CHANGELOG.md

develop

• Bugfix: Return correct recording uri for wav49 format with Asterisk.
• Change: Remove support for FreeSWITCH translator on Inbound EventSocket
• Bugfix: Return the correct error when the call is down when stopping an output rather than crashing the translator (#256)
• Bugfix: Handle more AMI error responses which mean the channel is not found

v2.7.5 - 2015-08-27

• Bugfix: Alternative fix for "Avoid race conditions in processing calls with interactions between them". The original fix in 2.7.4 introduced crashes relating to creating call actors at high call volume.

v2.7.4 - 2015-08-19

• YANKED
• Bugfix: Avoid race conditions in processing calls with interactions between them

v2.7.3 - 2015-08-18

• Bugfix: Handle correct event for confirming that a component stop was completed on Asterisk 13

v2.7.2 - 2015-08-10

• Bugfix: Process joined events on Asterisk 13 in any order, avoiding Join command timeouts

v2.7.1 - 2015-07-14

• Bugfix: Ensure components are deregistered from asterisk translator once the call is ended (#250)

v2.7.0 - 2015-06-09

• Feature: Support for Asterisk 13 (AMI v2)
• Feature: Pass all possible MRCP recognition headers through Asterisk #244
• Bugfix: Handle an illegal AMI response by Asterisk as a failure with code -1
• Bugfix: Ensure a useful error is raised when attempting to join to a call which doesn't exist #529
• Bugfix: Support SSML in MRCPRecog case on Asterisk. This is the case where a prompt component wants to render using Asterisk and recognise using UniMRCP. This worked fine with a uri-list which already has test coverage, but threw (and silently swallowed) an exception for an SSML doc (as provided by Adhearsion) resulting in a hung call with no audio, and then a timeout exception in Adhearsion. #241

v2.6.0 - 2015-02-01

• Feature: Support recognition-timeout settings on UniMRCP-based ASR on Asterisk (#228)
• Feature: Implement redirect command on Asterisk (#219)
• Feature: Allow arbitrary media headers (#237)
• Bugfix: Complete 1-to-1 mapping of UniMRCP RECOG_COMPLETION_CAUSE values to Punchblock::Complete events

v2.5.3 - 2014-09-22

• Bugfix: Prevent Asterisk translator death due to dead DTMF recognizers (#221, adhearsion/adhearsion#479)
• Bugfix: Be more intelligent about only stripping true file extensions off filenames for playback on Asterisk (adhearsion/adhearsion#482)
• Bugfix: Handle a corner case crash where a recognition request is interrupted directly after a successful recognition has completed
• Bugfix: Avoid cases where executing AGI commands (reported as sending SIP messages) might block indefinitely due to a race condition in fetching their response value

v2.5.2 - 2014-04-03

• Bugfix: No longer confound start and stop beeps on record commands (#220)

v2.5.1 - 2014-03-31

• Bugfix: Ensure DTMF recognisers get shut down when finishing
• Bugfix: Hangup reason on Asterisk was :hangup, should be :hungup. See the Rayo spec.

v2.5.0 - 2014-03-05

• Feature: Support language, sensitivity and minimum confidence on UniMRCP-based ASR on Asterisk
• Feature: Support sending messages to calls (eg SIP MESSAGE) on Asterisk

v2.4.2 - 2014-03-04

• Bugfix: Rayo events should not include their timestamp in comparison. This is not useful in applications, and makes testing more difficult.

Note: 2.4.1 was not released due to a tagging mishap. Its contents was released as 2.4.2.

v2.4.0 - 2014-03-01

• Feature: Add support for requesting calls with a specific URI
• Feature: Allow generation of a random call URI for a client
• Feature: Rayo events should be timestamped with dispatch or receipt time (#213)

v2.3.1 - 2014-02-13

• Bugfix: Ensure commands can be associated on the wire even before they're executed
• Bugfix: Ensure a command is always transitioned to is requested state prior to receiving a response

v2.3.0 - 2014-01-30

• Feature: Add input-timers-started event
• Feature: Allow client to send XMPP messages through Blather
• Bugfix: Ensure that max duration expiration of Asterisk recordings does not crash the translator when it extends past the call's destruction

v2.2.2 - 2014-01-21

• Bugfix: Delay sending Answered event on call until AsyncAGI is invoked

v2.2.1 - 2014-01-17

• Bugfix: Punchblock::Event::Complete::Reason is now a Punchblock::Event

v2.2.0 - 2014-01-15

• Feature: Support Rayo CPA and Fax specifications
• Feature: Implement a simple filter for AMI events -> Rayo events in Punchblock::Translator::Asterisk.event_filter=
• Feature: Output on Asterisk supports a new :native_or_unimrcp renderer which allows for fallback of output rendering from Asterisk native when possible, to a TTS engine when it's not.
• Feature: Support output repeat-times on Asterisk.
• Bugfix: Remove per-call/component actors from Asterisk translator for performance/stability super-charge

v2.1.1 - 2013-12-19

• Bugfix: Allow sending string SSML docs via Rayo
• Bugfix: Ensure that joined calls on Asterisk do not have implicitly linked lifecycles. Previously, joinees would be hungup when the joiner exited the bridge.

