Allow support for RFC3262 provisional ACK tags
Condense MWI notifications into a single NOTIFY.
Media Codec(s) to allow
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
Default Music On Hold class
Authentication object used for outbound requests
Proxy through which to send requests, a full SIP URI must be provided
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
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
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.
Be aware that the
If media_address is specified, this option causes the RTP instance to be bound to the specified ip address which causes the packets to be sent from that address.
Number of simultaneous Asynchronous Operations
IP Address and optional port to bind to for this transport
File containing a list of certificates to read (TLS ONLY)
Path to directory containing a list of certificates to read (TLS ONLY)
Certificate file for endpoint (TLS ONLY)
Preferred cryptography cipher names (TLS ONLY)
Domain the transport comes from
External IP address to use in RTP handling
External address for SIP signalling
External port for SIP signalling
Method of SSL transport (TLS ONLY)
Network to consider local (used for NAT purposes).
Password required for transport
Private key file (TLS ONLY)
Protocol to use for SIP traffic
Require client certificate (TLS ONLY)
Must be of type 'transport'.
Require verification of client certificate (TLS ONLY)
Require verification of server certificate (TLS ONLY)
Enable TOS for the signalling sent over this transport
Enable COS for the signalling sent over this transport
The timeout (in milliseconds) to set on WebSocket connections.
Set transaction timer T1 value (milliseconds).
Set transaction timer B value (milliseconds).
Use the short forms of common SIP header names.
Initial number of threads in the res_pjsip threadpool.
The amount by which the number of threads is incremented when necessary.
Number of seconds before an idle thread should be disposed of.
Maximum number of threads in the res_pjsip threadpool. A value of 0 indicates no maximum.
Disable automatic switching from UDP to TCP transports.
Must be of type 'system'.
Value used in Max-Forwards header for SIP requests.
The interval (in seconds) to send keepalives to active connection-oriented transports.
The maximum amount of time from startup that qualifies should be attempted on all contacts. If greater than the qualify_frequency for an aor, qualify_frequency will be used instead.
Must be of type 'global'.
Value used in User-Agent header for SIP requests and Server header for SIP responses.
When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us.
Endpoint to use when sending an outbound request to a URI without a specified endpoint.
Enable/Disable SIP debug logging. Valid options include yes
no or a host address
The order by which endpoint identifiers are processed and checked. Identifier names are usually derived from and can be found in the endpoint identifier module itself (res_pjsip_endpoint_identifier_*)
When Asterisk generates an outgoing SIP request, the From header username will be set to this value if there is no better option (such as CallerID) to be used.
This documentation was imported from Asterisk Version GIT-13-25ec63aM8c15f30