From: George Joseph Date: Thu, 16 Jan 2025 21:54:35 +0000 (-0700) Subject: docs: Add version information to configObject and configOption XML elements X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=a22dc33057fdde04f313feb1d27a3a143de1b738;p=thirdparty%2Fasterisk.git docs: Add version information to configObject and configOption XML elements Most of the configObjects and configOptions that are implemented with ACO or Sorcery now have `/` elements added. There are probably some that the script I used didn't catch. The version tags were determined by the following... * Do a git blame on the API call that created the object or option. * From the commit hash, grab the summary line. * Do a `git log --grep ` to find the cherry-pick commits in all branches that match. * Do a `git patch-id` to ensure the commits are all related and didn't get a false match on the summary. * Do a `git tag --contains ` to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the API was last touched. configObjects and configOptions elements implemented with the base ast_config APIs were just not possible to find due to the non-deterministic way they are accessed. Also note that if the API call was on modified after it was added, the version will be the one it was last modified in. Final note: The configObject and configOption elements were introduced in 12.0.0 so options created before then may not have any XML documentation. --- diff --git a/apps/app_agent_pool.c b/apps/app_agent_pool.c index 3bff4c719b..b3f78a23bc 100644 --- a/apps/app_agent_pool.c +++ b/apps/app_agent_pool.c @@ -268,11 +268,13 @@ Unused, but reserved. + 12.0.0 Configure an agent for the pool. + 12.0.0 Enable to require the agent to acknowledge a call. Enable to require the agent to give a DTMF acknowledgement @@ -282,6 +284,7 @@ + 12.0.0 DTMF key sequence the agent uses to acknowledge a call. The option is overridden by AGENTACCEPTDTMF on agent login. @@ -290,6 +293,7 @@ + 12.0.0 Time the agent has to acknowledge a call before being logged off. Set how many seconds a call for the agent has to wait for the @@ -302,6 +306,7 @@ + 12.0.0 Minimum time the agent has between calls. Set the minimum amount of time in milliseconds after @@ -311,12 +316,14 @@ + 12.0.0 Music on hold class the agent listens to between calls. + 12.0.0 Enable to automatically record calls the agent takes. Enable recording calls the agent takes automatically by @@ -327,12 +334,14 @@ + 12.4.0 Sound file played to alert the agent when a call is present. + 12.0.0 A friendly name for the agent used in log messages. diff --git a/apps/app_skel.c b/apps/app_skel.c index 01bc8f98cc..5d5b0cddd6 100644 --- a/apps/app_skel.c +++ b/apps/app_skel.c @@ -88,42 +88,55 @@ + 12.0.0 Options that apply globally to app_skel + 11.0.0 The number of games a single execution of SkelGuessNumber will play + 11.0.0 Should the computer cheat? If enabled, the computer will ignore winning guesses. + 12.0.0 Prompts for SkelGuessNumber to play + 11.0.0 A prompt directing the user to enter a number less than the max number + 11.0.0 The sound file to play when a wrong guess is made + 11.0.0 The sound file to play when a correct guess is made + 11.0.0 The sound file to play when a guess is too low + 11.0.0 The sound file to play when a guess is too high + 11.0.0 The sound file to play when a player loses + 12.0.0 Defined levels for the SkelGuessNumber game + 11.0.0 The maximum in the range of numbers to guess (1 is the implied minimum) + 11.22.013.9.0 The maximum number of guesses before a game is considered lost diff --git a/apps/confbridge/conf_config_parser.c b/apps/confbridge/conf_config_parser.c index 6c3582edf3..e4cb9179de 100644 --- a/apps/confbridge/conf_config_parser.c +++ b/apps/confbridge/conf_config_parser.c @@ -48,6 +48,7 @@ Unused, but reserved. + 12.0.0 A named profile to apply to specific callers. Callers in a ConfBridge have a profile associated with them that determine their options. A configuration section is determined to be a @@ -55,6 +56,7 @@ of user. + 11.0.0 Define this configuration category as a user profile. The type parameter determines how a context in the configuration file is interpreted. @@ -66,9 +68,11 @@ + 11.0.0 Sets if the user is an admin or not + 15.5.0 Sets if events are send to the user If events are enabled for this bridge and this option is set, users will receive events like join, leave, talking, etc. via text @@ -78,6 +82,7 @@ interface. + 15.5.0 Sets if events are echoed back to the user that triggered them If events are enabled for this user and this option @@ -86,24 +91,31 @@ + 11.0.0 Sets if this is a marked user or not + 11.0.0 Sets if all users should start out muted + 11.0.0 Play MOH when user is alone or waiting on a marked user + 11.0.0 Silence enter/leave prompts and user intros for this user + 16.26.018.12.019.4.0 Determines if the user also hears the join sound when they enter a conference + 11.0.0 Sets if the number of users should be announced to the user + 11.0.0 Announce user count to all the other users when this user joins Sets if the number of users should be announced to all the other users in the conference when this user joins. This option can be either set to 'yes' or @@ -112,40 +124,52 @@ + 11.0.0 Announce to a user when they join an empty conference + 11.0.0 Sets if the user must wait for a marked user to enter before joining a conference + 11.0.0 Kick the user from the conference when the last marked user leaves + 16.29.018.15.019.7.0 Kick the user from the conference when any marked user leaves + 11.0.0 Set whether or not notifications of when a user begins and ends talking should be sent out as events over AMI + 11.0.0 Sets whether or not DTMF should pass through the conference + 11.0.0 Prompt user for their name when joining a conference and play it to the conference when they enter + 13.0.0 Prompt user for their name when joining a conference and play it to the