v2.1.0 - 2013-11-12

• Feature: Support RubySpeech builtin grammars on Asterisk and FreeSWITCH
• Update: Update to stable release of Virtus

v2.0.2 - 2013-10-17

• Bugfix: Reject commands against components which have finished on Asterisk, and garbage collect them
• Bugfix: Register/lookup components by their full URI rather than component ID since the component ID may only be unique per call
• Bugfix: Hold back Virtus dependency to avoid API-breaking changes

v2.0.1 - 2013-09-17

• Bugfix: Allow audio file URIs with file extensions on Asterisk
• Bugfix: Input timers were being started before output finished on Asterisk composed prompts
• Bugfix: Input initial timers were being started on Asterisk composed prompts even if the prompt was barged
• Bugfix: Output was being interrupted on Asterisk composed prompts at every DTMF keypress, even if the output was already finished

v2.0.0 - 2013-08-29

• Feature: Compliance with v0.2 of the published Rayo spec (http://xmpp.org/extensions/xep-0327.html)
• Feature: Add support for Rayo Prompt component
• Feature: Added FS support for initial timeout and final timeout on Record
• Change: Models are now plain ruby objects, not XML nodes, and are imported from/exported to XML when necessary for communicating over XMPP.
• Change: #headers and AMI #attributes now do not have their names modified. A header of 'Call-ID' will no longer be modified to :call_id.
• Change: AMI Events/Actions now have #headers(=) rather than #attributes(=)
• Change: Remove event queue
• Change: Removed media_engine and default_voice settings
• Bugfix: Reconnect dead Asterisk streams correctly
• Bugfix: Include AMI response text_body in AMI component complete events
• Bugfix: Avoid crashing translators (Asterisk or FreeSWITCH) by instructing them to call back to terminated Call objects
• Bugfix: Detect MRCPSynth failure in output component
• Bugfix: Handle AMI errors indicating dead channels correctly

v1.9.4 - 2013-06-08

• Bugfix: Finish more setup before sending output ref on Asterisk
• Bugfix: Allow early media TTS on Asterisk in addition to audio playback
• Bugfix: Correctly mark Asterisk calls as answered after successfully executing an answer command

v1.9.3 - 2013-05-16

• Bugfix: Improve error messages when trying to execute stop commands on components in an invalid state

v1.9.2 - 2013-05-10

• Bugfix: We were raising an exception on connection shutdown due to waiting for the connection to end incorrectly.
• Bugfix/Perf: FreeSWITCH Call actors were being kept alive after hangup for no reason
• Bugfix/Perf: FreeSWITCH component complete events were looping out of the actor
• Perf: We were wasting CPU cycles listening to all ES events when we really don't need to

v1.9.1 - 2013-05-08

• Bugfix: AMI errors indicating dead channels were not being handled correctly
• Bugfix: We were broken on Celluloid 0.14 due to changes in block execution semantics between actors

v1.9.0 - 2013-05-03

• Feature: Use RubyAMI 2.0 with a single AMI connection.
• Feature: Cache channel variables on Asterisk calls.
• Feature: Allow optional sending of end event when breaking from AsyncAGI on Asterisk. This enables dialplan handback. Only triggers if the channel variable 'PUNCHBLOCK_END_ON_ASYNCAGI_BREAK' is set.
• Bugfix: Avoid DTMF recognizer failures and race conditions by bringing DTMFRecognizer back into the Input component actor.
• Bugfix: Catch Asterisk AMI errors in all cases and fail accordingly, instead of ploughing ahead in the face of adversity.
• Bugfix: Improve performance of Asterisk implementation by no longer spinning up a component actor for AGI command execution.

