Builtin: DTMF and Speech

Ruben Rios

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Mar 14, 2020, 10:52:40 AM3/14/20

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Hello,

I need to use builtin:dtmf/digits and builtin:speech/transcribe grammars both together in one play_and_detect_speech command but it is not working. It is only recognized dtmf not both (dtmf and speech). When I executed each grammars individually, it is working fine. The problem is that I need to recognize dtmf and speech in the same command.

Please can you tell me where is my error in the following code? The command that I executed to recognize dtmf and speech from FS is:

session:execute("play_and_detect_speech","say:{speech-language=es-ES,prosody-rate=medium}Hola en que te puedo ayudar? detect:unimrcp {start-input-timers=false,no-input-timeout=15000,speech-language=es-ES}builtin:dtmf/digits,builtin:speech/transcribe")

I have attached part of the unimrcp server log:

MRCP/2.0 272 RECOGNIZE 1
Channel-Identifier: 8aa18edad94b4b36@speechrecog
Content-Type: text/uri-list
Cancel-If-Queue: false
Start-Input-Timers: false
Speech-Language: es-ES
No-Input-Timeout: 15000
Content-Length: 45

builtin:dtmf/digits,builtin:speech/transcribe
2020-03-14 11:18:59:136230 [INFO] Assign Control Channel <8aa18edad94b4b36@speechrecog> to Connection 192.168.80.133:1544 <-> 192.168.80.133:55562 [0] -> [1]
2020-03-14 11:18:59:136276 [INFO] Process RECOGNIZE Request <8aa18edad94b4b36@speechrecog> [1]
2020-03-14 11:18:59:136378 [INFO] Init DTMF Detector: interdigit-timeout=5000 ms, term-timeout=10000 ms, term-char=#, length=0, min-length=0, max-length=0 <8aa18edad94b4b36>
2020-03-14 11:18:59:136417 [INFO] Create gRPC Stream <8aa18edad94b4b36@gdf>
2020-03-14 11:18:59:137140 [INFO] Process RECOGNIZE Response <8aa18edad94b4b36@speechrecog> [1]
2020-03-14 11:18:59:137155 [INFO] State Transition IDLE -> RECOGNIZING <8aa18edad94b4b36@speechrecog>
2020-03-14 11:18:59:137198 [INFO] Send MRCPv2 Data 192.168.80.133:1544 <-> 192.168.80.133:55562 [83 bytes]

Best Regards,

Ruben

Arsen Chaloyan

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Mar 14, 2020, 3:05:36 PM3/14/20

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Hello Ruben,

Use \n as s separator

session:execute("play_and_detect_speech","say:{speech-language=es-ES,prosody-rate=medium}Hola en que te puedo ayudar? detect:unimrcp {start-input-timers=false,no-input-timeout=15000,speech-language=es-ES}builtin:dtmf/digits\nbuiltin:speech/transcribe")

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Ruben Rios

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Mar 14, 2020, 3:13:10 PM3/14/20

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Hi Arsen,

Thank you very much, with \ n separator it works perfect.

Best regards

Ruben

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sin...@gmail.com

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Jul 16, 2020, 2:42:32 AM7/16/20

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i use unimrcp in astersik ,and got the same problem.. use \ n separator it still didn't work.

i use agi &java to do with asterisk. ..here is the log:( when press number,the voice do not stop. and the dtmf number can not add in recognize_result, too.

[2020-07-16 14:25:15] NOTICE[18870][C-0000040a]: app_synthandrecog.c:1333 app_synthandrecog_exec: SynthAndRecog() prompt: 请问您需要什么帮助？

[2020-07-16 14:25:15] NOTICE[18870][C-0000040a]: app_synthandrecog.c:1341 app_synthandrecog_exec: SynthAndRecog() grammar: builtin:dtmf/digitsbuiltin:speech/transcribe

[2020-07-16 14:25:15] NOTICE[18870][C-0000040a]: app_synthandrecog.c:1372 app_synthandrecog_exec: SynthAndRecog() options: dttc=#&v=siyue&pv=15&pr=-200&t=20000&b=1&ct=0.5&&spl=zh-CN&uer=1

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option dttc: #

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option v: siyue

[2020-07-16 14:25:15] WARNING[18870][C-0000040a]: app_synthandrecog.c:1137 synthandrecog_option_apply: Unknown option: v