conference when they enter. The user will be asked to review the recording of their name before entering the conference. + 11.0.0 Sets a PIN the user must enter before joining the conference + 11.0.0 The MOH class to use for this user + 11.0.0 Sound file to play to the user when they join a conference + 11.0.0 Apply a denoise filter to the audio before mixing Sets whether or not a denoise filter should be applied to the audio before mixing or not. Off by default. Requires @@ -157,6 +181,7 @@ + 11.0.0 Drop what Asterisk detects as silence from audio sent to the bridge This option drops what Asterisk detects as silence from @@ -167,6 +192,7 @@ + 11.0.0 The number of milliseconds of silence necessary to declare talking stopped. The time in milliseconds of sound falling below the @@ -200,6 +226,7 @@ + 11.10.012.3.0 Average magnitude threshold to determine talking. The minimum average magnitude per sample in a frame @@ -230,6 +257,7 @@ + 11.0.0 Place a jitter buffer on the user's audio stream before audio mixing is performed Enabling this option places a jitterbuffer on the user's audio stream @@ -241,12 +269,15 @@ + 11.0.0 When using the CONFBRIDGE dialplan function, use a user profile as a template for creating a new temporary profile + 13.7.0 Kick the user out of the conference after this many seconds. 0 means there is no timeout for the user. + 16.10.017.4.0 Sets if text messages are sent to the user. If text messaging is enabled for this user then text messages will be sent to it. These may be events or from other @@ -254,10 +285,12 @@ messages are sent to the user. + 16.19.018.5.0 Sets if a user's channel should be answered if currently unanswered. + 12.0.0 A named profile to apply to specific bridges. ConfBridge bridges have a profile associated with them that determine their options. A configuration section is determined to be a @@ -265,6 +298,7 @@ of bridge. + 12.0.0 Define this configuration category as a bridge profile The type parameter determines how a context in the configuration file is interpreted. @@ -276,9 +310,11 @@ + 11.0.0 Place a jitter buffer on the conference's audio stream + 18.22.020.7.021.2.0 Set the internal native sample rate for mixing the conference Sets the internal native sample rate the @@ -291,6 +327,7 @@ + 18.22.020.7.021.2.0 Set the maximum native sample rate for mixing the conference Sets the maximum native sample rate the @@ -300,6 +337,7 @@ + 11.7.0 The language used for announcements to the conference. By default, announcements to a conference use English. Which means @@ -309,6 +347,7 @@ + 11.0.0 Sets the internal mixing interval in milliseconds for the bridge Sets the internal mixing interval in milliseconds for the bridge. This @@ -320,6 +359,7 @@ + 15.0.0 If true binaural conferencing with stereo audio is active Activates binaural mixing for a conference bridge. @@ -327,6 +367,7 @@ + 11.0.0 Record the conference starting with the first active user's entrance and ending with the last active user's exit Records the conference call starting when the first user @@ -338,6 +379,7 @@ + 11.0.0 The filename of the conference recording When record_conference is set to yes, the specific name of the @@ -351,6 +393,7 @@ + 12.0.0 Append to record file when starting/stopping on same conference recording When record_file_append is set to yes, stopping and starting recording on a @@ -360,6 +403,7 @@ + 14.0.0 Append the start time to the record_file name so that it is unique. When record_file_timestamp is set to yes, the start time is appended to @@ -368,6 +412,7 @@ + 14.0.0 Pass additional options to MixMonitor when recording Pass additional options to MixMonitor when record_conference is set to yes. @@ -375,6 +420,7 @@ + 14.0.0 Execute a command after recording ends Executes the specified command when recording ends. Any strings matching ^{X} will be @@ -382,6 +428,7 @@ + 13.10.0 The name of the context into which to register the name of the conference bridge as NoOP() at priority 1 When set this will cause the name of the created conference to be registered @@ -398,6 +445,7 @@ + 11.0.0 Sets how confbridge handles video distribution to the conference participants Sets how confbridge handles video distribution to the conference participants. @@ -435,6 +483,7 @@ + 11.0.0 Limit the maximum number of participants for a single conference This option limits the number of participants for a single @@ -446,6 +495,7 @@ + 13.19.015.2.0 Override the various conference bridge sound files All sounds in the conference are customizable using the bridge profile options below. @@ -491,6 +541,7 @@ + 15.0.0 Sets the amount of time in milliseconds after sending a video update to discard subsequent video updates Sets the amount of time in milliseconds after sending a video update request @@ -501,6 +552,7 @@ + 15.4.0 Sets the interval in milliseconds that a combined REMB frame will be sent to video sources Sets the interval in milliseconds that a combined REMB frame will be sent @@ -513,6 +565,7 @@ + 15.4.0 Sets how REMB reports are generated from multiple sources Sets how REMB reports are combined from multiple sources to form one. A REMB report @@ -551,6 +604,7 @@ remb_estimated_bitrate + 16.15.017.9.018.1.0 Sets the estimated bitrate sent to each participant in REMB reports When remb_behavior is set to force, @@ -560,6 +614,7 @@ remb_behavior + 15.5.0 Enables events for this bridge If enabled, recipients who joined the bridge via a channel driver @@ -571,10 +626,12 @@ + 11.0.0 When using the CONFBRIDGE dialplan function, use a bridge profile as a template for creating a new temporary profile + 12.0.0 A conference user menu Conference users, as defined by a conf_user, @@ -582,6 +639,7 @@ ConfBridge application. + 12.0.0 Define this configuration category as a menu The type parameter determines how a context in the configuration file is interpreted. @@ -593,6 +651,7 @@ + 13.0.0 When using the CONFBRIDGE dialplan function, use