v1.8.2 - 2013-04-19

• Bugfix: Input initial timeout was being set as a float rather than an integer

v1.8.2 - 2013-04-19

• Bugfix: Input initial timeout was being set as a float rather than an integer

v1.8.1 - 2013-03-25

• Bugfix: FreeSWITCH was requiring a from attribute on a dial command
• Bugfix: Asterisk translator now properly checks for existence of the recordings directory
• Bugfix: Components should transition state before unblocking
• Bugfix: Asterisk joins are now more robustly responded to when the join begins
• Bugfix: On FreeSWITCH, only events relating to bridge start/end should be delivered to bridged calls
• Bugfix: On FreeSWITCH, a voice value on an audio-only output component should not prevent execution
• Bugfix: XMPP Ping should be an IQ get, not set
• Bugfix: Stop command should be in Rayo ext namespace
• Bugfix: XMPP specs were mistakenly resetting the logger object for other tests.
• CS: Avoid Celluloid deprecation warnings

v1.8.0 - 2013-01-10

• Feature: Join command now enforces a list of valid direction attribute values
• Feature: Added support for media direction to the Record component
• Feature: Record direction support on FS
• Bugfix: Fixed answering during early media on FS
• Bugfix: Doing multiple recordings on Asterisk during the lifetime of a call was crashing Punchblock

v1.7.1 - 2012-12-17

• Bugfix: Deal with nil media engines on FS/* properly

v1.7.0 - 2012-12-13

• Feature: Support for the renderer attribute added to the Output component.
• Feature: FreeSWITCH and Asterisk translators now use the :renderer attribute on Output
• Bugfix: Fixed scenario where executing the ANSWER application on FreeSWITCH on an already answered call caused FS to stop accepting commands.
• Bugfix: Plug a severe memory leak
• Bugfix: Raise an error immediately if trying to execute an invalid media engine on Asterisk
• Bugfix: Handle a wider variety of types when configuring media engines on Asterisk and FreeSWITCH, such as Strings instead of Symbols

v1.6.1 - 2012-11-14

• Bugfix: Safer component attribute writer conversion

v1.6.0 - 2012-11-14

• Feature: Set dial headers on FreeSWITCH originate command (SIP only)
• Feature: Set dial headers on Asterisk originate command (SIP only)
• Bugfix: Headers were being re-written downcased and with underscores
• Bugfix: Ensure all numeric component attributes are written as the correct type

v1.5.3 - 2012-11-01

• Doc: Add link to docs for unrenderable doc error

v1.5.2 - 2012-10-25

• Bugfix: Use correct GRXML content type
• Bugfix: Fix UniMRCP for documents containing commas
• Bugfix: Bump Celluloid dependency to avoid issues with serialising AMI

v1.5.1 - 2012-10-11

• Update: Bump Celluloid dependency
• Bugfix: Input grammars referenced by URL now no longer specify a content type
• Bugfix: FreeSWITCH Dial#from values now parsed more flexibly

v1.5.0 - 2012-10-01

• Feature: Input component now supports grammar URLs
• Bugfix: Hanging up Asterisk calls now correctly specifies normal clearing cause
• Doc: Fix a bunch of API documentation

v1.4.1 - 2012-09-06

• Bugfix: Cleaning up DTMF handlers for input components with a dead call should not crash on FreeSWITCH
• Bugfix: Reduced a race condition on FreeSWITCH when dispatching events from calls to dead components
• Bugfix: Events relevant to bridged channels were not being routed to the call
• Bugfix: Components using #stop_by_redirect now return an error response if stopped when they are complete
• Bugfix: Hold back ruby_ami and ruby_fs dependencies pending fixes for Celluloid 0.12.0

v1.4.0 - 2012-08-07

• Feature: FreeSWITCH support (mostly complete, experimental, proceed with caution)
• Bugfix: Report the correct caller ID in offers from Asterisk
• Bugfix: Strip out caller ID name from dial commands on Asterisk

v1.3.0 - 2012-07-22

• Change: Asterisk output now uses Playback rather than STREAM FILE
• Feature: The recordings dir is now checked for existence on startup, and logs an error if it is not there. Asterisk only.
• Feature: Punchblock now logs an error if it was unable to add the redirect context on Asterisk on startup.
• Feature: Output component now exposes #recording and #recording_uri for easy access to record results.
• Feature: Early media support for Asterisk, using Progress to start an early media session
• Feature: Output component on Asterisk now supports early media. If the line is not answered, it runs Progress followed by Playback with noanswer.
• Feature: Record component on Asterisk now raises if called on an unanswered call
• Feature: Input component on Asterisk works the same whether the call is answered or unanswered
• Feature: AMI events are emitted to the relevant calls
• Feature: Simpler method of getting hold of a new client/connection
• Bugfix: AMI events are processed in order by the translator
• Bugfix: Asterisk calls and components are removed from registries when they die
• Bugfix: Commands for unknown calls/components respond with the correct :item_not_found name
• Bugfix: AMI events relevant to a particular call are emitted by that call to the client
• Bugfix: Asterisk calls send an error complete event for their dying components
• Bugfix: Asterisk translator sends an error end event for its dying calls
• Bugfix: Use the primitive version of AGI ANSWER, rather than an app
• Bugfix: Outbound calls which never begin progress on Asterisk end with an error
• Bugfix: Asterisk now responds correctly to unjoin commands
• Bugfix: Allow nil reject reasons
• Bugfix: Asterisk translator now does NOT answer the call automatically when Output, Input or Record are used.

