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PJSIP_ENDPOINT()

Synopsis

Get information about a PJSIP endpoint

Description

Syntax

PJSIP_ENDPOINT(name,field)
Arguments

  • name - The name of the endpoint to query.

  • field - The configuration option for the endpoint to query for. Supported options are those fields on the endpoint object in pjsip.conf.

    • 100rel - Allow support for RFC3262 provisional ACK tags

    • aggregate_mwi - Condense MWI notifications into a single NOTIFY.

    • allow - Media Codec(s) to allow

    • codec_prefs_incoming_offer - Codec negotiation prefs for incoming offers.

    • codec_prefs_outgoing_offer - Codec negotiation prefs for outgoing offers.

    • codec_prefs_incoming_answer - Codec negotiation prefs for incoming answers.

    • codec_prefs_outgoing_answer - Codec negotiation prefs for outgoing answers.

    • allow_overlap - Enable RFC3578 overlap dialing support.

    • overlap_context - Dialplan context to use for RFC3578 overlap dialing.

    • 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 - Mitigation of direct media (re)INVITE glare

    • direct_media_method - Direct Media method type

    • trust_connected_line - Accept Connected Line updates from this endpoint

    • send_connected_line - Send Connected Line updates to this endpoint

    • connected_line_method - Connected line method type

    • direct_media - Determines whether media may flow directly between endpoints.

    • disable_direct_media_on_nat - Disable direct media session refreshes when NAT obstructs the media session

    • disallow - Media Codec(s) to disallow

    • dtmf_mode - 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 - Force use of return port

    • ice_support - Enable the ICE mechanism to help traverse NAT

    • identify_by - Way(s) for the 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 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 - Allow use of IPv6 for RTP traffic

    • rtp_symmetric - Enforce that RTP must be symmetric

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

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

    • send_pai - Send the P-Asserted-Identity header

    • send_rpid - Send the Remote-Party-ID header

    • rpid_immediate - Immediately send connected line updates on unanswered incoming calls.

    • tenantid - The tenant ID for this endpoint.

    • timers_min_se - Minimum session timers expiration period

    • timers - Session timers for SIP packets

    • timers_sess_expires - Maximum session timer expiration period

    • transport - Explicit transport configuration to use

    • trust_id_inbound - Accept identification information received from this endpoint

    • trust_id_outbound - Send private identification details to the endpoint.

    • type - Must be of type 'endpoint'.

    • use_ptime - Use Endpoint's requested packetization interval

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

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

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

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

    • g726_non_standard - Force g.726 to use AAL2 packing order when negotiating g.726 audio

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

    • t38_udptl - Whether T.38 UDPTL support is enabled or not

    • t38_udptl_ec - T.38 UDPTL error correction method

    • t38_udptl_maxdatagram - T.38 UDPTL maximum datagram size

    • fax_detect - 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 - Whether NAT support is enabled on UDPTL sessions

    • t38_udptl_ipv6 - Whether IPv6 is used for UDPTL Sessions

    • t38_bind_udptl_to_media_address - Bind the UDPTL instance to the media_adress

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

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

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

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

    • allow_transfer - Determines whether SIP REFER transfers are allowed for this endpoint

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

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

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

    • sdp_session - 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 - Determines if endpoint is allowed to initiate subscriptions with Asterisk.

    • sub_min_expiry - 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 use 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_auto_generate_cert - Whether or not to automatically generate an ephemeral X.509 certificate

    • 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.

    • preferred_codec_only - Respond to a SIP invite with the single most preferred codec (DEPRECATED)

    • incoming_call_offer_pref - Preferences for selecting codecs for an incoming call.

    • outgoing_call_offer_pref - Preferences for selecting codecs for an outgoing call.

    • rtp_keepalive - Number of seconds between RTP comfort noise keepalive packets.

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

    • rtp_timeout_hold - 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 - Allow the sending and receiving RTP codec to differ

    • rtcp_mux - Enable RFC 5761 RTCP multiplexing on the RTP port

    • refer_blind_progress - Whether to notifies all the progress details on blind transfer

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

    • max_audio_streams - The maximum number of allowed audio streams for the endpoint

    • max_video_streams - The maximum number of allowed video streams for the endpoint

    • bundle - Enable RTP bundling

    • webrtc - Defaults and enables some options that are relevant to WebRTC

    • incoming_mwi_mailbox - Mailbox name to use when incoming MWI NOTIFYs are received

    • follow_early_media_fork - Follow SDP forked media when To tag is different

    • accept_multiple_sdp_answers - Accept multiple SDP answers on non-100rel responses

    • suppress_q850_reason_headers - Suppress Q.850 Reason headers for this endpoint

    • ignore_183_without_sdp - Do not forward 183 when it doesn't contain SDP

    • stir_shaken - Enable STIR/SHAKEN support on this endpoint

    • stir_shaken_profile - STIR/SHAKEN profile containing additional configuration options

    • allow_unauthenticated_options - Skip authentication when receiving OPTIONS requests

    • security_negotiation - The kind of security agreement negotiation to use. Currently, only mediasec is supported.

    • security_mechanisms - List of security mechanisms supported.

    • geoloc_incoming_call_profile - Geolocation profile to apply to incoming calls

    • geoloc_outgoing_call_profile - Geolocation profile to apply to outgoing calls

    • send_aoc - Send Advice-of-Charge messages

    • suppress_moh_on_sendonly - Suppress playing MOH to party A if party B sends "sendonly" or "inactive" in an SDP

Generated Version

This documentation was generated from Asterisk branch 20 using version GIT