a menu profile as a template for creating a new temporary profile diff --git a/channels/chan_motif.c b/channels/chan_motif.c index 5ec348d590..fe26905648 100644 --- a/channels/chan_motif.c +++ b/channels/chan_motif.c @@ -149,38 +149,50 @@ + 12.0.0 The configuration for an endpoint. + 11.0.0 Default dialplan context that incoming sessions will be routed to + 11.0.0 A callgroup to assign to this endpoint. + 11.0.0 A pickup group to assign to this endpoint. + 11.0.0 The default language for this endpoint. + 11.0.0 Default music on hold class for this endpoint. + 11.0.0 Default parking lot for this endpoint. + 11.0.0 Accout code for CDR purposes + 13.0.0 Codecs to allow + 13.0.0 Codecs to disallow + 11.0.0 Connection to accept traffic on and on which to send traffic out + 11.0.0 The transport to use for the endpoint. The default outbound transport for this endpoint. Inbound @@ -206,9 +218,11 @@ + 11.0.0 Maximum number of ICE candidates to offer + 11.0.0 Maximum number of payloads to offer diff --git a/main/cdr.c b/main/cdr.c index fb43c87d94..e7dc50acca 100644 --- a/main/cdr.c +++ b/main/cdr.c @@ -82,8 +82,10 @@ + 12.0.0 Global settings applied to the CDR engine. + 12.0.0 Enable/disable verbose CDR debugging. When set to True, verbose updates of changes in CDR information will be logged. Note that this is only @@ -91,12 +93,14 @@ + 12.0.0 Enable/disable CDR logging. Define whether or not to use CDR logging. Setting this to "no" will override any loading of backend CDR modules. + 16.24.018.10.019.2.0 Whether CDR is enabled on a channel by default Define whether or not CDR should be enabled on a channel by default. Setting this to "yes" will enable CDR on every channel unless it is explicitly disabled. @@ -112,6 +116,7 @@ + 16.30.018.16.019.8.020.1.0 Whether CDR is updated or forked by bridging changes. Define whether or not CDR should be updated by bridging changes. This includes entering and leaving bridges and call parking. @@ -122,6 +127,7 @@ + 16.30.018.16.019.8.020.1.0 Whether CDR is updated or forked by dial updates. Define whether or not CDR should be updated by dial updates. If this is set to "no", a single CDR will be used for the channel, even if @@ -135,6 +141,7 @@ + 12.0.0 Log calls that are never answered and don't set an outgoing party. Define whether or not to log unanswered calls that don't involve an outgoing party. Setting @@ -147,12 +154,14 @@ + 12.0.0 Log congested calls. Define whether or not to log congested calls. Setting this to "yes" will report each call that fails to complete due to congestion conditions. + 12.0.0 Don't produce CDRs while executing hangup logic As each CDR for a channel is finished, its end time is updated @@ -167,6 +176,7 @@ + 12.0.0 Count microseconds for billsec purposes Normally, the billsec field logged to the CDR backends is simply the end time (hangup time) minus the answer time in seconds. Internally, @@ -178,6 +188,7 @@ + 12.0.0 Submit CDRs to the backends for processing in batches Define the CDR batch mode, where instead of posting the CDR at the end of every call, the data will be stored in a buffer to help alleviate load on the @@ -188,12 +199,14 @@ + 12.0.0 The maximum number of CDRs to accumulate before triggering a batch Define the maximum number of CDRs to accumulate in the buffer before posting them to the backend engines. batch must be set to yes. + 13.22.015.5.0 The maximum time to accumulate CDRs before triggering a batch Define the maximum time to accumulate CDRs before posting them in a batch to the backend engines. If this time limit is reached, then it will post the records, regardless of the value @@ -202,6 +215,7 @@ + 12.0.0 Post batched CDRs on their own thread instead of the scheduler The CDR engine uses the internal asterisk scheduler to determine when to post records. Posting can either occur inside the scheduler thread, or a new @@ -212,6 +226,7 @@ + 12.0.0 Block shutdown of Asterisk until CDRs are submitted When shutting down asterisk, you can block until the CDRs are submitted. If you don't, then data will likely be lost. You can always check the size of diff --git a/main/cel.c b/main/cel.c index 98d31b551a..25c23d9083 100644 --- a/main/cel.c +++ b/main/cel.c @@ -63,20 +63,25 @@ + 12.0.0 Options that apply globally to Channel Event Logging (CEL) + 12.0.0 Determines whether CEL is enabled + 12.0.0 The format to be used for dates when logging + 12.0.0 List of apps for CEL to track A case-insensitive, comma-separated list of applications to track when one or both of APP_START and APP_END events are flagged for tracking + 12.0.0 List of events for CEL to track A case-sensitive, comma-separated list of event names to track. These event names do not include the leading AST_CEL. diff --git a/main/features_config.c b/main/features_config.c index 5b6e209463..22e4a1a801 100644 --- a/main/features_config.c +++ b/main/features_config.c @@ -33,24 +33,31 @@ Features Configuration + 12.0.0 + 12.0.0 Milliseconds allowed between digit presses when entering a feature code. + 12.0.0 Sound to play when automixmon is activated + 12.0.0 Sound to play when automixmon is attempted but fails to start + 12.0.0 Seconds allowed between digit presses when dialing a transfer destination + 12.0.0 Seconds to wait for attended transfer destination to answer + 12.0.0 Hang up the call entirely if the attended transfer fails When this option is set to no, then Asterisk will attempt to @@ -65,14 +72,17 @@ + 12.0.0 Seconds to wait between attempts to re-dial transfer destination atxferdropcall + 12.0.0 Number of times to re-attempt dialing a transfer destination atxferdropcall + 12.0.0 Sound to play to during transfer and transfer-like operations. This sound will play to the transferrer and transfer target channels when @@ -81,6 +91,7 @@ + 12.0.0 Sound to play to a transferee when a transfer fails @@ -129,27 +140,35 @@ + 12.0.0 Sound to play to picker when a call is picked up + 