v1.2.0 - 2012-04-29

• Feature: Basic support for record component on Asterisk, using MixMonitor. Currently unsupported options include: start_paused, initial_timeout, final_timeout. Hints are additionally not supported, and recordings are stored on the * machine's local filesystem.

v1.1.0 - 2012-04-26

• Feature: Implement Reject on Asterisk
• Bugfix: No longer generate warnings
• Bugfix: Set 'to' attribute on an offer from Asterisk to something useful if the dnid is 'unknown'
• Bugfix: Include caller ID name in 'from' attribute on an offer from Asterisk
• Bugfix: Removed media engine switching on Asterisk Input component - fixes broken input when using app_swift or unimrcp for output
• Update: Better dependency version fixing

v1.0.0 - 2012-04-11

• Stable release :D
• Bugfix: Any issue in compiling an output document into executable elements on Asterisk should return an unrenderable doc error

v0.12.0 - 2012-03-30

• API Change: #call_id and #mixer_name attributes changed to #target_call_id and #target_mixer_name
• API Change: #other_call_id attributes changed to #call_id to better align with Rayo

v0.11.0 - 2012-03-29

• Feature: Input & Output components on Asterisk now responds to a Stop command
• Feature: started/stopped-speaking events are now handled
• Bugfix: Asterisk output component considers an SSML doc w/ a string node w/o spaces to be a filename
• Bugfix: ProtocolError should behave like a normal exception, just with extra attributes

v0.10.0 - 2012-03-19

• Feature: app_swift is now supported on Asterisk with a media_engine type of :swift
• Feature: Asterisk calls now support the Join API
• Feature: On Asterisk, Punchblock creates a context and extension to redirect calls to
• Bugfix: Unjoining calls now redirects both legs
• Bugfix: Unlink events on Asterisk correctly send Unjoin Punchblock events
• Bugfix: The Asterisk translator now ignores calls to 'h' or of type 'Kill'
• Bugfix: Handle more XMPP connection errors gracefully
• Bugfix: The XMPP connection ready event is now available to external handlers
• Bugfix: The Asterisk connection now passes the :media_engine option down to the translator
• Bugfix: Connections now always respond to #connected?
• Bugfix: Connection termination handled gracefully on Asterisk

v0.9.2 - 2012-02-18

• Feature: Asterisk calls receiving media commands are implicitly answered
• Bugfix: Unrenderable output documents on Asterisk should return a sensible error
• Bugfix: Log the target of commands correctly
• Bugfix: Do not wrap exceptions in ProtocolError

v0.9.1 - 2012-01-30

• Bugfix: Closing an disconnected XMPP connection is a no-op

v0.9.0 - 2012-01-30

• Bugfix: Remove the rest of the deprecated Tropo components (conference)
• Feature: Outbound dials on Asterisk now respect the dial timeout
• Bugfix: Registering stanza handlers on an XMPP connection now sets them in the correct order such that they do not override the internally defined handlers

v0.8.4 - 2012-01-19

• Bugfix: End, Ringing & Answered events are allowed to have headers
• Feature: Dial commands may have an optional timeout

v0.8.3 - 2012-01-17

• Feature: Return an error when trying to execute a command for unknown calls/components or when not understood
• Feature: Log calls/translator shutting down
• Feature: Calls and components should log their IDs
• Feature: Components marked as internal should send events directly to the component node
• Bugfix: Fix Asterisk Call and Component logger IDs
• Bugfix: Fix a stupidly high log level
• Bugfix: AGI commands executed by a call/component that are a translation of a Rayo command should be marked internal
• Bugfix: Asterisk components should sent events via the connection
• Bugfix: Shutting down an asterisk connection should do a cascading shutdown of the translator and all of its calls
• Bugfix: Component actors should be terminated once they've sent a complete event
• Bugfix: Component events should be sent with the call ID
• Bugfix: AMIAction components do not have a call
• Bugfix: Add test coverage for comparison of complete events
• Bugfix: A call being hung up should terminate the call actor
• Bugfix: Fix a mock expectation error in a test

