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Option Name | Type | Default Value | Regular Expression | Description |
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| | | 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 |
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| Media Codec(s) to disallow |
| | | DTMF mode | |
|
| | IP address used in SDP for media handling | |
| | | | 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. | |
| | | 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. |
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On inbound SIP messages from this endpoint, the Contact header will be changed to have the source IP address and port. This option does not affect outbound messages send to this endpoint.
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rpid_immediate
When enabled, immediately send 180 Ringing or 183 Progress response messages to the caller if the connected line information is updated before the call is answered. This can send a 180 Ringing response before the call has even reached the far end. The caller can start hearing ringback before the far end even gets the call. Many phones tend to grab the first connected line information and refuse to update the display if it changes. The first information is not likely to be correct if the call goes to an endpoint not under the control of this Asterisk box.
When disabled, a connected line update must wait for another reason to send a message with the connected line information to the caller before the call is answered. You can trigger the sending of the information by using an appropriate dialplan application such as Ringing.
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Option Name | Type | Default Value | Regular Expression | Description | |
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| | | | Value used in Max-Forwards header for SIP requests. | |
| | | | The interval (in seconds) to send keepalives to active connection-oriented transports. | |
| |
| | Must be of type 'global'. | |
| | | | Value used in User-Agent header for SIP requests and Server header for SIP responses. | |
| | | | 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_*) |
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This documentation was imported from Asterisk Version SVN-branch-13-r433222r433694