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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

allow_overlap

Boolean

yes

false

Enable RFC3578 overlap dialing support.

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

trust_connected_line

Boolean

yes

false

Accept Connected Line updates from this endpoint

send_connected_line

Boolean

yes

false

Send Connected Line updates to this endpoint

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

bind_rtp_to_media_address

Boolean

no

false

Bind the RTP instance to the media_address

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,ip

false

Way(s) for Endpoint the 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

mwi_subscribe_replaces_unsolicited

Boolean

no

false

An MWI subscribe will replace sending unsolicited NOTIFYs

voicemail_extension

Custom

 

false

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

moh_suggest

String

default

false

Default Music On Hold class

outbound_auth

Custom

 

false

Authentication object(s) used for outbound requests

outbound_proxy

String

 

false

Proxy through which Full SIP URI of the outbound proxy used 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_history_info

Boolean

no

false

Send the History-Info header, conveying the diversion information to the called and calling user agents

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 Explicit transport configuration to use

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 packetization 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.

g726_non_standard

Boolean

no

false

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

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

fax_detect_timeout

Unsigned Integer

0

false

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

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

moh_passthrough

Boolean

no

false

Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side

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 Custom

 

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.

rtp_keepalive

Unsigned Integer

0

false

Number of seconds between RTP comfort noise keepalive packets.

rtp_timeout

Unsigned Integer

0

false

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

rtp_timeout_hold

Unsigned Integer

0

false

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

acl

Custom

 

false

List of IP ACL section names in acl.conf

deny

Custom

 

false

List of IP addresses to deny access from

permit

Custom

 

false

List of IP addresses to permit access from

contact_acl

Custom

 

false

List of Contact ACL section names in acl.conf

contact_deny

Custom

 

false

List of Contact header addresses to deny

contact_permit

Custom

 

false

List of Contact header addresses to permit

subscribe_context

String

 

false

Context for incoming MESSAGE requests.

contact_user

Custom

 

false

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

asymmetric_rtp_codec

Boolean

no

false

Allow the sending and receiving RTP codec to differ

rtcp_mux

Boolean

no

false

Enable RFC 5761 RTCP multiplexing on the RTP port

refer_blind_progress

Boolean

yes

false

Whether to notifies all the progress details on blind transfer

notify_early_inuse_ringing

Boolean

no

false

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

incoming_mwi_mailbox

String

 

false

Mailbox name to use when incoming MWI NOTIFYs are received

follow_early_media_fork

Boolean

yes

false

Follow SDP forked media when To tag is different

accept_multiple_sdp_answers

Boolean

no

false

Accept multiple SDP answers on non-100rel responses

suppress_q850_reason_headers

Boolean

no

false

Suppress Q.850 Reason headers for this endpoint

ignore_183_without_sdp

Boolean

no

false

Do not forward 183 when it doesn't contain SDP

Configuration Option Descriptions

...

This is a comma-delimited list of auth sections defined in pjsip.conf to be used to verify inbound connection attempts.

Endpoints without an authentication object configured will allow connections without vertification.verification.

Info
titleNote

Using the same auth section for inbound and outbound authentication is not recommended. There is a difference in meaning for an empty realm setting between inbound and outbound authentication uses. See the auth realm description for details.

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endpoint_callerid
endpoint_callerid

...

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.

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endpoint_dtmf_mode
endpoint_dtmf_mode

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  • rfc4733 - DTMF is sent out of band of the main audio stream. This supercedes the older RFC-2833 used within the older chan_sip.
  • inband - DTMF is sent as part of audio stream.
  • info - DTMF is sent as SIP INFO packets.
  • auto - DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.
  • auto_info - DTMF is sent as RFC 4733 if the other side supports it or as SIP INFO if not.

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endpoint_media_address
endpoint_media_address

...

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endpoint_identify_by
endpoint_identify_by

identify_by

Endpoints and aors AORs can be identified in multiple ways. Currently, the supported options are username, which matches the endpoint or aor id This option is a comma separated list of methods the endpoint can be identified.

Info
titleNote

This option controls both how an endpoint is matched for incoming traffic and also how an AOR is determined if a registration occurs. You must list at least one method that also matches for AORs or the registration will fail.

  • username - Matches the endpoint or AOR ID based on the username and domain in the From header (or To header for

...

  • AORs). If an exact match on both username and domain/realm fails, the match is retried with just the username.
  • auth_username - Matches the endpoint or AOR ID based on the username and realm in the Authentication header.