12.0.0 Sound to play to picker when a call cannot be picked up + 13.1.0 Number of dial attempts allowed when attempting a transfer + 13.1.0 Sound that is played when an incorrect extension is dialed and the transferer should try again. + 13.1.0 Sound that is played when an incorrect extension is dialed and the transferer has no attempts remaining. + 16.29.018.15.019.7.0 Sound that is played to the transferer when a transfer is initiated. If empty, no sound will be played. + 12.0.0 DTMF options that can be triggered during bridged calls + 12.0.0 DTMF sequence to initiate an attended transfer The transferee parties will be placed on hold and the @@ -161,6 +180,7 @@ + 12.0.0 DTMF sequence to initiate a blind transfer The transferee parties will be placed on hold and the @@ -177,6 +197,7 @@ + 12.0.0 DTMF sequence to park a call The parking lot used to park the call is determined by using either the @@ -187,6 +208,7 @@ + 12.0.0 DTMF sequence to start or stop MixMonitor on a call This will cause the channel that pressed the DTMF sequence @@ -206,6 +228,7 @@ + 12.0.0 Section for defining custom feature invocations during a call The applicationmap is an area where new custom features can be created. Items @@ -258,6 +281,7 @@ + 12.0.0 Groupings of items from the applicationmap Feature groups allow for multiple applicationmap items to be diff --git a/main/named_acl.c b/main/named_acl.c index 134cb34c7a..bcd3936f62 100644 --- a/main/named_acl.c +++ b/main/named_acl.c @@ -53,11 +53,14 @@ + 12.0.0 Options for configuring a named ACL + 11.0.0 An address/subnet from which to allow access + 11.0.0 An address/subnet from which to disallow access diff --git a/main/stasis.c b/main/stasis.c index 05a7a505f7..2e68cdf17b 100644 --- a/main/stasis.c +++ b/main/stasis.c @@ -65,20 +65,26 @@ + 12.8.013.1.0 Settings that configure the threadpool Stasis uses to deliver some messages. + 12.8.013.1.0 Initial number of threads in the message bus threadpool. + 12.8.013.1.0 Number of seconds before an idle thread is disposed of. + 12.8.013.1.0 Maximum number of threads in the threadpool. + 13.0.0 Stasis message types for which to decline creation. + 12.8.013.1.0 The message type to decline. This configuration option defines the name of the Stasis diff --git a/main/udptl.c b/main/udptl.c index b1b30a59e9..ec4ec9bad6 100644 --- a/main/udptl.c +++ b/main/udptl.c @@ -81,29 +81,38 @@ + 12.0.0 Global options for configuring UDPTL + 11.0.0 The start of the UDPTL port range + 11.0.0 The end of the UDPTL port range + 11.0.0 Whether to enable or disable UDP checksums on UDPTL traffic + 11.0.0 The number of error correction entries in a UDPTL packet + 11.0.0 The span over which parity is calculated for FEC in a UDPTL packet + 11.0.0 Whether to only use even-numbered UDPTL ports + 11.0.0 Removed + 11.0.0 Removed diff --git a/res/res_aeap.c b/res/res_aeap.c index e78956e20b..188ced5055 100644 --- a/res/res_aeap.c +++ b/res/res_aeap.c @@ -38,17 +38,22 @@ Asterisk External Application Protocol (AEAP) module for Asterisk + 18.12.019.4.0 AEAP client options + 18.12.019.4.0 Must be of type 'client'. + 18.12.019.4.0 The URL of the server to connect to. + 18.12.019.4.0 The application protocol. + 18.12.019.4.0 Optional media codec(s) If this is specified, Asterisk will use this for codec related negotiations diff --git a/res/res_ari.c b/res/res_ari.c index 929e6c4ada..a31708c169 100644 --- a/res/res_ari.c +++ b/res/res_ari.c @@ -82,8 +82,10 @@ HTTP binding for the Stasis API + 12.0.0 General configuration settings + 12.0.0 Enable/disable the ARI module This option enables or disables the ARI module. @@ -97,6 +99,7 @@ + 11.11.012.4.0 The timeout (in milliseconds) to set on WebSocket connections. If a websocket connection accepts input slowly, the timeout @@ -105,22 +108,28 @@ + 12.0.0 Responses from ARI are formatted to be human readable + 12.0.0 Realm to use for authentication. Defaults to Asterisk REST Interface. + 12.0.0 Comma separated list of allowed origins, for Cross-Origin Resource Sharing. May be set to * to allow all origins. + 14.2.0 Comma separated list of channel variables to display in channel json. + 12.0.0 Per-user configuration settings + 13.30.016.7.017.1.0 Define this configuration section as a user. @@ -129,12 +138,15 @@ + 13.30.016.7.017.1.0 When set to yes, user is only authorized for read-only requests + 13.30.016.7.017.1.0 Crypted or plaintext password (see password_format) + 12.0.0 password_format may be set to plain (the default) or crypt. When set to crypt, crypt(3) is used to validate the password. A crypted password can be generated using mkpasswd -m sha-512. When set to plain, the password is in plaintext diff --git a/res/res_geolocation/geoloc_doc.xml b/res/res_geolocation/geoloc_doc.xml index f6fd3d8c67..988d8ca02e 100644 --- a/res/res_geolocation/geoloc_doc.xml +++ b/res/res_geolocation/geoloc_doc.xml @@ -5,16 +5,19 @@ Core Geolocation Support + 16.29.018.15.019.7.0 Location Parameters for defining a Location object + 16.28.018.14.019.6.0 Must be of type 'location'. + 16.28.018.14.019.6.0 Location specification type @@ -46,6 +49,7 @@ + 16.28.018.14.019.6.0 Location information The contents of this parameter are specific to the @@ -73,6 +77,7 @@ + 16.28.018.14.019.6.0 Fully qualified host name This parameter isn't required but if provided, RFC8787 says it MUST be a fully @@ -83,6 +88,7 @@ + 16.28.018.14.019.6.0 Location determination method This is a rarely used field in the specification that would @@ -101,6 +107,7 @@ + 16.28.018.14.019.6.0 Level of confidence This is a rarely used field in the specification that would @@ -131,15 +138,18 @@ + 16.29.018.15.019.7.0 Profile Parameters for defining a Profile object + 16.28.018.14.019.6.0 Must be of type 'profile'. + 16.28.018.14.019.6.0 PIDF-LO element to place this profile in @@ -158,17 +168,21 @@ + 16.28.018.14.019.6.0 Reference to a location object + 16.28.018.14.019.6.0 Reference to a location