PJSIP_ENDPOINT()¶
Synopsis¶
Get information about a PJSIP endpoint
Description¶
Syntax¶
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
-
Generated Version¶
This documentation was generated from Asterisk branch 22 using version GIT