...

  • If an exact match on both username and domain/realm fails, the match

...

  • is retried with just the username.
    Info
    titleNote

...

  • This method of identification has some security considerations because an Authentication header is not present on the first message of a dialog when digest authentication is used. The client can't generate it until the server sends the challenge in a 401 response. Since Asterisk normally sends a security event when an incoming request can't be matched to an endpoint, using

...

  • this method requires that the security event be deferred until a request is received with the Authentication header and only generated if the username doesn't result in a match. This may result in a delay before an attack is recognized. You can control how many unmatched requests are received from a single ip address before a security event is generated using the unidentified_request parameters in the "global" configuration object.

...

titleNote

...

  • ip - Matches the endpoint based on the source IP address.
    This method of identification is not

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  • configured here but simply allowed by this configuration option. See the documentation for the identify configuration section for more details on

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  • this method of endpoint identification.

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  • header - Matches the endpoint based on a configured SIP header value.
    This method of identification is not configured here but simply allowed by this configuration option. See the documentation for the identify configuration section

...

  • username
  • auth_username
  • for more details on this method of endpoint identification.

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endpoint_redirect_method
endpoint_redirect_method

...

Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state changes happen for any of the specified mailboxes. More than one mailbox can be specified with a comma-delimited string. app_voicemail mailboxes must be specified as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by external sources, such as through the res_mwi_external _mwi module, you must specify strings supported by the external system.

For endpoints that SUBSCRIBE for MWI, use the mailboxes option in your AOR configuration.

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endpoint_outbound_auth
endpoint_outbound_auth

outbound_auth

This is a comma-delimited list of auth sections defined in pjsip.conf used to respond to outbound connection authentication challenges.

Info
titleNote

Using the same auth section for inbound and outbound authentication is not recommended. There is a difference in meaning for an empty realm setting between inbound and outbound authentication uses. See the auth realm description for details.

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endpoint_rewrite_contact
endpoint_rewrite_contact

...

On inbound SIP messages from this endpoint, the Contact header or an appropriate Record-Route header will be changed to have the source IP address and port. This option does not affect outbound messages sent to this endpoint. This option helps servers communicate with endpoints that are behind NATs. This option also helps reuse reliable transport connections such as TCP and TLS.

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endpoint_rpid_immediate
endpoint_rpid_immediate

...

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endpoint_timers_min_se
endpoint_timers_min_se

timers_min_se

Minimium Minimum session timer expiration period. Time in seconds.

...

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endpoint_timers_sess_expires
endpoint_timers_sess_expires

timers_sess_expires

Maximium Maximum session timer expiration period. Time in seconds.

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endpoint_transport
endpoint_transport

transport

This will set the desired force the endpoint to use the specified transport configuration to send SIP data through.

...

messages. You need to already know what kind of transport (UDP/TCP/IPv4/etc) the endpoint device will use.

Info
titleWarningNote

Not specifying a transport will DEFAULT to select the first configured transport in pjsip.conf which is valid for compatible with the URI we are trying to contact.

...

  • none - No error correction should be used.
  • fec - Forward error correction should be used.
  • redundancy - Redundacy Redundancy error correction should be used.

...

This option only applies if media_encryption is set to dtls.

It can be one of the following values:

  • no - meaning no verificaton is done.
  • fingerprint - meaning to verify the remote fingerprint.
  • certificate - meaning to verify the remote certificate.
  • yes - meaning to verify both the remote fingerprint and certificate.

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endpoint_dtls_rekey
endpoint_dtls_rekey

...

If specified, incoming SUBSCRIBE requests will be searched for the matching extension in the indicated context. If no subscribe_context is specified, then the context setting is used.

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endpoint_contact_user
endpoint_contact_user

contact_user

On outbound requests, force the user portion of the Contact header to this value.

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endpoint_asymmetric_rtp_codec
endpoint_asymmetric_rtp_codec

asymmetric_rtp_codec

When set to "yes" the codec in use for sending will be allowed to differ from that of the received one. PJSIP will not automatically switch the sending one to the receiving one.

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endpoint_rtcp_mux
endpoint_rtcp_mux

rtcp_mux

With this option enabled, Asterisk will attempt to negotiate the use of the "rtcp-mux" attribute on all media streams. This will result in RTP and RTCP being sent and received on the same port. This shifts the demultiplexing logic to the application rather than the transport layer. This option is useful when interoperating with WebRTC endpoints since they mandate this option's use.

