

The configuration for a location of an endpoint

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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.

contact_expiration_check_interval

Unsigned Integer

30

false

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

disable_multi_domain

Boolean

no

false

Disable Multi Domain support

max_initial_qualify_time

Unsigned Integer

0

false

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

Unsigned Integer

5

false

The number of seconds over which to accumulate unidentified requests.

unidentified_request_count

Unsigned Integer

5

false

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

unidentified_request_prune_interval

Unsigned Integer

30

false

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

type

None

 

false

Must be of type 'global'.

user_agent

String

Asterisk PBX GIT-13-dc230f3acb614d

false

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

regcontext

String

 

false

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

String

default_outbound_endpoint

false

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

default_voicemail_extension

String

 

false

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

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_*). You can use the CLI command "pjsip show identifiers" to see the identifiers currently available.

default_from_user

String

asterisk

false

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

String

asterisk

false

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

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This documentation was imported from Asterisk Version GIT-13-dc230f3acb614d