Speech to Text Issue

suhatakbulak · October 23, 2024, 12:01pm

The user presses 1 in the IVR menu and is transferred to the extension number, and I accept the call. Sometimes what they say appears in the console, but frequently, it returns an error. The error message only points to the SpeechCreate() function, and I don’t understand why.

[custom-ivr2-agi]
exten => s,1,NoOp("Press 1")
   same => n,Answer()
   same => n,Playback(tr/custom/aydinlatma_metini)
   same => n,SpeechCreate(my-speech-to-text)
   same => n,SpeechStart()
   same => n,SpeechBackground(tr/custom/ttsmaker-file-2024-10-23-12-4-21)
   same => n,Verbose(0,${SPEECH_TEXT(0)})
   same => n,SpeechDestroy()
   same => n,Hangup()

144	[2024-10-23 11:57:01] VERBOSE[7065][C-00000009] pbx.c: Executing [s@ivr-2:12] GotoIf("PJSIP/anonymous-00000008", "0?#,1") in new stack	
145	[2024-10-23 11:57:01] VERBOSE[7065][C-00000009] pbx.c: Executing [s@ivr-2:13] GotoIf("PJSIP/anonymous-00000008", "0?t,1") in new stack	
146	[2024-10-23 11:57:01] VERBOSE[7065][C-00000009] pbx.c: Executing [s@ivr-2:14] ExecIf("PJSIP/anonymous-00000008", "0?Set(LOCALEXT=1)") in new stack	
147	[2024-10-23 11:57:01] VERBOSE[7065][C-00000009] pbx.c: Executing [s@ivr-2:15] GotoIf("PJSIP/anonymous-00000008", "0?i,1") in new stack	
148	[2024-10-23 11:57:01] VERBOSE[7065][C-00000009] pbx.c: Executing [s@ivr-2:16] GotoIf("PJSIP/anonymous-00000008", "0?from-did-direct-ivr,1,1") in new stack	
149	[2024-10-23 11:57:01] VERBOSE[7065][C-00000009] pbx.c: Executing [s@ivr-2:17] Goto("PJSIP/anonymous-00000008", "1,1") in new stack	
150	[2024-10-23 11:57:01] VERBOSE[7065][C-00000009] pbx.c: Executing [1@ivr-2:1] Set("PJSIP/anonymous-00000008", "__ivrreturn=0") in new stack	
151	[2024-10-23 11:57:01] VERBOSE[7065][C-00000009] pbx.c: Executing [1@ivr-2:2] Goto("PJSIP/anonymous-00000008", "custom-ivr2-agi,s,1") in new stack	
152	[2024-10-23 11:57:01] VERBOSE[7065][C-00000009] pbx.c: Executing [s@custom-ivr2-agi:1] NoOp("PJSIP/anonymous-00000008", ""Çağrı 1 de"") in new stack	
153	[2024-10-23 11:57:01] VERBOSE[7065][C-00000009] pbx.c: Executing [s@custom-ivr2-agi:2] Answer("PJSIP/anonymous-00000008", "") in new stack	
154	[2024-10-23 11:57:01] VERBOSE[7065][C-00000009] pbx.c: Executing [s@custom-ivr2-agi:3] Playback("PJSIP/anonymous-00000008", "tr/custom/aydinlatma_metini") in new stack	
159	[2024-10-23 11:57:14] VERBOSE[7065][C-00000009] pbx.c: Executing [s@custom-ivr2-agi:4] SpeechCreate("PJSIP/anonymous-00000008", "my-speech-to-text") in new stack	
160	[2024-10-23 11:57:14] VERBOSE[7065][C-00000009] pbx.c: Executing [s@custom-ivr2-agi:5] SpeechStart("PJSIP/anonymous-00000008", "") in new stack	
161	[2024-10-23 11:57:14] VERBOSE[7065][C-00000009] pbx.c: Spawn extension (custom-ivr2-agi, s, 5) exited non-zero on 'PJSIP/anonymous-00000008'

jcolp · October 23, 2024, 12:22pm

You haven’t stated what engine/functionality you are using to do speech to text, but the most likely cause is that the underlying speech engine failed to initialize/create the speech resource.

suhatakbulak · October 23, 2024, 1:56pm

[my-speech-totext]
type=client
codecs=!all,ulaw
url=ws://127.0.0.1:9099
protocol=speech__to_text

jcolp · October 23, 2024, 1:58pm

Okay, so you’re using AEAP I assume then. I would suggest looking at whatever is on the other side and seeing if anything stands out as to a failure there, and also examine the underlying JSON traffic over the websocket.

suhatakbulak · October 23, 2024, 2:26pm

Everything seems fine on the WebSocket side, and port 9099 is open. This issue doesn’t occur all the time; sometimes the connection is successful, and the translation completes. Also, I’d like to ask if I need to update Asterisk for the @vosk-api library (at least to compile res_speech_vosk.so)

jcolp · October 23, 2024, 2:29pm

Not everything is fine there. The output shows an error response, “codec alaw not supported” so a request was done with alaw as the codec, why that is I don’t know. The configuration you’ve provided should limit it to ulaw.

Vosk is an outside project, someone else may have input on it.

What version of Asterisk are you using?

suhatakbulak · October 23, 2024, 2:44pm

I am using the master version on GitHub. No connections are coming through

jcolp · October 23, 2024, 2:46pm

Were you on a prior version, and now you’ve changed it?

jcolp · October 23, 2024, 2:48pm

And you’ll probably need to get an Asterisk debug log and examine the output to see if it gives any hints.

system · November 23, 2024, 2:48pm

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