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endpoint_refer_blind_progress
endpoint_refer_blind_progress

refer_blind_progress

Some SIP phones (Mitel/Aastra, Snom) expect a sip/frag "200 OK" after REFER has been accepted. If set to no then asterisk will not send the progress details, but immediately will send "200 OK".

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endpoint_notify_early_inuse_ringing
endpoint_notify_early_inuse_ringing

notify_early_inuse_ringing

Control whether dialog-info subscriptions get 'early' state on Ringing when already INUSE.

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endpoint_incoming_mwi_mailbox
endpoint_incoming_mwi_mailbox

incoming_mwi_mailbox

If an MWI NOTIFY is received from this endpoint, this mailbox will be used when notifying other modules of MWI status changes. If not set, incoming MWI NOTIFYs are ignored.

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endpoint_follow_early_media_fork
endpoint_follow_early_media_fork

follow_early_media_fork

On outgoing calls, if the UAS responds with different SDP attributes on subsequent 18X or 2XX responses (such as a port update) AND the To tag on the subsequent response is different than that on the previous one, follow it. This usually happens when the INVITE is forked to multiple UASs and more than one sends an SDP answer.

Info
titleNote

This option must also be enabled in the system section for it to take effect here.

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endpoint_accept_multiple_sdp_answers
endpoint_accept_multiple_sdp_answers

accept_multiple_sdp_answers

On outgoing calls, if the UAS responds with different SDP attributes on non-100rel 18X or 2XX responses (such as a port update) AND the To tag on the subsequent response is the same as that on the previous one, process the updated SDP. This can happen when the UAS needs to change ports for some reason such as using a separate port for custom ringback.

Info
titleNote

This option must also be enabled in the system section for it to take effect here.

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endpoint_suppress_q850_reason_headers
endpoint_suppress_q850_reason_headers

suppress_q850_reason_headers

Some devices can't accept multiple Reason headers and get confused when both 'SIP' and 'Q.850' Reason headers are received. This option allows the 'Q.850' Reason header to be suppressed.

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endpoint_ignore_183_without_sdp
endpoint_ignore_183_without_sdp

ignore_183_without_sdp

Certain SS7 internetworking scenarios can result in a 183 to be generated for reasons other than early media. Forwarding this 183 can cause loss of ringback tone. This flag emulates the behavior of chan_sip and prevents these 183 responses from being forwarded.

auth

Authentication type

Configuration Option Reference

Option Name

Type

Default Value

Regular Expression

Description

auth_type

Custom

userpass

false

Authentication type

nonce_lifetime

Unsigned Integer

32

false

Lifetime of a nonce associated with this authentication config.

md5_cred

String

 

false

MD5 Hash used for authentication.

password

String

 

false

PlainText Plain text password used for authentication.

realm

String

 

false

SIP realm for endpoint

type

None

 

false

Must be 'auth'

username

String

 

false

Username to use for account

...

Only used when auth_type is userpass.

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auth_realm
auth_realm

realm

The treatment of this value depends upon how the authentication object is used.

When used as an inbound authentication object, the realm is sent as part of the challenge so the peer can know which key to use when responding. An empty value will use the global section's default_realm value when issuing a challenge.

When used as an outbound authentication object, the realm is matched with the received challenge realm to determine which authentication object to use when responding to the challenge. An empty value matches any challenging realm when determining which authentication object matches a received challenge.

Info
titleNote

Using the same auth section for inbound and outbound authentication is not recommended. There is a difference in meaning for an empty realm setting between inbound and outbound authentication uses.

domain_alias

Domain Alias

Configuration Option Reference

...

Option Name

Type

Default Value

Regular Expression

Description

async_operations

Unsigned Integer

1

false

Number of simultaneous Asynchronous Operations

bind

Custom

 

false

IP Address and optional port to bind to for this transport

ca_list_file

Custom

 

false

File containing a list of certificates to read (TLS ONLY, not WSS)

ca_list_path

Custom

 

false

Path to directory containing a list of certificates to read (TLS ONLY, not WSS)

cert_file

Custom

 

false

Certificate file for endpoint (TLS ONLY, not WSS)

cipher

Custom

 

false

Preferred cryptography cipher names (TLS ONLY, not WSS)

domain

String

 

false

Domain the transport comes from

external_media_address

String

 

false

External IP address to use in RTP handling

external_signaling_address

String

 

false

External address for SIP signalling

external_signaling_port

Unsigned Integer

0

false

External port for SIP signalling

method

Custom

 

false

Method of SSL transport (TLS ONLY, not WSS)

local_net

Custom

 

false

Network to consider local (used for NAT purposes).

