* sending jambonz:error when the incoming message is not parsable
https://github.com/jambonz/jambonz-feature-server/issues/1094
* writing an alert when incoming paylod is invalid
* added content to the jambonz:error payload
* removing video media from sdp if the call is an audio call. This is to avoid sending video media to destination if the incoming call is an audio call
* calling removeVideoSdp only when the environment variable JAMBONES_VIDEO_CALLS_ENABLED_IN_FS is set to true, this will ensure there are no regression issues for audio calls
* fixed jslint errors
* fixed dial transcribe is not able to received final transcribe when close call.
* wip
* fix review comment
* support call session delay detroy ep when current task is transcribe
* wip
* wip
* fixed review comments
* fixed review comments
* support kill dial if sd ep is media timeout
* support kill dial if sd ep is media timeout
* support kill dial if sd ep is media timeout
* add media timeout reason header to bye message
* wip
* wip
* make configuration for freeswitch media timeout
* make configuration for freeswitch media timeout
* wip
* feature server should send USER call to the sbc sip that is connect with the user
* feature server should send USER call to the sbc sip that is connect with the user
* feature server should send USER call to the sbc sip that is connect with the user
* fix review comment
* add env variable to enable the feature
* add env variable to enable the feature
* add env variable to enable the feature
* minor test update
---------
Co-authored-by: Dave Horton <daveh@beachdognet.com>
* include X-CID for dial outbound if the call-session is outbound
* include X-CID for dial outbound if the call-session is outbound
* include X-CID for dial outbound if the call-session is outbound
* include X-CID for dial outbound if the call-session is outbound
* initial support for coaching mode in conference
* wip
* wip
* add support for answer verb
* wip
* wip
* wip
* wip
* wip
* updates to rename option to dub
* wip
* wip
* wip
* update verb-specs
* wip
* wip
* wip
* wip
* wip
* wip
* wip
* wip
* add option to boost audio signal in main channel
* wip
* wip
* wip
* wip
* wip
* wip
* for now, bypass use of streaming apis when generating tts audio for dub tracks
* add nested dub to dial
* wip
* add support for filler noise
* kill filler noise when gather killed
* wip
* wip
* while using sayOnTrack, we have to enclose the say command in double quotes
* disableTtsStreaming = false
* allow transcribe of b leg only on dial verb
* dub.say can either be text or object like say verb with text and synthesizer
* remove loop for sayOnTrack
* update speech-utils
* fixes for testing transcribe verb and support for dub and boostAudioSignal in lcc commands
* add dial.boostAudioSignal
* fix bug where session-level recognizer settings incorrectly overwrite verb-level settings
* update verb specs
* update dial to support array of dub verbs
* fix bug setting gain
* lint
* wip
* update speech-utils
* use new endpoint methods for mod_dub
---------
Co-authored-by: Dave Horton <daveh@beachdognet.com>
* fix release freeswitch media properly
* if a leg is opus, modify b leg offer opus first
* if a leg is opus, modify b leg offer opus first
* wip
* wip
* fix review comments
* fix review comments
* fix review comments
* add methods to lookupTrunkbyPhone
* change the object name
* fix typo in readme
* export method with return
* add checks to dial verb
* sans extra spaces
* change the variable name for lookup
* fixes from testing
* modify Task#exec to take resources as an object rather than argument list
* pass 2 endpoints to Transcribe when invoked in a SipRec call session
* logging
* change siprec invite to sendrecv just so freeswitch does not try to reinvite (TODO: block outgoing media at rtpengine)
* Config: when enabling recording, block until siprec dialog is established
* missed play verb in commit 031c79d
* linting
* bugfix: get final transcript in siprec call