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option pv: 15

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option pr: -200

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option t: 20000

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option b: 1

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option ct: 0.5

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option spl: zh-CN

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option uer: 1

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: audio_queue.c:253 audio_queue_create: (ASR-12) Audio queue created

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: speech_channel.c:252 speech_channel_create: Created speech channel: Name=ASR-12, Type=RECOGNIZER, Codec=LPCM, Rate=8000 on SIP/8001-00000809

[2020-07-16 14:25:15] DEBUG[20172]: app_synthandrecog.c:248 speech_on_channel_add: (ASR-12) speech_on_channel_add

[2020-07-16 14:25:15] DEBUG[20172]: app_synthandrecog.c:268 speech_on_channel_add: (ASR-12) DTMF generator created

[2020-07-16 14:25:15] NOTICE[20172]: app_synthandrecog.c:287 speech_on_channel_add: (ASR-12) Channel ready codec=LPCM, sample rate=8000

[2020-07-16 14:25:15] DEBUG[20172]: speech_channel.c:109 speech_channel_set_state_unlocked: (ASR-12) CLOSED ==> READY

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: speech_channel.c:501 speech_channel_open: (ASR-12) channel is ready

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: app_synthandrecog.c:1471 app_synthandrecog_exec: (ASR-12) Determine grammar type: builtin:dtmf/digitsbuiltin:speech/transcribe

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: app_synthandrecog.c:849 recog_channel_load_grammar: (ASR-12) Loading grammar name=grammar-0, type=text/uri-list, data=builtin:dtmf/digitsbuiltin:speech/transcribe

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: app_synthandrecog.c:1502 app_synthandrecog_exec: (ASR-12) Add prompt: 请问您需要什么帮助？

[2020-07-16 14:25:15] NOTICE[18870][C-0000040a]: app_synthandrecog.c:1601 app_synthandrecog_exec: (ASR-12) Recognizing, Start-Input-Timers: 0

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: speech_channel.c:619 speech_channel_set_params: (ASR-12) Confidence-Threshold: 0.5

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: speech_channel.c:619 speech_channel_set_params: (ASR-12) Speech-Language: zh-CN

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: speech_channel.c:619 speech_channel_set_params: (ASR-12) Dtmf-Term-Char: #

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: speech_channel.c:619 speech_channel_set_params: (ASR-12) Recognition-Timeout: 20000

[2020-07-16 14:25:15] DEBUG[20172]: app_synthandrecog.c:928 recog_on_message_receive: (ASR-12) RECOGNIZE IN PROGRESS

[2020-07-16 14:25:15] DEBUG[20172]: speech_channel.c:109 speech_channel_set_state_unlocked: (ASR-12) READY ==> PROCESSING

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: audio_queue.c:253 audio_queue_create: (TTS-12) Audio queue created

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: speech_channel.c:252 speech_channel_create: Created speech channel: Name=TTS-12, Type=SYNTHESIZER, Codec=LPCM, Rate=8000 on SIP/8001-00000809

[2020-07-16 14:25:15] DEBUG[20172]: app_synthandrecog.c:248 speech_on_channel_add: (TTS-12) speech_on_channel_add

[2020-07-16 14:25:15] NOTICE[20172]: app_synthandrecog.c:287 speech_on_channel_add: (TTS-12) Channel ready codec=LPCM, sample rate=8000

[2020-07-16 14:25:15] DEBUG[20172]: speech_channel.c:109 speech_channel_set_state_unlocked: (TTS-12) CLOSED ==> READY

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: speech_channel.c:501 speech_channel_open: (TTS-12) channel is ready

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: speech_channel.c:619 speech_channel_set_params: (TTS-12) Prosody-Volume: 15

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: speech_channel.c:619 speech_channel_set_params: (TTS-12) Prosody-Rate: -200

[2020-07-16 14:25:15] DEBUG[18870][C-0000040a]: speech_channel.c:619 speech_channel_set_params: (TTS-12) Speech-Language: zh-CN

[2020-07-16 14:25:15] DEBUG[20172]: app_synthandrecog.c:329 synth_on_message_receive: (TTS-12) REQUEST IN PROGRESS

[2020-07-16 14:25:15] DEBUG[20172]: speech_channel.c:109 speech_channel_set_state_unlocked: (TTS-12) READY ==> PROCESSING

[2020-07-16 14:25:17] DTMF[18870][C-0000040a]: channel.c:4214 __ast_read: DTMF begin '8' received on SIP/8001-00000809

