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  1. Endpoint
  2.  
  3. The *Endpoint* is the primary configuration object. It contains the core SIP
  4. related options only, endpoints are *NOT* dialable entries of their own.
  5. Communication with another SIP device is accomplished via Addresses of Record
  6. (AoRs) which have one or more contacts assicated with them. Endpoints *NOT*
  7. configured to use a 'transport' will default to first transport found in
  8. "pjsip.conf" that matches its type.
  9. Example: An Endpoint has been configured with no transport. When it comes time
  10. to call an AoR, PJSIP will find the first transport that matches the type. A
  11. SIP URI of 'sip:5000@[11::33]' will use the first IPv6 transport and try to
  12. send the request.
  13. If the anonymous endpoint identifier is in use an endpoint with the name
  14. "anonymous@domain" will be searched for as a last resort. If this is not found
  15. it will fall back to searching for "anonymous". If neither endpoints are found
  16. the anonymous endpoint identifier will not return an endpoint and anonymous
  17. calling will not be possible.
  18.  
  19. 100rel                    -- Allow support for RFC3262 provisional ACK tags
  20. aggregate_mwi             -- Condense MWI notifications into a single NOTIFY.
  21. allow                     -- Media Codec(s) to allow
  22. allow_overlap             -- Enable RFC3578 overlap dialing support.
  23. aors                      -- AoR(s) to be used with the endpoint
  24. auth                      -- Authentication Object(s) associated with the endpoint
  25. callerid                  -- CallerID information for the endpoint
  26. callerid_privacy          -- Default privacy level
  27. callerid_tag              -- Internal id_tag for the endpoint
  28. context                   -- Dialplan context for inbound sessions
  29. direct_media_glare_mitigation -- Mitigation of direct media (re)INVITE glare
  30. direct_media_method       -- Direct Media method type
  31. connected_line_method     -- Connected line method type
  32. direct_media              -- Determines whether media may flow directly between endpoints.
  33. disable_direct_media_on_nat -- Disable direct media session refreshes when NAT obstructs the med
  34. disallow                  -- Media Codec(s) to disallow
  35. dtmf_mode                 -- DTMF mode
  36. media_address             -- IP address used in SDP for media handling
  37. bind_rtp_to_media_address -- Bind the RTP instance to the media_address
  38. force_rport               -- Force use of return port
  39. ice_support               -- Enable the ICE mechanism to help traverse NAT
  40. identify_by               -- Way(s) for Endpoint to be identified
  41. redirect_method           -- How redirects received from an endpoint are handled
  42. mailboxes                 -- NOTIFY the endpoint when state changes for any of the specified m
  43. mwi_subscribe_replaces_unsolicited -- An MWI subscribe will replace sending unsolicited NOTIFYs
  44. voicemail_extension       -- The voicemail extension to send in the NOTIFY Message-Account hea
  45. moh_suggest               -- Default Music On Hold class
  46. outbound_auth             -- Authentication object(s) used for outbound requests
  47. outbound_proxy            -- Full SIP URI of the outbound proxy used to send requests
  48. rewrite_contact           -- Allow Contact header to be rewritten with the source IP address-p
  49. rtp_ipv6                  -- Allow use of IPv6 for RTP traffic
  50. rtp_symmetric             -- Enforce that RTP must be symmetric
  51. send_diversion            -- Send the Diversion header, conveying the diversion information to
  52. send_pai                  -- Send the P-Asserted-Identity header
  53. send_rpid                 -- Send the Remote-Party-ID header
  54. rpid_immediate            -- Immediately send connected line updates on unanswered incoming ca
  55. timers_min_se             -- Minimum session timers expiration period
  56. timers                    -- Session timers for SIP packets
  57. timers_sess_expires       -- Maximum session timer expiration period
  58. transport                 -- Desired transport configuration
  59. trust_id_inbound          -- Accept identification information received from this endpoint
  60. trust_id_outbound         -- Send private identification details to the endpoint.
  61. type                      -- Must be of type 'endpoint'.
  62. use_ptime                 -- Use Endpoint's requested packetisation interval
  63. use_avpf                  -- Determines whether res_pjsip will use and enforce usage of AVPF f
  64. force_avp                 -- Determines whether res_pjsip will use and enforce usage of AVP, r
  65. media_use_received_transport -- Determines whether res_pjsip will use the media transport receive
  66. media_encryption          -- Determines whether res_pjsip will use and enforce usage of media
  67. media_encryption_optimistic -- Determines whether encryption should be used if possible but does
  68. g726_non_standard         -- Force g.726 to use AAL2 packing order when negotiating g.726 audi
  69. inband_progress           -- Determines whether chan_pjsip will indicate ringing using inband
  70. call_group                -- The numeric pickup groups for a channel.
