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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
`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`	`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.
`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
`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.
`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.

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Info

title	Note

Be aware that the external_media_address option, set in Transport configuration, can also affect the final media address used in the SDP.

Anchor

	endpoint_bind_rtp_to_media_address
	endpoint_bind_rtp_to_media_address

bind_rtp_to_media_address

If media_address is specified, this option causes the RTP instance to be bound to the specified ip address which causes the packets to be sent from that address.

Anchor

	endpoint_identify_by
	endpoint_identify_by

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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`	`String` `Custom`		`false`	File containing a list of certificates to read (TLS ONLY)
`ca_list_path`	`String` `Custom`		`false`	Path to directory containing a list of certificates to read (TLS ONLY)
`cert_file`	`String` `Custom`		`false`	Certificate file for endpoint (TLS ONLY)
`cipher`	`Custom`		`false`	Preferred cryptography cipher names (TLS ONLY)
`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)
`local_net`	`Custom`		`false`	Network to consider local (used for NAT purposes).
`password`	`String`		`false`	Password required for transport
`priv_key_file`	`String` `Custom`		`false`	Private key file (TLS ONLY)
`protocol`	`Custom`	`udp`	`false`	Protocol to use for SIP traffic
`require_client_cert`	`Custom`		`false`	Require client certificate (TLS ONLY)
`type`	`None` `Custom`		`false`	Must be of type 'transport'.
`verify_client`	`Custom`		`false`	Require verification of client certificate (TLS ONLY)
`verify_server`	`Custom`		`false`	Require verification of server certificate (TLS ONLY)
`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.

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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`	`0` `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.
`type`	`None`		`false`	Must be of type 'system'.

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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.
`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.
`type`	`None`		`false`	Must be of type 'global'.
`user_agent`	`String`	`Asterisk PBX GIT-13-25ec63aM8c15f30`	`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.
`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_*)
`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.

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This documentation was imported from Asterisk Version GIT-13-25ec63aM8c15f30

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Key

bind_rtp_to_media_address