[2020-07-16 14:25:17] DTMF[18870][C-0000040a]: channel.c:4225 __ast_read: DTMF begin passthrough '8' on SIP/8001-00000809

[2020-07-16 14:25:18] DTMF[18870][C-0000040a]: channel.c:4128 __ast_read: DTMF end '8' received on SIP/8001-00000809, duration 200 ms

[2020-07-16 14:25:18] DTMF[18870][C-0000040a]: channel.c:4169 __ast_read: DTMF end accepted with begin '8' on SIP/8001-00000809

[2020-07-16 14:25:18] DTMF[18870][C-0000040a]: channel.c:4198 __ast_read: DTMF end passthrough '8' on SIP/8001-00000809

[2020-07-16 14:25:18] DEBUG[18870][C-0000040a]: app_synthandrecog.c:1719 app_synthandrecog_exec: (ASR-12) User pressed DTMF key (56)

[2020-07-16 14:25:18] NOTICE[18870][C-0000040a]: app_synthandrecog.c:1726 app_synthandrecog_exec: (ASR-12) DTMF digit queued (8)

Vahagn Kocharyan

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Jul 16, 2020, 6:06:00 AM7/16/20

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Please see this discussion https://groups.google.com/forum/?utm_medium=email&utm_source=footer#!msg/unimrcp/sJjtwSp2X6k/fSMKS7GfAwAJ

To view this discussion on the web visit https://groups.google.com/d/msgid/unimrcp/832f77ca-bdee-4fc0-8e6f-77f746e67247n%40googlegroups.com.

sin...@gmail.com

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Jul 16, 2020, 9:53:33 AM7/16/20

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ok,thx

Vahagn Kocharyan

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Jul 16, 2020, 10:06:37 AM7/16/20

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mrcp conf example

please compare it with your mrcp.conf.

you must have "telephone-event" added to the codec on unimrcp server and in mrcp.conf file

[uni2]
; MRCP settings
version = 2
;
; SIP settings
server-ip = 192.168.88.82
server-port = 1554
; SIP user agent
client-ip = 192.168.88.200
client-port = 25097
sip-transport = udp
;
; RTP factory
rtp-ip = 192.168.88.200
rtp-port-min = 28000
rtp-port-max = 29000
;
; Jitter buffer settings
playout-delay = 50
max-playout-delay = 200
; RTP settings
ptime = 20
codecs = PCMU PCMA L16/96/8000 telephone-event/101/8000

; === RTCP settings ===
rtcp = 1
rtcp-bye = 2
rtcp-tx-interval = 5000
rtcp-rx-resolution = 1000

if it don't work provide logs from asterisk and unimrcp server

thanks

Vahagn

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Vahagn Kocharyan

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Jul 16, 2020, 10:11:51 AM7/16/20

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fo asterisk you must set separator like ","

example

builtin:speech/transcribe?projectid="your projectId",builtin:dtmf/digits?

sin...@gmail.com

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Jul 16, 2020, 11:21:59 PM7/16/20

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thx.kochary... I changed as your said but still can not disturb voice and catch the dtmf :

this is the log:

[2020-07-17 11:19:21] NOTICE[17683][C-00000a61]: app_synthandrecog.c:1333 app_synthandrecog_exec: SynthAndRecog() prompt: 您好，请用普通话说出您要找的人的姓名

[2020-07-17 11:19:21] NOTICE[17683][C-00000a61]: app_synthandrecog.c:1341 app_synthandrecog_exec: SynthAndRecog() grammar: builtin:speech/transcribe?projectid=pro001,builtin:dtmf/digits?

[2020-07-17 11:19:21] NOTICE[17683][C-00000a61]: app_synthandrecog.c:1372 app_synthandrecog_exec: SynthAndRecog() options: t=5000&b=1&ct=0.2&sl=0.8&spl=zh-CN&rm=hotword

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option t: 5000

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option b: 1

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option ct: 0.2

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option sl: 0.8

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option spl: zh-CN

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1161 synthandrecog_options_parse: Apply option rm: hotword

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: audio_queue.c:253 audio_queue_create: (ASR-16) Audio queue created

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: speech_channel.c:252 speech_channel_create: Created speech channel: Name=ASR-16, Type=RECOGNIZER, Codec=LPCM, Rate=8000 on SIP/8001-000014c1