v0.8.2 - 2012-01-10

• Feature: Support outbound dial on Asterisk
• Bugfix: Asterisk hangup causes should map to correct Rayo End reason

v0.8.1 - 2012-01-09

• Feature: Support DTMF Input components on Asterisk

v0.8.0 - 2012-01-06

• Feature: Expose Blather's connection timeout config when creating a Connection::XMPP
• Bugfix: Remove some deprecated Tropo extension components
• Bugfix: Remove reconnection logic since it really belongs in the consumer
• Feature: Raise a DisconnectedError when a disconnection is detected

v0.7.2 - 2011-12-28

• Feature: Allow sending commands to mixers easily
• Feature: Allow configuration of Rayo XMPP domains (root, call and mixer)
• Feature: Log Blather messages to the trace log level
• Feature: Return an error when trying to execute an Output on Asterisk with unsupported options set
• Feature: Add basic support for media output via MRCPSynth on Asterisk
• API change: Rename mixer_id to mixer_name to align with change to Rayo
• Bugfix: Handle and expose RubySpeech GRXML documents on Input/Ask properly
• Bugfix: Compare ProtocolErrors correctly
• Bugfix: Asterisk media output should default to Asterisk native output (STREAM FILE)
• Bugfix: An Output node's default max_time value should be nil rather than zero

v0.7.1 - 2011-11-24

• [FEATURE] Add Connection#not_ready!, to instruct the server not to send any more offers.
• [BUGFIX] Translate all exceptions raised by the XMPP connection into a ProtocolError
• [UPDATE] Blather dependency to >= 0.5.9

v0.7.0 - 2011-11-22

• Bugfix: Some spec mistakes
• Feature: Allow execution of actions against global components on Asterisk
• API change: The console has been removed
• API change: Components no longer expose a FutureResource at #complete_event, and instead wrap its API in the same way as #response and #response=. Any consumer code which does some_component.complete_event.resource or some_component.complete_event.resource= should now use some_component.complete_event and some_component.complete_event=
• Feature: Added the max-silence attribute to the Input component
• Bugfix: Bump the Celluloid dependency to avoid spec failures on JRuby and monkey-patching for mockability
• API change: Event handlers registered on components are no longer triggered by incoming events internally to Punchblock. These events must be consumed via a Client's event handlers or event queue and manually triggered on a component using ComponentNode#trigger_event_handler

v0.6.2

Feature: Added basic support for running Punchblock apps on Asterisk. Calls coming in to AsyncAGI result in the client receiving an Offer, hangup events are sent, and accept/answer/hangup commands work.

API change: The logger is now set using Punchblock.logger= rather than as a hash key to Connection.new

v0.6.1

• Feature: Allow instructing the connection we are ready. An XMPP connection will send initial presence with a status of 'chat' to the rayo domain
• Bugfix: When running on Asterisk, two FullyBooted events will now trigger a connected event
• Bugfix: No longer ignore offers from the specified rayo domain on XMPP connections
• Feature: Tag all event objects with the XMPP domain they came from

v0.6.0

• API change: Punchblock consumers now need to instantiate both a Connection and a Client (see the punchblock-console gem for an example)
• Feature: Added a Connection for Asterisk, utilising RubyAMI to open an AMI connection to Asterisk, and allowing AMI actions to be executed. AMI events are handled by the client event handler.
• Deprecation: The punchblock-console and the associated DSL are now deprecated and the punchblock-console gem should be used instead

v0.5.1

API change: Connections now raise a Punchblock::Connection::Connected instance as an event, rather than the class itself

v0.5.0

• Refactoring/API change: Client and connection level concerns are now separated, and one must create a Connection to be passed to a Client on creation. Client now has the choice between an event queue and guarded event handlers.

v0.4.3

• Feature: Support for meta-data on recordings (size & duration)
• Feature: Allow specifying all of Blather's setup options (required to use PB as an XMPP component)
• Bugfix: Rayo events are discarded if they don't come from the specified domain
• Bugfix: Component execution in the sample DSL now doesn't expect events on the main queue

v0.4.2

• Bugfix: Conference complete event was not being handled

v0.4.1

• Feature/API change: Components no longer have an event queue, but instead it is possible to define guarded event handlers via #register_event_handler

v0.4.0

• First public release