  71. pickup_group              -- The numeric pickup groups that a channel can pickup.
  72. named_call_group          -- The named pickup groups for a channel.
  73. named_pickup_group        -- The named pickup groups that a channel can pickup.
  74. device_state_busy_at      -- The number of in-use channels which will cause busy to be returne
  75. t38_udptl                 -- Whether T.38 UDPTL support is enabled or not
  76. t38_udptl_ec              -- T.38 UDPTL error correction method
  77. t38_udptl_maxdatagram     -- T.38 UDPTL maximum datagram size
  78. fax_detect                -- Whether CNG tone detection is enabled
  79. fax_detect_timeout        -- How long into a call before fax_detect is disabled for the call
  80. t38_udptl_nat             -- Whether NAT support is enabled on UDPTL sessions
  81. t38_udptl_ipv6            -- Whether IPv6 is used for UDPTL Sessions
  82. tone_zone                 -- Set which country's indications to use for channels created for t
  83. language                  -- Set the default language to use for channels created for this end
  84. one_touch_recording       -- Determines whether one-touch recording is allowed for this endpoi
  85. record_on_feature         -- The feature to enact when one-touch recording is turned on.
  86. record_off_feature        -- The feature to enact when one-touch recording is turned off.
  87. rtp_engine                -- Name of the RTP engine to use for channels created for this endpo
  88. allow_transfer            -- Determines whether SIP REFER transfers are allowed for this endpo
  89. user_eq_phone             -- Determines whether a user=phone parameter is placed into the requ
  90. moh_passthrough           -- Determines whether hold and unhold will be passed through using r
  91. sdp_owner                 -- String placed as the username portion of an SDP origin (o=) line.
  92. sdp_session               -- String used for the SDP session (s=) line.
  93. tos_audio                 -- DSCP TOS bits for audio streams
  94. tos_video                 -- DSCP TOS bits for video streams
  95. cos_audio                 -- Priority for audio streams
  96. cos_video                 -- Priority for video streams
  97. allow_subscribe           -- Determines if endpoint is allowed to initiate subscriptions with
  98. sub_min_expiry            -- The minimum allowed expiry time for subscriptions initiated by th
  99. from_user                 -- Username to use in From header for requests to this endpoint.
  100. mwi_from_user             -- Username to use in From header for unsolicited MWI NOTIFYs to thi
  101. from_domain               -- Domain to user in From header for requests to this endpoint.
  102. dtls_verify               -- Verify that the provided peer certificate is valid
  103. dtls_rekey                -- Interval at which to renegotiate the TLS session and rekey the SR
  104. dtls_cert_file            -- Path to certificate file to present to peer
  105. dtls_private_key          -- Path to private key for certificate file
  106. dtls_cipher               -- Cipher to use for DTLS negotiation
  107. dtls_ca_file              -- Path to certificate authority certificate
  108. dtls_ca_path              -- Path to a directory containing certificate authority certificates
  109. dtls_setup                -- Whether we are willing to accept connections, connect to the othe
  110. dtls_fingerprint          -- Type of hash to use for the DTLS fingerprint in the SDP.
  111. srtp_tag_32               -- Determines whether 32 byte tags should be used instead of 80 byte
  112. set_var                   -- Variable set on a channel involving the endpoint.
  113. message_context           -- Context to route incoming MESSAGE requests to.
  114. accountcode               -- An accountcode to set automatically on any channels created for t
  115. rtp_keepalive             -- Number of seconds between RTP comfort noise keepalive packets.
  116. rtp_timeout               -- Maximum number of seconds without receiving RTP (while off hold)
  117. rtp_timeout_hold          -- Maximum number of seconds without receiving RTP (while on hold) b
  118. acl                       -- List of IP ACL section names in acl.conf
  119. deny                      -- List of IP addresses to deny access from
  120. permit                    -- List of IP addresses to permit access from
  121. contact_acl               -- List of Contact ACL section names in acl.conf
  122. contact_deny              -- List of Contact header addresses to deny
  123. contact_permit            -- List of Contact header addresses to permit
  124. subscribe_context         -- Context for incoming MESSAGE requests.
  125. contact_user              -- Force the user on the outgoing Contact header to this value.
  126. asymmetric_rtp_codec      -- Allow the sending and receiving RTP codec to differ
  127. rtcp_mux                  -- Enable RFC 5761 RTCP multiplexing on the RTP port
  128.  