[2020-07-17 11:19:21] DEBUG[10788]: app_synthandrecog.c:248 speech_on_channel_add: (ASR-16) speech_on_channel_add

[2020-07-17 11:19:21] DEBUG[10788]: app_synthandrecog.c:268 speech_on_channel_add: (ASR-16) DTMF generator created

[2020-07-17 11:19:21] NOTICE[10788]: app_synthandrecog.c:287 speech_on_channel_add: (ASR-16) Channel ready codec=LPCM, sample rate=8000

[2020-07-17 11:19:21] DEBUG[10788]: speech_channel.c:109 speech_channel_set_state_unlocked: (ASR-16) CLOSED ==> READY

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: speech_channel.c:501 speech_channel_open: (ASR-16) channel is ready

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1471 app_synthandrecog_exec: (ASR-16) Determine grammar type: builtin:speech/transcribe?projectid=pro001

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: app_synthandrecog.c:849 recog_channel_load_grammar: (ASR-16) Loading grammar name=grammar-0, type=text/uri-list, data=builtin:speech/transcribe?projectid=pro001

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1471 app_synthandrecog_exec: (ASR-16) Determine grammar type: builtin:dtmf/digits?

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: app_synthandrecog.c:849 recog_channel_load_grammar: (ASR-16) Loading grammar name=grammar-1, type=text/uri-list, data=builtin:dtmf/digits?

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1502 app_synthandrecog_exec: (ASR-16) Add prompt: 您好，请用普通话说出您要找的人的姓名

[2020-07-17 11:19:21] NOTICE[17683][C-00000a61]: app_synthandrecog.c:1601 app_synthandrecog_exec: (ASR-16) Recognizing, Start-Input-Timers: 0

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: speech_channel.c:619 speech_channel_set_params: (ASR-16) Confidence-Threshold: 0.2

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: speech_channel.c:619 speech_channel_set_params: (ASR-16) Speech-Language: zh-CN

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: speech_channel.c:619 speech_channel_set_params: (ASR-16) Sensitivity-Level: 0.8

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: speech_channel.c:619 speech_channel_set_params: (ASR-16) Recognition-Mode: hotword

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: speech_channel.c:619 speech_channel_set_params: (ASR-16) Recognition-Timeout: 5000

[2020-07-17 11:19:21] DEBUG[10788]: app_synthandrecog.c:928 recog_on_message_receive: (ASR-16) RECOGNIZE IN PROGRESS

[2020-07-17 11:19:21] DEBUG[10788]: speech_channel.c:109 speech_channel_set_state_unlocked: (ASR-16) READY ==> PROCESSING

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: audio_queue.c:253 audio_queue_create: (TTS-16) Audio queue created

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: speech_channel.c:252 speech_channel_create: Created speech channel: Name=TTS-16, Type=SYNTHESIZER, Codec=LPCM, Rate=8000 on SIP/8001-000014c1

[2020-07-17 11:19:21] DEBUG[10788]: app_synthandrecog.c:248 speech_on_channel_add: (TTS-16) speech_on_channel_add

[2020-07-17 11:19:21] NOTICE[10788]: app_synthandrecog.c:287 speech_on_channel_add: (TTS-16) Channel ready codec=LPCM, sample rate=8000

[2020-07-17 11:19:21] DEBUG[10788]: speech_channel.c:109 speech_channel_set_state_unlocked: (TTS-16) CLOSED ==> READY

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: speech_channel.c:501 speech_channel_open: (TTS-16) channel is ready

[2020-07-17 11:19:21] DEBUG[17683][C-00000a61]: speech_channel.c:619 speech_channel_set_params: (TTS-16) Speech-Language: zh-CN

[2020-07-17 11:19:21] DEBUG[10788]: app_synthandrecog.c:329 synth_on_message_receive: (TTS-16) REQUEST IN PROGRESS

[2020-07-17 11:19:21] DEBUG[10788]: speech_channel.c:109 speech_channel_set_state_unlocked: (TTS-16) READY ==> PROCESSING

[2020-07-17 11:19:22] DTMF[17683][C-00000a61]: channel.c:4214 __ast_read: DTMF begin '2' received on SIP/8001-000014c1

[2020-07-17 11:19:22] DTMF[17683][C-00000a61]: channel.c:4225 __ast_read: DTMF begin passthrough '2' on SIP/8001-000014c1

[2020-07-17 11:19:22] DTMF[17683][C-00000a61]: channel.c:4128 __ast_read: DTMF end '2' received on SIP/8001-000014c1, duration 200 ms