password

String

 

false

Password required for transport

priv_key_file

Custom

 

false

Private key file (TLS ONLY, not WSS)

protocol

Custom

udp

false

Protocol to use for SIP traffic

require_client_cert

Custom

 

false

Require client certificate (TLS ONLY, not WSS)

type

Custom

 

false

Must be of type 'transport'.

verify_client

Custom

 

false

Require verification of client certificate (TLS ONLY, not WSS)

verify_server

Custom

 

false

Require verification of server certificate (TLS ONLY, not WSS)

tos

Custom

0

false

Enable TOS for the signalling sent over this transport

cos

Unsigned Integer

0

false

Enable COS for the signalling sent over this transport

websocket_write_timeout

Integer

100

false

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

allow_reload

Boolean

no

false

Allow this transport to be reloaded.

symmetric_transport

Boolean

no

false

Use the same transport for outgoing requests as incoming ones.

Configuration Option Descriptions

...

  • default - The default as defined by PJSIP. This is currently TLSv1, but may change with future releases.
  • unspecified - This option is equivalent to setting 'default'
  • tlsv1
  • tlsv1_1
  • tlsv1_2
  • sslv2
  • sslv3
  • sslv23

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transport_local_net
transport_local_net

...

Allow this transport to be reloaded when res_pjsip is reloaded. This option defaults to "no" because reloading a transport may disrupt in-progress calls.

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transport_symmetric_transport
transport_symmetric_transport

symmetric_transport

When a request from a dynamic contact comes in on a transport with this option set to 'yes', the transport name will be saved and used for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's saved as a contact uri parameter named 'x-ast-txp' and will display with the contact uri in CLI, AMI, and ARI output. On the outgoing request, if a transport wasn't explicitly set on the endpoint AND the request URI is not a hostname, the saved transport will be used and the 'x-ast-txp' parameter stripped from the outgoing packet.

contact

A way of creating an aliased name to a SIP URI

...

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 challenge response 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.

endpoint

String

 

false

Endpoint name

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.

prune_on_boot

Boolean

no

false

A contact that cannot survive a restart/boot.

Configuration Option Descriptions

...

qualify_timeout

If the contact doesn't repond respond to the OPTIONS request before the timeout, the contact is marked unavailable. If 0 no timeout. Time in fractional seconds.

...

If true and a qualify request receives a challenge or authenticate response then authentication is attempted before declaring the contact available.

Info
titleNote

This option does nothing as we will always complete the challenge response authentication if the qualify request is challenged.

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contact_outbound_proxy
contact_outbound_proxy

...

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

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contact_prune_on_boot
contact_prune_on_boot

prune_on_boot

The option is set if the incoming SIP REGISTER contact is rewritten on a reliable transport and is not intended to be configured manually.

aor

The configuration for a location of an endpoint

...

Option Name

Type

Default Value

Regular Expression

Description

contact

Custom

 

false

Permanent contacts assigned to AoR

default_expiration

Unsigned Integer

3600

false

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

mailboxes

String

 

false

Allow subscriptions for the specified mailbox(es)

voicemail_extension

Custom

 

false

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

maximum_expiration

Unsigned Integer

7200

false

Maximum time to keep an AoR

max_contacts

Unsigned Integer

0

false

Maximum number of contacts that can bind to an AoR

minimum_expiration

Unsigned Integer

60

false

Minimum keep alive time for an AoR

remove_existing

Boolean

no

false

Determines whether new contacts replace existing ones.

type

None

 

false

Must be of type 'aor'.

qualify_frequency

Unsigned Integer

0

false

Interval at which to qualify an AoR

qualify_timeout

Double

3.0

false

Timeout for qualify

authenticate_qualify

Boolean

no

false

Authenticates a qualify request challenge response if needed

outbound_proxy

String

 

false

Outbound proxy used when sending OPTIONS request

support_path

Boolean

no

false

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

...

This option applies when an external entity subscribes to an AoR for Message Waiting Indications. The mailboxes specified will be subscribed to. More than one mailbox can be specified with a comma-delimited string. app_voicemail mailboxes must be specified as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by external sources, such as through the res_mwi_external _mwi module, you must specify strings supported by the external system.