object + 16.28.018.14.019.6.0 Reference to a location object + 16.28.018.14.019.6.0 location specification type xxxx @@ -176,6 +190,7 @@ + 16.28.018.14.019.6.0 Notes to be added to the outgoing PIDF-LO document The specification of this parameter will cause a @@ -186,15 +201,18 @@ + 16.28.018.14.019.6.0 Sets the value of the Geolocation-Routing header. + 16.29.018.15.019.7.0 Sets if empty Civic Address elements should be suppressed from the PIDF-LO document. + 16.28.018.14.019.6.0 Determine which profile on a channel should be used diff --git a/res/res_hep.c b/res/res_hep.c index 36f7e43ec8..bd38b22b85 100644 --- a/res/res_hep.c +++ b/res/res_hep.c @@ -45,6 +45,7 @@ Resource for integration with Homer using HEPv3 + 12.2.0 General settings. The general settings section contains information @@ -52,6 +53,7 @@ + 12.2.0 Enable or disable packet capturing. @@ -61,6 +63,7 @@ + 12.2.0 The preferred type of UUID to pass to Homer. @@ -70,15 +73,19 @@ + 13.16.014.5.0 The address and port of the Homer server to send packets to. + 12.2.0 If set, the authentication password to send to Homer. + 12.2.0 The ID for this capture agent. + 18.16.020.1.0 The name for this capture agent. diff --git a/res/res_http_media_cache.c b/res/res_http_media_cache.c index 92e49ccbb1..1d0ea41162 100644 --- a/res/res_http_media_cache.c +++ b/res/res_http_media_cache.c @@ -36,29 +36,38 @@ HTTP media cache + 18.18.020.3.0 General configuration + 18.18.020.3.0 The maximum time the transfer is allowed to complete in seconds. See https://curl.se/libcurl/c/CURLOPT_TIMEOUT.html for details. + 18.18.020.3.0 The HTTP User-Agent to use for requests. See https://curl.se/libcurl/c/CURLOPT_USERAGENT.html for details. + 18.18.020.3.0 Follow HTTP 3xx redirects on requests. See https://curl.se/libcurl/c/CURLOPT_FOLLOWLOCATION.html for details. + 18.18.020.3.0 The maximum number of redirects to follow. See https://curl.se/libcurl/c/CURLOPT_MAXREDIRS.html for details. + 18.18.020.3.0 The proxy to use for requests. See https://curl.se/libcurl/c/CURLOPT_PROXY.html for details. + 18.18.020.3.0 The comma separated list of allowed protocols for the request. Available with cURL 7.85.0 or later. See https://curl.se/libcurl/c/CURLOPT_PROTOCOLS_STR.html for details. + 18.18.020.3.0 The comma separated list of allowed protocols for redirects. Available with cURL 7.85.0 or later. See https://curl.se/libcurl/c/CURLOPT_REDIR_PROTOCOLS_STR.html for details. + 18.18.020.3.0 The life-time for DNS cache entries. See https://curl.se/libcurl/c/CURLOPT_DNS_CACHE_TIMEOUT.html for details. diff --git a/res/res_parking.c b/res/res_parking.c index 644337de24..642e3ec39d 100644 --- a/res/res_parking.c +++ b/res/res_parking.c @@ -32,8 +32,10 @@ + 12.0.0 Options that apply to every parking lot + 12.0.0 Enables dynamically created parkinglots. If the option is enabled then the following variables can @@ -70,12 +72,15 @@ + 12.0.0 Defined parking lots for res_parking to use to park calls on + 12.0.0 The name of the context where calls are parked and picked up from. This option is only used if parkext is set. + 12.0.0 Extension to park calls to this parking lot. If this option is used, this extension will automatically @@ -96,9 +101,11 @@ + 12.0.0 If yes, the extension registered as parkext will park exclusively to this parking lot. + 12.0.0 Numerical range of parking spaces which can be used to retrieve parked calls. If parkext is set, these extensions @@ -108,15 +115,19 @@ + 12.0.0 If yes, this parking lot will add hints automatically for parking spaces. + 12.0.0 Amount of time a call will remain parked before giving up (in seconds). + 12.0.0 Which music class to use for parked calls. They will use the default if unspecified. + 12.0.0 Determines what should be done with the parked channel if no one picks it up before it times out. Valid Options: @@ -162,9 +173,11 @@ + 12.0.0 Timeout for the Dial extension created to call back the parker when a parked call times out. + 12.0.0 Context where parked calls will enter the PBX on timeout when comebacktoorigin=no The extension the call enters will prioritize the flattened peer name in this context. If the flattened peer name extension is unavailable, then the 's' extension in this context will be @@ -172,12 +185,14 @@ + 12.0.0 If the name of a sound file is provided, use this as the courtesy tone By default, this tone is only played to the caller of a parked call. Who receives the tone can be changed using the parkedplay option. + 12.0.0 Who we should play the courtesytone to on the pickup of a parked call from this lot @@ -190,30 +205,35 @@ + 12.0.0 Who to apply the DTMF transfer features to when parked calls are picked up or timeout. + 12.0.0 Who to apply the DTMF parking feature to when parked calls are picked up or timeout. + 12.0.0 Who to apply the DTMF hangup feature to when parked calls are picked up or timeout. + 12.0.0 Who to apply the DTMF MixMonitor recording feature to when parked calls are picked up or timeout. + 12.0.0 Rule to use when trying to figure out which parking space a call should be parked with. diff --git a/res/res_pjproject.c b/res/res_pjproject.c index 8826473262..de1acca13a 100644 --- a/res/res_pjproject.c +++ b/res/res_pjproject.c @@ -67,6 +67,7 @@ + 13.8.0 PJPROJECT to Asterisk Log Level Mapping Warnings and errors in the pjproject libraries are generally handled by Asterisk. In many cases, Asterisk wouldn't even consider them to @@ -78,24 +79,31 @@ 'log_mappings' or it won't be found. + 13.8.0 Must be of type 'log_mappings'. + 13.8.0 A comma separated list of pjproject log levels to map to Asterisk LOG_ERROR. + 13.8.0 A comma separated list of pjproject log levels to map to Asterisk LOG_WARNING. + 13.8.0 A comma separated list of pjproject log levels to map to Asterisk LOG_NOTICE. + 13.8.0 A comma separated list of pjproject log levels to map to Asterisk LOG_VERBOSE. + 13.8.0 A comma separated list of pjproject log levels to