[2020-07-17 11:19:22] DTMF[17683][C-00000a61]: channel.c:4169 __ast_read: DTMF end accepted with begin '2' on SIP/8001-000014c1

[2020-07-17 11:19:22] DTMF[17683][C-00000a61]: channel.c:4198 __ast_read: DTMF end passthrough '2' on SIP/8001-000014c1

[2020-07-17 11:19:22] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1719 app_synthandrecog_exec: (ASR-16) User pressed DTMF key (50)

[2020-07-17 11:19:22] NOTICE[17683][C-00000a61]: app_synthandrecog.c:1726 app_synthandrecog_exec: (ASR-16) DTMF digit queued (2)

[2020-07-17 11:19:22] DEBUG[10790]: app_synthandrecog.c:1038 recog_stream_read: (ASR-16) DTMF frame written

[2020-07-17 11:19:23] DTMF[17683][C-00000a61]: channel.c:4214 __ast_read: DTMF begin '5' received on SIP/8001-000014c1

[2020-07-17 11:19:23] DTMF[17683][C-00000a61]: channel.c:4225 __ast_read: DTMF begin passthrough '5' on SIP/8001-000014c1

[2020-07-17 11:19:23] DTMF[17683][C-00000a61]: channel.c:4128 __ast_read: DTMF end '5' received on SIP/8001-000014c1, duration 200 ms

[2020-07-17 11:19:23] DTMF[17683][C-00000a61]: channel.c:4169 __ast_read: DTMF end accepted with begin '5' on SIP/8001-000014c1

[2020-07-17 11:19:23] DTMF[17683][C-00000a61]: channel.c:4198 __ast_read: DTMF end passthrough '5' on SIP/8001-000014c1

[2020-07-17 11:19:23] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1719 app_synthandrecog_exec: (ASR-16) User pressed DTMF key (53)

[2020-07-17 11:19:23] NOTICE[17683][C-00000a61]: app_synthandrecog.c:1726 app_synthandrecog_exec: (ASR-16) DTMF digit queued (5)

[2020-07-17 11:19:23] DEBUG[10790]: app_synthandrecog.c:1038 recog_stream_read: (ASR-16) DTMF frame written

[2020-07-17 11:19:24] DTMF[17683][C-00000a61]: channel.c:4214 __ast_read: DTMF begin '2' received on SIP/8001-000014c1

[2020-07-17 11:19:24] DTMF[17683][C-00000a61]: channel.c:4225 __ast_read: DTMF begin passthrough '2' on SIP/8001-000014c1

[2020-07-17 11:19:24] DTMF[17683][C-00000a61]: channel.c:4128 __ast_read: DTMF end '2' received on SIP/8001-000014c1, duration 200 ms

[2020-07-17 11:19:24] DTMF[17683][C-00000a61]: channel.c:4169 __ast_read: DTMF end accepted with begin '2' on SIP/8001-000014c1

[2020-07-17 11:19:24] DTMF[17683][C-00000a61]: channel.c:4198 __ast_read: DTMF end passthrough '2' on SIP/8001-000014c1

[2020-07-17 11:19:24] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1719 app_synthandrecog_exec: (ASR-16) User pressed DTMF key (50)

[2020-07-17 11:19:24] NOTICE[17683][C-00000a61]: app_synthandrecog.c:1726 app_synthandrecog_exec: (ASR-16) DTMF digit queued (2)

[2020-07-17 11:19:24] DEBUG[10790]: app_synthandrecog.c:1038 recog_stream_read: (ASR-16) DTMF frame written

[2020-07-17 11:19:25] DEBUG[10788]: app_synthandrecog.c:367 synth_on_message_receive: (TTS-16) SPEAK-COMPLETE

[2020-07-17 11:19:25] DEBUG[10788]: speech_channel.c:109 speech_channel_set_state_unlocked: (TTS-16) PROCESSING ==> READY

[2020-07-17 11:19:25] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1679 app_synthandrecog_exec: (ASR-16) Start input timers

[2020-07-17 11:19:25] DEBUG[17683][C-00000a61]: app_synthandrecog.c:536 recog_channel_start_input_timers: (ASR-16) Sending START-INPUT-TIMERS request

[2020-07-17 11:19:25] DEBUG[10788]: app_synthandrecog.c:960 recog_on_message_receive: (ASR-16) Timers started

