Hello, I am new to Asterisk. I want to use asterisk for speech to text conversion.
I am using https://github.com/asterisk/aeap-speech-to-text for this. The node js application is running on port 9099.
**aeap.conf -**
[my-speech-to-text]
type=client
codecs=!all,ulaw
url=ws://127.0.0.1:9099
protocol=speech_to_text
**extensions.conf -**
[office-phones]
exten => 550,1,NoOp()
same => n,Answer()
same => n,SpeechCreate(my-speech-to-text)
same => n,SpeechStart()
same => n,SpeechBackground(hello-world)
same => n,Verbose(0,${SPEECH_TEXT(0)})
same => n,SpeechDestroy()
same => n,Hangup()
**pjsip.conf**
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
[alice-softphone]
type=endpoint
context=office-phones
disallow=all
allow=ulaw
auth=alice-auth
aors=alice-softphone
[alice-auth]
type=auth
auth_type=userpass
username=alice-softphone
password=Secret123
[alice-softphone]
type=aor
max_contacts=1
i am using asterisk version 21 for this, but when i am making a call from my softphone it logs
Executing [550@office-phones:1] NoOp(“PJSIP/bob-softphone-00000027”, “”) in new stack
** – Executing [550@office-phones:2] Answer(“PJSIP/bob-softphone-00000027”, “”) in new stack**
** > 0x7f257801ae80 – Strict RTP learning after remote address set to: 10.42.0.36:54717**
** > 0x7f257801ae80 – Strict RTP switching to RTP target address 10.42.0.36:54717 as source**
Got RTP packet from 10.42.0.36:54717 (type 95, seq 045978, ts 2006699797, len 000001)
Got RTP packet from 10.42.0.36:54717 (type 00, seq 045979, ts 2006699797, len 000160)
** – Executing [550@office-phones:3] SpeechCreate(“PJSIP/bob-softphone-00000027”, “my-speech-to-text”) in new stack**
** – Executing [550@office-phones:4] SpeechStart(“PJSIP/bob-softphone-00000027”, “”) in new stack**
** == Spawn extension (office-phones, 550, 4) exited non-zero on ‘PJSIP/bob-softphone-00000027’**
and my node server is not called as it doesn’t logs client connected.
Please help here.