map to Asterisk LOG_DEBUG. + 16.21.018.7.0 A comma separated list of pjproject log levels to map to Asterisk LOG_TRACE. diff --git a/res/res_pjsip/pjsip_config.xml b/res/res_pjsip/pjsip_config.xml index 91d4994053..5cc60962c0 100644 --- a/res/res_pjsip/pjsip_config.xml +++ b/res/res_pjsip/pjsip_config.xml @@ -6,6 +6,7 @@ SIP Resource using PJProject + 12.0.0 Endpoint The Endpoint is the primary configuration object. @@ -29,6 +30,7 @@ + 12.2.0 Allow support for RFC3262 provisional ACK tags @@ -63,15 +65,18 @@ + 12.0.0 Condense MWI notifications into a single NOTIFY. When enabled, aggregate_mwi condenses message waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled, individual NOTIFYs are sent for each mailbox. + 13.0.0 Media Codec(s) to allow + 18.0.0 Codec negotiation prefs for incoming offers. @@ -133,6 +138,7 @@ + 18.0.0 Codec negotiation prefs for outgoing offers. @@ -195,6 +201,7 @@ + 18.0.0 Codec negotiation prefs for incoming answers. @@ -253,6 +260,7 @@ + 18.0.0 Codec negotiation prefs for outgoing answers. @@ -311,9 +319,11 @@ + 13.15.014.4.0 Enable RFC3578 overlap dialing support. + 18.17.020.2.0 Dialplan context to use for RFC3578 overlap dialing. Dialplan context to use for overlap dialing extension matching. @@ -324,12 +334,14 @@ + 12.0.0 AoR(s) to be used with the endpoint List of comma separated AoRs that the endpoint should be associated with. + 12.2.0 Authentication Object(s) associated with the endpoint This is a comma-delimited list of auth sections defined @@ -346,6 +358,7 @@ + 12.2.0 CallerID information for the endpoint Must be in the format Name <Number>, @@ -353,6 +366,7 @@ + 12.7.0 Default privacy level @@ -369,12 +383,15 @@ + 12.2.0 Internal id_tag for the endpoint + 12.0.0 Dialplan context for inbound sessions + 12.2.0 Mitigation of direct media (re)INVITE glare @@ -396,6 +413,7 @@ + 12.2.0 Direct Media method type Method for setting up Direct Media between endpoints. @@ -409,12 +427,15 @@ + 13.24.016.1.0 Accept Connected Line updates from this endpoint + 13.24.016.1.0 Send Connected Line updates to this endpoint + 12.2.0 Connected line method type Method used when updating connected line information. @@ -435,15 +456,19 @@ + 12.0.0 Determines whether media may flow directly between endpoints. + 12.0.0 Disable direct media session refreshes when NAT obstructs the media session + 13.0.0 Media Codec(s) to disallow + 12.2.0 DTMF mode This setting allows to choose the DTMF mode for endpoint communication. @@ -467,6 +492,7 @@ + 18.22.020.7.021.2.0 IP address used in SDP for media handling At the time of SDP creation, the IP address defined here will be used as @@ -479,6 +505,7 @@ + 13.8.0 Bind the RTP instance to the media_address If media_address is specified, this option causes the RTP instance to be bound to the @@ -487,12 +514,15 @@ + 12.0.0 Force use of return port + 12.0.0 Enable the ICE mechanism to help traverse NAT + 13.19.015.2.0 Way(s) for the endpoint to be identified Endpoints and AORs can be identified in multiple ways. This @@ -569,6 +599,7 @@ + 12.2.0 How redirects received from an endpoint are handled When a redirect is received from an endpoint there are multiple ways it can be handled. @@ -590,6 +621,7 @@ + 12.0.0 NOTIFY the endpoint when state changes for any of the specified mailboxes Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state @@ -604,15 +636,19 @@ + 13.24.015.7.0 An MWI subscribe will replace sending unsolicited NOTIFYs + 13.9.0 The voicemail extension to send in the NOTIFY Message-Account header + 12.0.0 Default Music On Hold class + 12.2.0 Authentication object(s) used for outbound requests This is a comma-delimited list of auth @@ -627,9 +663,11 @@ + 12.0.0 Full SIP URI of the outbound proxy used to send requests + 12.0.0 Allow Contact header to be rewritten with the source IP address-port On inbound SIP messages from this endpoint, the Contact header or an @@ -641,26 +679,33 @@ + 12.0.0 Allow use of IPv6 for RTP traffic + 12.0.0 Enforce that RTP must be symmetric + 12.0.0 Send the Diversion header, conveying the diversion information to the called user agent + 13.38.016.15.017.9.018.1.0 Send the History-Info header, conveying the diversion information to the called and calling user agents + 12.0.0 Send the P-Asserted-Identity header + 12.0.0 Send the Remote-Party-ID header + 13.4.0 Immediately send connected line updates on unanswered incoming calls. When enabled, immediately send 180 Ringing @@ -682,6 +727,7 @@ + 18.25.020.10.021.5.0 The tenant ID for this endpoint. Sets the tenant ID for this endpoint. When a channel is created, @@ -690,12 +736,14 @@ + 12.0.0 Minimum session timers expiration period Minimum session timer expiration period. Time in seconds. + 12.2.0 Session timers for SIP packets @@ -708,12 +756,14 @@ + 12.0.0 Maximum session timer expiration period Maximum session timer expiration period. Time in seconds. + 12.0.0 Explicit transport configuration to use This will force the endpoint to use the @@ -730,6 +780,7 @@ + 12.0.0 Accept identification information received from this endpoint This option determines whether Asterisk will accept identification from the endpoint from headers such as P-Asserted-Identity @@ -739,6 +790,7 @@ the endpoint. + 12.0.0 Send private identification details to the endpoint. This option determines whether res_pjsip will send private identification information to the endpoint. If no, @@ -752,12 +804,15 @@ provided in the request is private. + 12.0.0 Must be of type 'endpoint'. + 12.0.0 Use Endpoint's requested packetization interval + 12.0.0 Determines whether res_pjsip will use and enforce usage of AVPF for this endpoint. @@ -771,6 +826,7 @@ + 12.4.0 Determines whether res_pjsip will use and enforce usage of AVP, regardless of the RTP profile in use for this endpoint. @@ -783,6 +839,7 @@ + 12.4.0 Determines whether res_pjsip will use the media transport received in the offer SDP in the corresponding answer SDP. @@ -793,6 +850,7 @@ + 12.2.0 Determines whether res_pjsip will use and enforce usage of media encryption for this endpoint. @@ -812,6 +870,7 @@ + 13.1.0 Determines whether encryption should be used if possible but does not terminate the session if not achieved. @@ -820,6 +879,7 @@ + 13.5.0 Force g.726 to use AAL2 packing order when negotiating g.726 audio When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2 @@ -829,6 +889,7 @@ + 12.0.0 Determines whether chan_pjsip will indicate ringing using inband progress. @@ -841,6 +902,7 @@ + 12.2.0 The numeric pickup groups for a channel. Can be set to a comma separated list of numbers or ranges between the values @@ -848,6 +910,7 @@ + 12.2.0 The numeric pickup groups that a channel can pickup. Can be set to a comma separated list of numbers or ranges between the values @@ -855,6 +918,7 @@ + 12.2.0 The named pickup groups for a channel. Can be set to a comma separated list of case sensitive strings limited by @@ -862,6 +926,7 @@ + 12.2.0 The named pickup groups that a channel can pickup. Can be set to a comma separated list of case sensitive strings limited by @@ -869,6 +934,7 @@ + 12.0.0 The number of in-use channels which will cause busy to be returned as device state When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the @@ -876,6 +942,7 @@ + 12.0.0 Whether T.38 UDPTL support is enabled or not If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted @@ -883,6 +950,7 @@ + 12.2.0 T.38 UDPTL error correction method @@ -899,6 +967,7 @@ + 12.0.0 T.38 UDPTL maximum datagram size This option can be set to override the maximum datagram of a remote endpoint for broken @@ -906,6 +975,7 @@ + 12.0.0 Whether CNG tone detection is enabled This option can be set to send the session to the fax extension when a CNG tone is @@ -913,6 +983,7 @@ + 13.11.0 How long into a call before fax_detect is disabled for the call The option determines how many seconds into a call before the @@ -921,6 +992,7 @@ + 12.0.0 Whether NAT support is enabled on UDPTL sessions When enabled the UDPTL stack will send UDPTL packets to the source address of @@ -928,12 +1000,14 @@ + 12.0.0 Whether IPv6 is used for UDPTL Sessions When enabled the UDPTL stack will use IPv6. + 16.22.018.8.0 Bind the UDPTL instance to the media_adress If media_address is specified, this option causes the UDPTL instance to be bound to @@ -941,12 +1015,15 @@ + 12.0.0 Set which country's indications to use for channels created for this endpoint. + 12.0.0 Set the default language to use for channels created for this endpoint. + 12.0.0 Determines whether one-touch recording is allowed for this endpoint. record_on_feature @@ -954,6 +1031,7 @@ + 12.0.0 The feature to enact when one-touch recording is turned on. When an INFO request for one-touch recording arrives with a Record header set to "on", this @@ -967,6 +1045,7 @@ + 12.0.0 The feature to enact when one-touch recording is turned off. When an INFO request for one-touch recording arrives with a Record header set to "off", this @@ -980,63 +1059,79 @@ + 12.0.0 Name of the RTP engine to use for channels created for this endpoint + 12.0.0 Determines whether SIP REFER transfers are allowed for this endpoint + 13.2.0 Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number + 13.30.0 Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side + 12.0.0 String placed as the username portion of an SDP origin (o=) line. + 12.0.0 String used for the SDP session (s=) line. + 12.2.0 DSCP TOS bits for audio streams See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for more information about QoS settings + 12.2.0 DSCP TOS bits for video streams See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for more information about QoS settings + 12.0.0 Priority for audio streams See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for more information about QoS settings + 12.0.0 Priority for video streams See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for more information about QoS settings + 12.0.0 Determines if endpoint is allowed to initiate subscriptions with Asterisk. + 12.0.0 The minimum allowed expiry time for subscriptions initiated by the endpoint. + 13.18.014.7.0 Username to use in From header for requests to this endpoint. + 12.0.0 Username to use in From header for unsolicited MWI NOTIFYs to this endpoint. + 12.0.0 Domain to use in From header for requests to this endpoint. + 12.7.0 Verify that the provided peer certificate is valid This option only applies if media_encryption is @@ -1060,6 +1155,7 @@ + 12.7.0 Interval at which to renegotiate the TLS session and rekey the SRTP session This option only applies if media_encryption is @@ -1069,6 +1165,7 @@ + 15.2.0 Whether or not to automatically generate an ephemeral X.509 certificate @@ -1081,6 +1178,7 @@ + 12.2.0 Path to certificate file to present to peer This option only applies if media_encryption is @@ -1088,6 +1186,7 @@ + 12.2.0 Path to private key for certificate file This option only applies if media_encryption is @@ -1095,6 +1194,7 @@ + 12.2.0 Cipher to use for DTLS negotiation This option only applies if media_encryption is @@ -1105,6 +1205,7 @@ + 12.2.0 Path to certificate authority certificate This option only applies if media_encryption is @@ -1112,6 +1213,7 @@ + 12.2.0 Path to a directory containing certificate authority certificates This option only applies if media_encryption is @@ -1119,6 +1221,7 @@ + 12.2.0 Whether we are willing to accept connections, connect to the other party, or both. @@ -1139,6 +1242,7 @@ + 12.7.0 Type of hash to use for the DTLS fingerprint in the SDP. @@ -1152,6 +1256,7 @@ + 12.0.0 Determines whether 32 byte tags should be used instead of 80 byte tags. This option only applies if media_encryption is @@ -1159,6 +1264,7 @@ + 12.2.0 Variable set on a channel involving the endpoint. When a new channel is created using the endpoint set the specified @@ -1167,6 +1273,7 @@ + 13.5.0 Context