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
    

  • allow_overlap - Enable RFC3578 overlap dialing support.
    

  • 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.
    

  • 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 user 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 rather than advertising all joint codec capabilities. This limits the other side's codec choice to exactly what we prefer.
    

  • 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
    

Generated Version¶

This documentation was generated from Asterisk branch 16 using version GIT