...

Anchor
aor_maximum_expiration
aor_maximum_expiration

maximum_expiration

Maximium Maximum time to keep a peer with explicit expiration. Time in seconds.

...

Maximum number of contacts that can associate with this AoR. This value does not affect the number of contacts that can be added with the "contact" option. It only limits contacts added through external interaction, such as registration.

Info
titleNote

The rewrite_contact option registers the source address as the contact address to help with NAT and reusing connection oriented transports such as TCP and TLS. Unfortunately, refreshing a registration may register a different contact address and exceed max_contacts. The remove_existing option can help by removing the soonest to expire contact(s) over max_contacts which is likely the old rewrite_contact contact source address being refreshed.

Info
titleNote

This should be set to 1 and remove_existing set to yes if you wish to stick with the older chan_sip behaviour.

...

Minimum time to keep a peer with an explict explicit expiration. Time in seconds.

...

On receiving a new registration to the AoR should it remove enough existing contacts not added or updated by the registration to satisfy max_contacts? Any removed contacts will expire the soonest.

Info
titleNote

The rewrite_contact option registers the source address as the contact address to help with NAT and reusing connection oriented transports such as TCP and TLS. Unfortunately, refreshing a registration may register a different contact address and exceed max_contacts. The remove_existing

...

option can help by removing the soonest to expire contact(s) over max_contacts which is likely the old rewrite_contact contact source address being refreshed.

Info
titleNote

This should be set to yes and max_contacts set to 1 if you wish to stick with the older chan_sip behaviour.

...

qualify_timeout

If the contact doesn't repond respond to the OPTIONS request before the timeout, the contact is marked unavailable. If 0 no timeout. Time in fractional seconds.

...

If true and a qualify request receives a challenge or authenticate response then authentication is attempted before declaring the contact available.

Info
titleNote

This option does nothing as we will always complete the challenge response authentication if the qualify request is challenged.

Anchor
aor_outbound_proxy
aor_outbound_proxy

...

Option Name

Type

Default Value

Regular Expression

Description

timer_t1

Unsigned Integer

500

false

Set transaction timer T1 value (milliseconds).

timer_b

Unsigned Integer

32000

false

Set transaction timer B value (milliseconds).

compact_headers

Boolean

no

false

Use the short forms of common SIP header names.

threadpool_initial_size

Unsigned Integer

0

false

Initial number of threads in the res_pjsip threadpool.

threadpool_auto_increment

Unsigned Integer

5

false

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

threadpool_idle_timeout

Unsigned Integer

60

false

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

threadpool_max_size

Unsigned Integer

50

false

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

disable_tcp_switch

Boolean

yes

false

Disable automatic switching from UDP to TCP transports.

follow_early_media_fork

Boolean

yes

false

Follow SDP forked media when To tag is different

accept_multiple_sdp_answers

Boolean

no

false

Follow SDP forked media when To tag is the same

disable_rport

Boolean

no

false

Disable the use of rport in outgoing requests.

type

None

 

false

Must be of type 'system' UNLESS the object name is 'system'.

Configuration Option Descriptions

...

Disable automatic switching from UDP to TCP transports if outgoing request is too large. See RFC 3261 section 18.1.1.

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system_follow_early_media_fork
system_follow_early_media_fork

follow_early_media_fork

On outgoing calls, if the UAS responds with different SDP attributes on subsequent 18X or 2XX responses (such as a port update) AND the To tag on the subsequent response is different than that on the previous one, follow it.

Info
titleNote

This option must also be enabled on endpoints that require this functionality.

Anchor
system_accept_multiple_sdp_answers
system_accept_multiple_sdp_answers

accept_multiple_sdp_answers

On outgoing calls, if the UAS responds with different SDP attributes on non-100rel 18X or 2XX responses (such as a port update) AND the To tag on the subsequent response is the same as that on the previous one, process the updated SDP.

Info
titleNote

This option must also be enabled on endpoints that require this functionality.

Anchor
system_disable_rport
system_disable_rport

disable_rport

Remove "rport" parameter from the outgoing requests.

global

Options that apply globally to all SIP communications

...

