Get information about a PJSIP endpoint
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
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
allow_unauthenticated_options- Skip authentication when receiving OPTIONS requests
This documentation was imported from Asterisk Version GIT-16-39824c7