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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 |
| | | | 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 |
| | | DTMF mode | |
|
| | 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 the 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 packetization 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 |
| | | | 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 | |
| | | Whether to notifies dialog-info 'early' on InUse&Ringing state | |
|
| | Mailbox name to use when incoming MWI NOTIFYs are received |
Configuration Option Descriptions
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identify_by
Endpoints and aors AORs can be identified in multiple ways. Currently, the supported options are username
, which matches the endpoint or aor id This option is a comma separated list of methods the endpoint can be identified.
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This option controls both how an endpoint is matched for incoming traffic and also how an AOR is determined if a registration occurs. You must list at least one method that also matches for AORs or the registration will fail. |
username
- Matches the endpoint or AOR ID based on the username and domain in the From header (or To header for
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- AORs). If an exact match on both username and domain/realm fails, the match is retried with just the username.
auth_username
- Matches the endpoint or AOR ID based on the username and realm in the Authentication header.
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- If an exact match on both username and domain/realm fails, the match
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- is retried with just the username.
Info title Note
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This method of identification has some security considerations because an Authentication header is not present on the first message of a dialog when digest authentication is used. The client can't generate it until the server sends the challenge in a 401 response. Since Asterisk normally sends a security event when an incoming request can't be matched to an endpoint, using
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this method requires that the security event be deferred until a request is received with the Authentication header and only generated if the username doesn't result in a match. This may result in a delay before an attack is recognized. You can control how many unmatched requests are received from a single ip address before a security event is generated using the
unidentified_request
parameters in the "global" configuration object.
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title | Note |
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ip
- Matches the endpoint based on the source IP address.
This method of identification is not
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- configured here but simply allowed by this configuration option. See the documentation for the
identify
configuration section for more details on
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- this method of endpoint identification.
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header
- Matches the endpoint based on a configured SIP header value.
This method of identification is not configured here but simply allowed by this configuration option. See the documentation for theidentify
configuration section
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username
auth_username
- for more details on this method of endpoint identification.
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|
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On inbound SIP messages from this endpoint, the Contact header or an appropriate Record-Route header will be changed to have the source IP address and port. This option does not affect outbound messages sent to this endpoint. This option helps servers communicate with endpoints that are behind NATs. This option also helps reuse reliable transport connections such as TCP and TLS.
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timers_min_se
Minimium Minimum session timer expiration period. Time in seconds.
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timers_sess_expires
Maximium Maximum session timer expiration period. Time in seconds.
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none
- No error correction should be used.fec
- Forward error correction should be used.redundancy
- Redundacy Redundancy error correction should be used.
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Control whether dialog-info subscriptions get 'early' state on Ringing when already INUSE.
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incoming_mwi_mailbox
If an MWI NOTIFY is received from this endpoint, this mailbox will be used when notifying other modules of MWI status changes. If not set, incoming MWI NOTIFYs are ignored.
auth
Authentication type
Configuration Option Reference
Option Name | Type | Default Value | Regular Expression | Description |
---|---|---|---|---|
| | | Authentication type | |
| | | | Lifetime of a nonce associated with this authentication config. |
|
| | MD5 Hash used for authentication. | |
|
| | PlainText Plain text password used for authentication. | |
|
| | SIP realm for endpoint | |
| |
| | Must be 'auth' |
| |
| | Username to use for account |
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Option Name | Type | Default Value | Regular Expression | Description |
---|---|---|---|---|
| | | | 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. | |
| | | Allow this transport to be reloaded. | |
| | | Use the same transport for outgoing reqests requests as incoming ones. |
Configuration Option Descriptions
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default
- The default as defined by PJSIP. This is currently TLSv1, but may change with future releases.unspecified
- This option is equivalent to setting 'default'tlsv1
tlsv1_1
tlsv1_2
sslv2
sslv3
sslv23
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Option Name | Type | Default Value | Regular Expression | Description |
---|---|---|---|---|
| |
| | Must be of type 'contact'. |
| |
| | SIP URI to contact peer |
|
| | Time to keep alive a contact | |
| | | Interval at which to qualify a contact | |
| | | Timeout for qualify | |
| | | Authenticates a qualify request challenge response if needed | |
|
| | Outbound proxy used when sending OPTIONS request | |
| |
| | Stored Path vector for use in Route headers on outgoing requests. |
|
| | User-Agent header from registration. | |
|
| | Endpoint name | |
|
| | Asterisk Server name | |
|
| | IP-address of the last Via header from registration. | |
| | | IP-port of the last Via header from registration. | |
|
| | Call-ID header from registration. | |
| | | A contact that cannot survive a restart/boot. |
Configuration Option Descriptions
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qualify_timeout
If the contact doesn't repond respond to the OPTIONS request before the timeout, the contact is marked unavailable. If 0
no timeout. Time in fractional seconds.