[2020-07-17 11:19:25] DEBUG[17683][C-00000a61]: app_synthandrecog.c:1704 app_synthandrecog_exec: (ASR-16) Null frame. Hangup detected

[2020-07-17 11:19:25] DEBUG[17683][C-00000a61]: speech_channel.c:340 speech_channel_destroy: Destroy speech channel: Name=TTS-16, Type=SYNTHESIZER, Codec=LPCM, Rate=8000

[2020-07-17 11:19:25] DEBUG[17683][C-00000a61]: speech_channel.c:365 speech_channel_destroy: (TTS-16) Waiting for MRCP session to terminate

[2020-07-17 11:19:25] DEBUG[10788]: app_synthandrecog.c:222 speech_on_session_terminate: (TTS-16) speech_on_session_terminate

[2020-07-17 11:19:25] DEBUG[10788]: app_synthandrecog.c:230 speech_on_session_terminate: (TTS-16) Destroying MRCP session

[2020-07-17 11:19:25] DEBUG[10788]: speech_channel.c:109 speech_channel_set_state_unlocked: (TTS-16) READY ==> CLOSED

[2020-07-17 11:19:25] DEBUG[17683][C-00000a61]: audio_queue.c:197 audio_queue_destroy: (TTS-16) Audio queue destroyed

[2020-07-17 11:19:25] DEBUG[17683][C-00000a61]: speech_channel.c:340 speech_channel_destroy: Destroy speech channel: Name=ASR-16, Type=RECOGNIZER, Codec=LPCM, Rate=8000

[2020-07-17 11:19:25] DEBUG[17683][C-00000a61]: speech_channel.c:365 speech_channel_destroy: (ASR-16) Waiting for MRCP session to terminate

[2020-07-17 11:19:26] DEBUG[10788]: app_synthandrecog.c:222 speech_on_session_terminate: (ASR-16) speech_on_session_terminate

[2020-07-17 11:19:26] DEBUG[10788]: app_synthandrecog.c:225 speech_on_session_terminate: (ASR-16) DTMF generator destroyed

[2020-07-17 11:19:26] DEBUG[10788]: app_synthandrecog.c:230 speech_on_session_terminate: (ASR-16) Destroying MRCP session

[2020-07-17 11:19:26] DEBUG[10788]: speech_channel.c:109 speech_channel_set_state_unlocked: (ASR-16) PROCESSING ==> CLOSED

[2020-07-17 11:19:26] DEBUG[17683][C-00000a61]: audio_queue.c:197 audio_queue_destroy: (ASR-16) Audio queue destroyed

[2020-07-17 11:19:26] NOTICE[17683][C-00000a61]: app_synthandrecog.c:1292 synthandrecog_exit: SynthAndRecog() exiting status: INTERRUPTED on SIP/8001-000014c1

[2020-07-17 11:19:28] WARNING[17673][C-00000a60]: app_dial.c:1546 wait_for_answer: Unable to write frametype: 2

Vahagn Kocharyan

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Jul 17, 2020, 1:29:39 PM7/17/20

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you must have project id for both grammars.

you have "codecs = PCMU PCMA L16/96/8000 telephone-event/101/8000 in mrcp.conf" and

"<codecs own-preference="false">PCMU PCMA L16/96/8000 telephone-event/101/8000</codecs> " on unimrcpserver.conf ?

To view this discussion on the web visit https://groups.google.com/d/msgid/unimrcp/f545be27-59c6-41ec-bb3e-f94cb16ffdb2n%40googlegroups.com.

sin...@gmail.com

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Jul 18, 2020, 11:44:59 AM7/18/20

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yes, i check it twice.

here is the conf file:

mrcp.conf

[general]

default-asr-profile = speech-nuance5-mrcp2

default-tts-profile = speech-nuance5-mrcp2

log-level = WARNING

max-connection-count = 100

offer-new-connection = 1

[speech-nuance5-mrcp2]

version = 2

server-ip = 127.0.0.1

server-port = 8060

force-destination = 1

client-ip = 127.0.0.1

client-port = 5093

sip-transport = udp

rtp-ip = 127.0.0.1

rtp-port-min = 4000

rtp-port-max = 5000

playout-delay = 50

max-playout-delay = 200

ptime = 20

codecs = PCMU PCMA L16/96/8000 telephone-event/101/8000

rtcp = 1

rtcp-bye = 2

rtcp-tx-interval = 5000

rtcp-rx-resolution = 1000

unimrcpserver.xml

<?xml version="1.0" encoding="UTF-8"?>

<unimrcpserver xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"

