Asterisk Crash after script restart

thanks for your source code, there is only one example how to use aeap with google speech api and its yours. my problem is that when I stop and restart the index script and try to call again, the asterisk crash with segmentation fault. My Asterisk Version is 20.4.0 my dial plan:

exten => 550,1,NoOp()
        same => n,Answer()
        same => n,SpeechCreate(my-speech-to-text)
        same => n,NoOp(${ERROR})
        same => n,GotoIf($["${ERROR}" = "1"]?fail:next)
        same => n(next),SpeechStart()
        same => n,SpeechBackground(tt-monkeys)
        same => n,Verbose(0,${SPEECH_TEXT(0)})
        same => n,SpeechDestroy()
        same => n(fail),Hangup()

this is my aeap.conf:

[my-speech-to-text]
type=client
codecs=!all,ulaw
url=ws://127.0.0.1:9099/filestream
protocol=speech_to_text

it works perfectly but after I stop index.js and start it again, this is what I got:

       > 0x7fb57000e350 -- Strict RTP learning after remote address set to: 192.168.97.12:40048
    -- Executing [550@tgui-out:1] NoOp("SIP/121-00000003", "") in new stack
    -- Executing [550@tgui-out:2] Answer("SIP/121-00000003", "") in new stack
       > 0x7fb57000e350 -- Strict RTP switching to RTP target address 192.168.97.12:40048 as source
    -- Executing [550@tgui-out:3] SpeechCreate("SIP/121-00000003", "my-speech-to-text") in new stack
[Nov 11 15:18:09] ERROR[2736897]: res_aeap/transaction.c:148 transaction_end: AEAP transaction (0x7fb57001ec20): message 'setup' timed out
    -- Executing [550@tgui-out:4] NoOp("SIP/121-00000003", "1") in new stack
    -- Executing [550@tgui-out:5] GotoIf("SIP/121-00000003", "1?fail:next") in new stack
    -- Goto (tgui-out,550,10)
    -- Executing [550@tgui-out:10] Hangup("SIP/121-00000003", "") in new stack
  == Spawn extension (tgui-out, 550, 10) exited non-zero on 'SIP/121-00000003'
    -- Executing [h@tgui-out:1] Set("SIP/121-00000003", "COUNT=0") in new stack
    -- Executing [h@tgui-out:2] NoOp("SIP/121-00000003", "Hangup Cause: 16") in new stack
    -- Executing [h@tgui-out:3] NoOp("SIP/121-00000003", "") in new stack
    -- Executing [h@tgui-out:4] NoOp("SIP/121-00000003", "Who Hangup: ") in new stack
    -- Executing [h@tgui-out:5] AGI("SIP/121-00000003", "agi://127.0.0.1/tgui.agi?status=end") in new stack
 agi://127.0.0.1/tgui.agi?status=end: No script configured for URL 'agi://127.0.0.1/tgui.agi?status=end' (script 'tgui.agi')
    -- <SIP/121-00000003>AGI Script agi://127.0.0.1/tgui.agi?status=end completed, returning 0
    -- Executing [h@tgui-out:6] Set("SIP/121-00000003", "COUNT=1") in new stack
    -- Executing [h@tgui-out:7] GotoIf("SIP/121-00000003", "0?retry:checkstate") in new stack
    -- Goto (tgui-out,h,10)
    -- Executing [h@tgui-out:10] GotoIf("SIP/121-00000003", "0?reset:success") in new stack
    -- Goto (tgui-out,h,15)
    -- Executing [h@tgui-out:15] NoOp("SIP/121-00000003", "Agi-Success") in new stack
Segmentation fault (core dumped)

this is the dump info:

Stack trace of thread 2737132:
                #0  0x0000559352315ef4 ast_format_get_codec_name (asterisk + 0xf3ef4)
                #1  0x00007fb559cafc44 handle_response_setup (res_speech_aeap.so + 0x2c44)
                #2  0x00007fb559cbd4c9 aeap_receive (res_aeap.so + 0x84c9)
                #3  0x00005593523e917f dummy_start (asterisk + 0x1c717f)
                #4  0x00007fb57a69f802 start_thread (libc.so.6 + 0x9f802)
                #5  0x00007fb57a63f450 __clone3 (libc.so.6 + 0x3f450)

asterisk / aeap-speech-to-text

Asterisk Crash after script restart #7