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If true and a qualify request receives a challenge or authenticate response then authentication is attempted before declaring the contact available.
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This option does nothing as we will always complete the challenge response authentication if the qualify request is challenged. |
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The Call-ID header is automatically stored based on data present in incoming SIP REGISTER requests and is not intended to be configured manually.
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prune_on_boot
The option is set if the incoming SIP REGISTER contact is rewritten on a reliable transport and is not intended to be configured manually.
aor
The configuration for a location of an endpoint
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Option Name | Type | Default Value | Regular Expression | Description |
---|---|---|---|---|
|
| | Permanent contacts assigned to AoR | |
| | | | Default expiration time in seconds for contacts that are dynamically bound to an AoR. |
|
| | Allow subscriptions for the specified mailbox(es) | |
| |
| | The voicemail extension to send in the NOTIFY Message-Account header |
| | | Maximum time to keep an AoR | |
| | | Maximum number of contacts that can bind to an AoR | |
| | | Minimum keep alive time for an AoR | |
| | | Determines whether new contacts replace existing ones. | |
| |
| | Must be of type 'aor'. |
| | | Interval at which to qualify an AoR | |
| | | Timeout for qualify | |
| | | Authenticates a qualify request challenge response if needed | |
|
| | Outbound proxy used when sending OPTIONS request | |
| | | Enables Path support for REGISTER requests and Route support for other requests. |
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maximum_expiration
Maximium Maximum time to keep a peer with explicit expiration. Time in seconds.
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Maximum number of contacts that can associate with this AoR. This value does not affect the number of contacts that can be added with the "contact" option. It only limits contacts added through external interaction, such as registration.
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The rewrite_contact option registers the source address as the contact address to help with NAT and reusing connection oriented transports such as TCP and TLS. Unfortunately, refreshing a registration may register a different contact address and exceed max_contacts. The remove_existing option can help by removing the soonest to expire contact(s) over max_contacts which is likely the old rewrite_contact contact source address being refreshed. |
Info | ||
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This should be set to |
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Minimum time to keep a peer with an explict explicit expiration. Time in seconds.
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On receiving a new registration to the AoR should it remove enough existing contacts not added or updated by the registration to satisfy max_contacts? Any removed contacts will expire the soonest.
Info | ||
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| ||
The rewrite_contact option registers the source address as the contact address to help with NAT and reusing connection oriented transports such as TCP and TLS. Unfortunately, refreshing a registration may register a different contact address and exceed max_contacts. The remove_existing |
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option can help by removing the soonest to expire contact(s) over max_contacts which is likely the old rewrite_contact contact source address being refreshed. |
Info | ||
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| ||
This should be set to |
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qualify_timeout
If the contact doesn't repond respond to the OPTIONS request before the timeout, the contact is marked unavailable. If 0
no timeout. Time in fractional seconds.
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If true and a qualify request receives a challenge or authenticate response then authentication is attempted before declaring the contact available.
Info | ||
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| ||
This option does nothing as we will always complete the challenge response authentication if the qualify request is challenged. |
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|
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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. | |
| | | | The interval (in seconds) to check for expired contacts. | |
| | | Disable Multi Domain support | ||
| | | | 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. | |
| | | The number of seconds over which to accumulate unidentified requests. | ||
| | | The number of unidentified requests from a single IP to allow. | ||
| | | | The interval at which unidentified requests are older than twice the unidentified_request_period are pruned. | |
| |
| | 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. | |
| |
| | The voicemail extension to send in the NOTIFY Message-Account header if not specified on endpoint or aor | |
| | | | 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_*). You can use the CLI command "pjsip show identifiers" to see the identifiers currently available. | ||
| | | | 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. | |
| | | | When Asterisk generates a challenge, the digest realm will be set to this value if there is no better option (such as auth/realm) to be used. | |
| | | MWI taskprocessor high water alert trigger level. | ||
| | | MWI taskprocessor low water clear alert level. | ||
| | | Enable/Disable sending unsolicited MWI to all endpoints on startup. | ||
| | | Enable/Disable ignoring SIP URI user field options. |
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If disabled it can improve realtime performace performance by reducing the number of database requstsrequests.
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|
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If you have this option enabled and there are semicolons in the user field of a SIP URI then the field is truncated at the first semicolon. This effectively makes the semicolon a non-usable character for PJSIP endpoint names, extensions, and AORs. This can be useful for improving compatability compatibility with an ITSP that likes to use user options for whatever reason.
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This documentation was imported from Asterisk Version GIT-13-13.15.0-rc1-3621494-ga62b48c9ffg6dbecc2319