  129. Authentication type
  130.  
  131.  Authentication objects hold the authentication information for use by other
  132. objects such as 'endpoints' or 'registrations'. This also allows for multiple
  133. objects to use a single auth object. See the 'auth_type' config option for
  134. password style choices.
  135.  
  136. auth_type                 -- Authentication type
  137. nonce_lifetime            -- Lifetime of a nonce associated with this authentication config.
  138. md5_cred                  -- MD5 Hash used for authentication.
  139. password                  -- PlainText password used for authentication.
  140. realm                     -- SIP realm for endpoint
  141. type                      -- Must be 'auth'
  142. username                  -- Username to use for account
  143.  
  144. Domain Alias
  145.  
  146.  Signifies that a domain is an alias. If the domain on a session is not found
  147. to match an AoR then this object is used to see if we have an alias for the AoR
  148. to which the endpoint is binding. This objects name as defined in configuration
  149. should be the domain alias and a config option is provided to specify the
  150. domain to be aliased.
  151.  
  152. type                      -- Must be of type 'domain_alias'.
  153. domain                    -- Domain to be aliased
  154.  
  155. SIP Transport
  156.  
  157.  *Transports*
  158. There are different transports and protocol derivatives supported by
  159. 'res_pjsip'. They are in order of preference: UDP, TCP, and WebSocket (WS).
  160. NOTE: Changes to transport configuration in pjsip.conf will only be effected on
  161. a complete restart of Asterisk. A module reload will not suffice.
  162.  
  163. async_operations          -- Number of simultaneous Asynchronous Operations
  164. bind                      -- IP Address and optional port to bind to for this transport
  165. ca_list_file              -- File containing a list of certificates to read (TLS ONLY)
  166. ca_list_path              -- Path to directory containing a list of certificates to read (TLS
  167. cert_file                 -- Certificate file for endpoint (TLS ONLY)
  168. cipher                    -- Preferred cryptography cipher names (TLS ONLY)
  169. domain                    -- Domain the transport comes from
  170. external_media_address    -- External IP address to use in RTP handling
  171. external_signaling_address -- External address for SIP signalling
  172. external_signaling_port   -- External port for SIP signalling
  173. method                    -- Method of SSL transport (TLS ONLY)
  174. local_net                 -- Network to consider local (used for NAT purposes).
  175. password                  -- Password required for transport
  176. priv_key_file             -- Private key file (TLS ONLY)
  177. protocol                  -- Protocol to use for SIP traffic
  178. require_client_cert       -- Require client certificate (TLS ONLY)
  179. type                      -- Must be of type 'transport'.
  180. verify_client             -- Require verification of client certificate (TLS ONLY)
  181. verify_server             -- Require verification of server certificate (TLS ONLY)
  182. tos                       -- Enable TOS for the signalling sent over this transport
  183. cos                       -- Enable COS for the signalling sent over this transport
  184. websocket_write_timeout   -- The timeout (in milliseconds) to set on WebSocket connections.
  185. allow_reload              -- Allow this transport to be reloaded.
  186. symmetric_transport       -- Use the same transport for outgoing reqests as incoming ones.
  187.  
  188. A way of creating an aliased name to a SIP URI
  189.  
  190.  Contacts are a way to hide SIP URIs from the dialplan directly. They are also
  191. used to make a group of contactable parties when in use with 'AoR' lists.
  192.  
  193. type                      -- Must be of type 'contact'.
  194. uri                       -- SIP URI to contact peer
  195. expiration_time           -- Time to keep alive a contact
  196. qualify_frequency         -- Interval at which to qualify a contact
  197. qualify_timeout           -- Timeout for qualify
  198. authenticate_qualify      -- Authenticates a qualify request if needed
  199. outbound_proxy            -- Outbound proxy used when sending OPTIONS request
  200. path                      -- Stored Path vector for use in Route headers on outgoing requests.
  201. user_agent                -- User-Agent header from registration.
  202. endpoint                  -- Endpoint name
  203. reg_server                -- Asterisk Server name
  204. via_addr                  -- IP-address of the last Via header from registration.
  205. via_port                  -- IP-port of the last Via header from registration.
  206. call_id                   -- Call-ID header from registration.
  207.  
  208. The configuration for a location of an endpoint
  209.  
  210.  An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
  211. AoRs are specified, an endpoint will not be reachable by Asterisk. Beyond that,
  212. an AoR has other uses within Asterisk, such as inbound registration.
  213.  An 'AoR' is a way to allow dialing a group of 'Contacts' that all use the same
  214.  'endpoint' for calls.