xsi:noNamespaceSchemaLocation="unimrcpserver.xsd"

version="1.0">

</properties>

<resource-factory>

</resource-factory>

<sip-uas id="SIP-Agent-1" type="SofiaSIP">

<sip-port>8060</sip-port>

<sip-transport>udp,tcp</sip-transport>

<ua-name>BaiduSpeech SofiaSIP</ua-name>

<sdp-origin>BaiduSpeechServer</sdp-origin>

<sip-session-expires>600</sip-session-expires>

<sip-min-session-expires>120</sip-min-session-expires>

</sip-uas>

<rtsp-uas id="RTSP-Agent-1" type="UniRTSP">

<rtsp-port>1554</rtsp-port>

<resource-map>

</resource-map>

<max-connection-count>100</max-connection-count>

<inactivity-timeout>600</inactivity-timeout>

<sdp-origin>UniMRCPServer</sdp-origin>

</rtsp-uas>

<mrcpv2-uas id="MRCPv2-Agent-1">

<mrcp-port>1544</mrcp-port>

<max-connection-count>400</max-connection-count>

<max-shared-use-count>400</max-shared-use-count>

<force-new-connection>true</force-new-connection>

<rx-buffer-size>1024</rx-buffer-size>

<tx-buffer-size>1024</tx-buffer-size>

<inactivity-timeout>600</inactivity-timeout>

<termination-timeout>3</termination-timeout>

</mrcpv2-uas>

<media-engine id="Media-Engine-1">

<realtime-rate>1</realtime-rate>

</media-engine>

<rtp-factory id="RTP-Factory-1">

<rtp-port-min>5000</rtp-port-min>

<rtp-port-max>6000</rtp-port-max>

</rtp-factory>

<plugin-factory>

</plugin-factory>

</components>

<rtp-settings id="RTP-Settings-1">

<jitter-buffer>

<playout-delay>50</playout-delay>

<max-playout-delay>600</max-playout-delay>

<time-skew-detection>1</time-skew-detection>

</jitter-buffer>

<codecs own-preference="false">PCMU PCMA L16/96/8000 telephone-event/101/8000</codecs>

<rtcp-bye>1</rtcp-bye>

<tx-interval>5000</tx-interval>

<rx-resolution>1000</rx-resolution>

</rtcp>

</rtp-settings>

</settings>

<mrcpv2-profile id="uni2">

<sip-uas>SIP-Agent-1</sip-uas>

<mrcpv2-uas>MRCPv2-Agent-1</mrcpv2-uas>

<media-engine>Media-Engine-1</media-engine>

<rtp-factory>RTP-Factory-1</rtp-factory>

<rtp-settings>RTP-Settings-1</rtp-settings>

</mrcpv2-profile>

<mrcpv1-profile id="uni1">

<rtsp-uas>RTSP-Agent-1</rtsp-uas>

<media-engine>Media-Engine-1</media-engine>

<rtp-factory>RTP-Factory-1</rtp-factory>

<rtp-settings>RTP-Settings-1</rtp-settings>

</mrcpv1-profile>

</profiles>

</unimrcpserver>

sin...@gmail.com

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Jul 20, 2020, 1:29:54 AM7/20/20

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Is there anything wrong?

Vahagn Kocharyan

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Jul 20, 2020, 6:00:09 AM7/20/20

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can you share your unimrcp server logs

Thanks

Vahagn

To view this discussion on the web visit https://groups.google.com/d/msgid/unimrcp/6fd11ad1-bd4a-4f21-a498-0e8c9ccafea2n%40googlegroups.com.

Wilmar Pérez

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Jul 20, 2020, 8:38:20 AM7/20/20

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Hola Ruben,

Have a look at this post: https://groups.google.com/g/unimrcp/c/sJjtwSp2X6k/m/6uLXF_8fCAAJ

I have the same scenario you have. The key was to modify builtin:dtmf/digits,builtin:speech/transcribe to builtin:dtmf/digits\nbuiltin:speech/transcribe. That is, change the , for \n to separate the grammar. You can shoot me a direct email in Spanish if you want to, I would certainly love to have contact with somebody that is going through the same sort of issues I have been dealing with with freespeech.

Best,

Wilmar

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