Allow support for RFC3262 provisional ACK tags
Condense MWI notifications into a single NOTIFY.
Media Codec(s) to allow
Enable RFC3578 overlap dialing support.
AoR(s) to be used with the endpoint
Authentication Object(s) associated with the endpoint
CallerID information for the endpoint
Default privacy level
Internal id_tag for the endpoint
Dialplan context for inbound sessions
Mitigation of direct media (re)INVITE glare
Direct Media method type
Connected line method type
Determines whether media may flow directly between endpoints.
Disable direct media session refreshes when NAT obstructs the media session
Media Codec(s) to disallow
IP address used in SDP for media handling
Bind the RTP instance to the media_address
Force use of return port
Enable the ICE mechanism to help traverse NAT
Way(s) for Endpoint to be identified
How redirects received from an endpoint are handled
NOTIFY the endpoint when state changes for any of the specified mailboxes
An MWI subscribe will replace sending unsolicited NOTIFYs
The voicemail extension to send in the NOTIFY Message-Account header
Default Music On Hold class
Authentication object(s) used for outbound requests
Full SIP URI of the outbound proxy used to send requests
Allow Contact header to be rewritten with the source IP address-port
Allow use of IPv6 for RTP traffic
Enforce that RTP must be symmetric
Send the Diversion header, conveying the diversion information to the called user agent
Send the P-Asserted-Identity header
Send the Remote-Party-ID header
Immediately send connected line updates on unanswered incoming calls.
Minimum session timers expiration period
Session timers for SIP packets
Maximum session timer expiration period
Desired transport configuration
Accept identification information received from this endpoint
Send private identification details to the endpoint.
Must be of type 'endpoint'.
Use Endpoint's requested packetisation interval
Determines whether res_pjsip will use and enforce usage of AVPF for this endpoint.
Determines whether res_pjsip will use and enforce usage of AVP, regardless of the RTP profile in use for this endpoint.
Determines whether res_pjsip will use the media transport received in the offer SDP in the corresponding answer SDP.
Determines whether res_pjsip will use and enforce usage of media encryption for this endpoint.
Determines whether encryption should be used if possible but does not terminate the session if not achieved.
Force g.726 to use AAL2 packing order when negotiating g.726 audio
Determines whether chan_pjsip will indicate ringing using inband progress.
The numeric pickup groups for a channel.
The numeric pickup groups that a channel can pickup.
The named pickup groups for a channel.
The named pickup groups that a channel can pickup.
The number of in-use channels which will cause busy to be returned as device state
Whether T.38 UDPTL support is enabled or not
T.38 UDPTL error correction method
T.38 UDPTL maximum datagram size
Whether CNG tone detection is enabled
How long into a call before fax_detect is disabled for the call
Whether NAT support is enabled on UDPTL sessions
Whether IPv6 is used for UDPTL Sessions
Set which country's indications to use for channels created for this endpoint.
Set the default language to use for channels created for this endpoint.
Determines whether one-touch recording is allowed for this endpoint.
The feature to enact when one-touch recording is turned on.
The feature to enact when one-touch recording is turned off.
Name of the RTP engine to use for channels created for this endpoint
Determines whether SIP REFER transfers are allowed for this endpoint
Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number
Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side
String placed as the username portion of an SDP origin (o=) line.
String used for the SDP session (s=) line.
DSCP TOS bits for audio streams
DSCP TOS bits for video streams
Priority for audio streams
Priority for video streams
Determines if endpoint is allowed to initiate subscriptions with Asterisk.
The minimum allowed expiry time for subscriptions initiated by the endpoint.
Username to use in From header for requests to this endpoint.
Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.
Domain to user in From header for requests to this endpoint.
Verify that the provided peer certificate is valid
Interval at which to renegotiate the TLS session and rekey the SRTP session
Path to certificate file to present to peer
Path to private key for certificate file
Cipher to use for DTLS negotiation
Path to certificate authority certificate
Path to a directory containing certificate authority certificates
Whether we are willing to accept connections, connect to the other party, or both.
Type of hash to use for the DTLS fingerprint in the SDP.
Determines whether 32 byte tags should be used instead of 80 byte tags.
Variable set on a channel involving the endpoint.
Context to route incoming MESSAGE requests to.
An accountcode to set automatically on any channels created for this endpoint.
Number of seconds between RTP comfort noise keepalive packets.
Maximum number of seconds without receiving RTP (while off hold) before terminating call.
Maximum number of seconds without receiving RTP (while on hold) before terminating call.
List of IP ACL section names in acl.conf
List of IP addresses to deny access from
List of IP addresses to permit access from
List of Contact ACL section names in acl.conf
List of Contact header addresses to deny
List of Contact header addresses to permit
Context for incoming MESSAGE requests.
Force the user on the outgoing Contact header to this value.
Allow the sending and receiving RTP codec to differ
Enable RFC 5761 RTCP multiplexing on the RTP port
Whether to notifies all the progress details on blind transfer
Configuration Option Descriptions
With this option enabled, Asterisk will attempt to negotiate the use of the "rtcp-mux" attribute on all media streams. This will result in RTP and RTCP being sent and received on the same port. This shifts the demultiplexing logic to the application rather than the transport layer. This option is useful when interoperating with WebRTC endpoints since they mandate this option's use.
Some SIP phones (Mitel/Aastra, Snom) expect a sip/frag "200 OK" after REFER has been accepted. If set to
no then asterisk will not send the progress details, but immediately will send "200 OK".
Configuration Option Reference
This documentation was imported from Asterisk Version GIT-14-bdc05146fa2e0a