to route incoming MESSAGE requests to. If specified, incoming MESSAGE requests will be routed to the indicated @@ -1175,6 +1282,7 @@ + 13.5.0 An accountcode to set automatically on any channels created for this endpoint. If specified, any channel created for this endpoint will automatically @@ -1182,6 +1290,7 @@ + 15.0.0 Respond to a SIP invite with the single most preferred codec (DEPRECATED) Respond to a SIP invite with the single most preferred codec rather than advertising all joint codec capabilities. This limits the other side's codec @@ -1195,6 +1304,7 @@ + 18.0.0 Preferences for selecting codecs for an incoming call. Based on this setting, a joint list of preferred codecs between those @@ -1221,6 +1331,7 @@ + 18.0.0 Preferences for selecting codecs for an outgoing call. Based on this setting, a joint list of preferred codecs between @@ -1252,6 +1363,7 @@ + 13.5.0 Number of seconds between RTP comfort noise keepalive packets. At the specified interval, Asterisk will send an RTP comfort noise frame. This may @@ -1260,6 +1372,7 @@ + 13.5.0 Maximum number of seconds without receiving RTP (while off hold) before terminating call. This option configures the number of seconds without RTP (while off hold) before @@ -1268,6 +1381,7 @@ + 13.5.0 Maximum number of seconds without receiving RTP (while on hold) before terminating call. This option configures the number of seconds without RTP (while on hold) before @@ -1276,6 +1390,7 @@ + 13.10.0 List of IP ACL section names in acl.conf This matches sections configured in acl.conf. The value is @@ -1283,6 +1398,7 @@ + 13.10.0 List of IP addresses to deny access from The value is a comma-delimited list of IP addresses. IP addresses may @@ -1292,6 +1408,7 @@ + 13.10.0 List of IP addresses to permit access from The value is a comma-delimited list of IP addresses. IP addresses may @@ -1301,6 +1418,7 @@ + 13.10.0 List of Contact ACL section names in acl.conf This matches sections configured in acl.conf. The value is @@ -1308,6 +1426,7 @@ + 13.10.0 List of Contact header addresses to deny The value is a comma-delimited list of IP addresses. IP addresses may @@ -1317,6 +1436,7 @@ + 13.10.0 List of Contact header addresses to permit The value is a comma-delimited list of IP addresses. IP addresses may @@ -1326,6 +1446,7 @@ + 13.11.0 Context for incoming MESSAGE requests. If specified, incoming SUBSCRIBE requests will be searched for the matching @@ -1335,12 +1456,14 @@ + 13.12.014.1.0 Force the user on the outgoing Contact header to this value. On outbound requests, force the user portion of the Contact header to this value. + 13.13.014.2.0 Allow the sending and receiving RTP codec to differ When set to "yes" the codec in use for sending will be allowed to differ from @@ -1349,6 +1472,7 @@ + 13.15.014.4.0 Enable RFC 5761 RTCP multiplexing on the RTP port With this option enabled, Asterisk will attempt to negotiate the use of the "rtcp-mux" @@ -1359,6 +1483,7 @@ + 13.17.014.6.0 Whether to notifies all the progress details on blind transfer Some SIP phones (Mitel/Aastra, Snom) expect a sip/frag "200 OK" @@ -1367,6 +1492,7 @@ + 13.17.014.6.0 Whether to notifies dialog-info 'early' on InUse&Ringing state Control whether dialog-info subscriptions get 'early' state @@ -1374,6 +1500,7 @@ + 15.0.0 The maximum number of allowed audio streams for the endpoint This option enforces a limit on the maximum simultaneous negotiated audio @@ -1381,6 +1508,7 @@ + 15.0.0 The maximum number of allowed video streams for the endpoint This option enforces a limit on the maximum simultaneous negotiated video @@ -1388,6 +1516,7 @@ + 15.0.0 Enable RTP bundling With this option enabled, Asterisk will attempt to negotiate the use of bundle. @@ -1396,6 +1525,7 @@ + 15.0.0 Defaults and enables some options that are relevant to WebRTC When set to "yes" this also enables the following values that are needed in @@ -1409,6 +1539,7 @@ + 13.18.014.7.015.1.0 Mailbox name to use when incoming MWI NOTIFYs are received If an MWI NOTIFY is received from this endpoint, @@ -1417,6 +1548,7 @@ + 13.22.015.5.0 Follow SDP forked media when To tag is different On outgoing calls, if the UAS responds with different SDP attributes @@ -1432,6 +1564,7 @@ + 13.22.015.5.0 Accept multiple SDP answers on non-100rel responses On outgoing calls, if the UAS responds with different SDP attributes @@ -1447,6 +1580,7 @@ + 13.23.015.6.0 Suppress Q.850 Reason headers for this endpoint Some devices can't accept multiple Reason headers and get confused @@ -1455,6 +1589,7 @@ + 13.26.016.3.0 Do not forward 183 when it doesn't contain SDP Certain SS7 internetworking scenarios can result in a 183 @@ -1465,6 +1600,7 @@ + 18.22.020.7.021.2.0 Enable STIR/SHAKEN support on this endpoint Enable STIR/SHAKEN support on this endpoint. On incoming INVITEs, @@ -1473,6 +1609,7 @@ + 16.26.018.12.019.4.0 STIR/SHAKEN profile containing additional configuration options A STIR/SHAKEN profile that is defined in stir_shaken.conf. Contains @@ -1480,6 +1617,7 @@ + 16.18.018.4.0 Skip authentication when receiving OPTIONS requests RFC 3261 says that the response to an OPTIONS request MUST be the @@ -1497,6 +1635,7 @@ + 21.0.0 The kind of security agreement negotiation to use. Currently, only mediasec is supported. @@ -1506,6 +1645,7 @@ + 21.0.0 List of security mechanisms supported. This is a comma-delimited list of security mechanisms to use. Each security mechanism @@ -1513,6 +1653,7 @@ + 16.28.018.14.019.6.0 Geolocation profile to apply to incoming calls This geolocation profile will be applied to all calls received @@ -1522,6 +1663,7 @@ + 16.28.018.14.019.6.0 Geolocation profile to apply to outgoing calls This geolocation profile will be applied to all calls received @@ -1531,9 +1673,11 @@ + 18.16.020.1.0 Send Advice-of-Charge messages + 20.11.021.6.022.1.0 Suppress playing MOH to party A if party B sends "sendonly" or "inactive" in an SDP @@ -1556,6 +1700,7 @@ + 12.0.0