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 90

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' UNLESS the object name is 'global'.

user_agent

String

Asterisk PBX GIT-13-bf2135913.15.0-rc1-2813-gca8f0b1

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 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

Unsigned Integer

500

false

MWI taskprocessor high water alert trigger level.

mwi_tps_queue_low

Integer

-1

false

MWI taskprocessor low water clear alert level.

mwi_disable_initial_unsolicited

Boolean

no

false

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

ignore_uri_user_options

Boolean

no

false

Enable/Disable ignoring SIP URI user field options.

use_callerid_contact

Boolean

no

false

Place caller-id information into Contact header

send_contact_status_on_update_registration

Boolean

yes

false

Enable sending AMI ContactStatus event when a device refreshes its registration.

taskprocessor_overload_trigger

Custom

global

false

Trigger scope for taskprocessor overloads

norefersub

Boolean

yes

false

Advertise support for RFC4488 REFER subscription suppression

Configuration Option Descriptions

...

If disabled it can improve realtime performace performance by reducing the number of database requstsrequests.

Anchor
global_unidentified_request_period
global_unidentified_request_period

...

Info
titleNote

One of the identifiers is "auth_username" which matches on the username in an Authentication header. This method has some security considerations because an Authentication header is not present on the first message of a dialog when digest authentication is used. The client can't generate it until the server sends the challenge in a 401 response. Since Asterisk normally sends a security event when an incoming request can't be matched to an endpoint, using auth_username requires that the security event be deferred until a request is received with the Authentication header and only generated if the username doesn't result in a match. This may result in a delay before an attack is recognized. You can control how many unmatched requests are received from a single ip address before a security event is generated using the unidentified_request parameters.

Anchor
global_mwi_tps_queue_high
global_mwi_tps_queue_high

mwi_tps_queue_high

On a heavily loaded system you may need to adjust the taskprocessor queue limits. If any taskprocessor queue size reaches its high water level then pjsip will stop processing new requests until the alert is cleared. The alert clears when all alerting taskprocessor queues have dropped to their low water clear level.

Anchor
global_mwi_tps_queue_low
global_mwi_tps_queue_low

mwi_tps_queue_low

On a heavily loaded system you may need to adjust the taskprocessor queue limits. If any taskprocessor queue size reaches its high water level then pjsip will stop processing new requests until the alert is cleared. The alert clears when all alerting taskprocessor queues have dropped to their low water clear level.

Info
titleNote

Set to -1 for the low water level to be 90% of the high water level.

Anchor
global_mwi_disable_initial_unsolicited
global_mwi_disable_initial_unsolicited

mwi_disable_initial_unsolicited

When the initial unsolicited MWI notification are enabled on startup then the initial notifications get sent at startup. If you have a lot of endpoints (thousands) that use unsolicited MWI then you may want to consider disabling the initial startup notifications.

When the initial unsolicited MWI notifications are disabled on startup then the notifications will start on the endpoint's next contact update.

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global_ignore_uri_user_options
global_ignore_uri_user_options

ignore_uri_user_options

If you have this option enabled and there are semicolons in the user field of a SIP URI then the field is truncated at the first semicolon. This effectively makes the semicolon a non-usable character for PJSIP endpoint names, extensions, and AORs. This can be useful for improving compatibility with an ITSP that likes to use user options for whatever reason.

Code Block
titleExample: Sample SIP URI
linenumberstrue


sip:1235557890;phone-context=national@x.x.x.x;user=phone

Code Block
titleExample: Sample SIP URI user field
linenumberstrue


1235557890;phone-context=national

Code Block
titleExample: Sample SIP URI user field truncated
linenumberstrue


1235557890

Info
titleNote

The caller-id and redirecting number strings obtained from incoming SIP URI user fields are always truncated at the first semicolon.

Anchor
global_use_callerid_contact
global_use_callerid_contact

use_callerid_contact

This option will cause Asterisk to place caller-id information into generated Contact headers.

Anchor
global_taskprocessor_overload_trigger
global_taskprocessor_overload_trigger

taskprocessor_overload_trigger

This option specifies the trigger the distributor will use for detecting taskprocessor overloads. When it detects an overload condition, the distrubutor will stop accepting new requests until the overload is cleared.

  • global - (default) Any taskprocessor overload will trigger.
  • pjsip_only - Only pjsip taskprocessor overloads will trigger.
  • none - No overload detection will be performed.
    Warning
    titleWarning

    The "none" and "pjsip_only" options should be used with extreme caution and only to mitigate specific issues. Under certain conditions they could make things worse.

Import Version

This documentation was imported from Asterisk Version GIT-13-bf2135913.15.0-rc1-2813-gca8f0b1