Comments (5)
yes, I will look into this.
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The reason you got the error above was that you did not compile the v1p1beta1 stubs into the freeswitch executable.
I have added support for v1p1beta google speech in the 'google-update' branch of this repo. To build this branch, use the 'google-update' branch of the ansible-role-fsmrf repo, which has these changes
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I've merged these changes into the master branch
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Thank you!
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Hello again, I have reviewed your implementation of add_alternative_language_codes, I have made an error in my code. It is better to use the same logic as with hints (separated by comma), because google can accept several (seems up to 3) alternative languages.
Thank you!
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Related Issues (20)
- Facing issue with Azure Transcribe module HOT 2
- mod_aws_transcribe locks when calls is destroyed before transcription is stopped HOT 3
- How to play the audio to caller? HOT 1
- How to activate mod_dialogflow in fusionpbx HOT 2
- lws_glue.cpp:185 connection failed: read failed HOT 1
- Smart Formatting on Deepgram HOT 1
- add support for Azure speech segmentation
- add support for google voice activity timeout
- mod_google_transcribe: Feature Request: Support for `google::cloud::speech::v2` HOT 2
- mod_google_transcribe: Feature Request: Make the sample rate in `RecognitionConfig` configurable
- mod_google_transcribe: Bug: Calling transcribe2_function in mod_google_transcribe.c sometimes crashes HOT 2
- mod_google_transcribe: Feature Request: Allow google speech to text endpoint to be configurable
- deepgram websocket connection closed after few seconds of transcription HOT 4
- mod_audio_fork: Do we need to disconnect audio fork connection?
- Executed the last dialplan instruction after use dialogflow_start
- freeswitch restart issue HOT 3
- **/usr/local/freeswitch/mod/mod_audio_fork.so: undefined symbol: fork_session_send_text**
- mod_audio_fork: memory leak in mod audio fork
- Ability to audio_fork callee channel mono to a wss-url
- Deepgram module latency issue
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