  215.  This can be used as another way of grouping a list of contacts to dial rather
  216.  than specifing them each directly when dialing via the dialplan. This must be
  217.  used in conjuction with the 'PJSIP_DIAL_CONTACTS'.
  218.  Registrations: For Asterisk to match an inbound registration to an endpoint,
  219.  the AoR object name must match the user portion of the SIP URI in the "To:"
  220.  header of the inbound SIP registration. That will usually be equivalent to the
  221.  "user name" set in your hard or soft phones configuration.
  222.  
  223. contact                   -- Permanent contacts assigned to AoR
  224. default_expiration        -- Default expiration time in seconds for contacts that are dynamica
  225. mailboxes                 -- Allow subscriptions for the specified mailbox(es)
  226. voicemail_extension       -- The voicemail extension to send in the NOTIFY Message-Account hea
  227. maximum_expiration        -- Maximum time to keep an AoR
  228. max_contacts              -- Maximum number of contacts that can bind to an AoR
  229. minimum_expiration        -- Minimum keep alive time for an AoR
  230. remove_existing           -- Determines whether new contacts replace existing ones.
  231. type                      -- Must be of type 'aor'.
  232. qualify_frequency         -- Interval at which to qualify an AoR
  233. qualify_timeout           -- Timeout for qualify
  234. authenticate_qualify      -- Authenticates a qualify request if needed
  235. outbound_proxy            -- Outbound proxy used when sending OPTIONS request
  236. support_path              -- Enables Path support for REGISTER requests and Route support for
  237.  
  238. Options that apply to the SIP stack as well as other system-wide settings
  239.  
  240.  The settings in this section are global. In addition to being global, the
  241. values will not be re-evaluated when a reload is performed. This is because the
  242. values must be set before the SIP stack is initialized. The only way to reset
  243. these values is to either restart Asterisk, or unload res_pjsip.so and then
  244. load it again.
  245.  
  246. timer_t1                  -- Set transaction timer T1 value (milliseconds).
  247. timer_b                   -- Set transaction timer B value (milliseconds).
  248. compact_headers           -- Use the short forms of common SIP header names.
  249. threadpool_initial_size   -- Initial number of threads in the res_pjsip threadpool.
  250. threadpool_auto_increment -- The amount by which the number of threads is incremented when nec
  251. threadpool_idle_timeout   -- Number of seconds before an idle thread should be disposed of.
  252. threadpool_max_size       -- Maximum number of threads in the res_pjsip threadpool. A value of
  253. disable_tcp_switch        -- Disable automatic switching from UDP to TCP transports.
  254. type                      -- Must be of type 'system'.
  255.  
  256. Options that apply globally to all SIP communications
  257.  
  258.  The settings in this section are global. Unlike options in the 'system'
  259. section, these options can be refreshed by performing a reload.
  260.  
  261. max_forwards              -- Value used in Max-Forwards header for SIP requests.
  262. keep_alive_interval       -- The interval (in seconds) to send keepalives to active connection
  263. contact_expiration_check_interval -- The interval (in seconds) to check for expired contacts.
  264. disable_multi_domain      -- Disable Multi Domain support
  265. max_initial_qualify_time  -- The maximum amount of time from startup that qualifies should be
  266. unidentified_request_period -- The number of seconds over which to accumulate unidentified reque
  267. unidentified_request_count -- The number of unidentified requests from a single IP to allow.
  268. unidentified_request_prune_interval -- The interval at which unidentified requests are older than twice
  269. type                      -- Must be of type 'global'.
  270. user_agent                -- Value used in User-Agent header for SIP requests and Server heade
  271. regcontext                -- When set, Asterisk will dynamically create and destroy a NoOp pri
  272. default_outbound_endpoint -- Endpoint to use when sending an outbound request to a URI without
  273. default_voicemail_extension -- The voicemail extension to send in the NOTIFY Message-Account hea
  274. debug                     -- Enable/Disable SIP debug logging.  Valid options include yes|no o
  275. endpoint_identifier_order -- The order by which endpoint identifiers are processed and checked
  276. default_from_user         -- When Asterisk generates an outgoing SIP request, the From header
  277. default_realm             -- When Asterisk generates a challenge, the digest realm will be set
  278. mwi_tps_queue_high        -- MWI taskprocessor high water alert trigger level.
  279. mwi_tps_queue_low         -- MWI taskprocessor low water clear alert level.
  280. mwi_disable_initial_unsolicited -- Enable/Disable sending unsolicited MWI to all endpoints on startu
  281. ignore_uri_user_options   -- Enable/Disable ignoring SIP URI user field options.

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