Corrects the parameter passed to _lccBoostAudioSignal to use
opts.boostAudioSignal instead of the entire opts object, ensuring
the boostAudioSignal option works correctly.
🤖 Generated with [Claude Code](https://claude.com/claude-code)
Co-authored-by: Claude Sonnet 4.5 <noreply@anthropic.com>
* fix: propagate SIP 488 error to SBC on endpoint allocation failure
When FreeSWITCH returns a SIP 488 'Not Acceptable Here' error during
endpoint allocation (e.g., codec incompatibility), this error was not
being propagated back to the SBC/client. Instead, the call would wait
indefinitely for websocket commands or return a generic 603 response.
Implementation:
- In _evalEndpointPrecondition(), detect SipError by checking
err.type === 'SipError' or err.name === 'SipError'
- Extract the SIP status code (e.g., 488), reason, and the Reason
header from the error response (e.g., Q.850;cause=88;text=INCOMPATIBLE_DESTINATION)
- Send the SIP error response immediately to the SBC with:
- X-Reason header: endpoint allocation failure details
- Reason header: original Q.850 cause from FreeSWITCH
- Notify call status change as Failed with proper SIP status
- Release the call immediately instead of waiting for commands
Also added fallback handling in InboundCallSession._onTasksDone() to
propagate the stored error if immediate send was not possible.
* wip
* Simplify SipError check to only use err.name
* refactor recording
removed the test of `(this.cs.accountInfo.account.record_all_calls || this.cs.application.record_all_calls` from backround-task-manager.jsL138 as this check is already done in call-session.js at Line 3007, also allows us to start the record from update or config verbs
* handle start recording for a call that is not yet answered
* return false if not changing recording state
* different check for status
* set hasRecording flag on callInfo when starting
* update redis on recording start
* lint
* update dependency
Update sentenceEndRegex to treat the following as sentence boundaries: ASCII .!? followed by whitespace or end-of-text; Arabic question mark (؟) and full stop (۔) with the same rule; Japanese 。, ！, ？ treated as boundaries regardless of following character; and double newlines (\n\n). This improves streaming chunking for mixed-language content.
* fix race condition where gather resolves with speech transcript but timeout timer gets set after the resolve and is left running after gather completes
* remove unneeded line of code
* compare sdp for transcoding
* refactor sdp check for leading codec
* fix reference to epOther
* minor changes
* minor
* fix#1447
* fix security issue
* use convenience getter appIsUsingWebsockets in CallSession
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Co-authored-by: Dave Horton <daveh@beachdognet.com>
* fix say verb does not close streaming when finish say
* wip
* wip
* ttsStreamingBuffer reset eventHandlerCount after remove listeners
* only send tokens to module if connected
* wip
* sent stream_open when successfully connected to vendor
* fixed gather does not start timeout on bargin
* with previous change, no need to emit playDone since no where in the code are we listening for it
---------
Co-authored-by: Dave Horton <daveh@beachdognet.com>
