unispeech / asterisk-unimrcp

UniMRCP modules for Asterisk
http://www.unimrcp.org/asterisk
GNU General Public License v2.0
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(ASR-0) Can't find profile, default #50

Closed MrityunjoyS closed 3 years ago

MrityunjoyS commented 3 years ago

I've configured asterisk, unimrcp and zoiper. My mrcp.conf file -->

[general] ; Default ASR and TTS profiles. default-asr-profile = speech-nuance5-mrcp2 default-tts-profile = speech-nuance5-mrcp2 ; UniMRCP logging level to appear in Asterisk logs. Options are: ; EMERGENCY|ALERT|CRITICAL|ERROR|WARNING|NOTICE|INFO|DEBUG --> log-level = DEBUG max-connection-count = 100 offer-new-connection = 1 ; rx-buffer-size = 1024 ; tx-buffer-size = 1024 ; request-timeout = 5000 ; speech-channel-timeout = 30000 ; ; Profile for Nuance Speech Server MRCPv1 ; [unimrcpv1] ; MRCP settings version = 1 ; ; RTSP settings server-ip = 192.168.1.9 server-port = 1554 resource-location = media speechsynth = speechsynthesizer ; RTP factory rtp-ip = 192.168.1.9 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 = 0 ; ; Profile for Nuance Speech Server MRCPv2 ; [unimrcpv2] ; MRCP settings version = 2 ; ; SIP settings server-ip = 192.168.1.9 server-port = 8004 ; SIP user agent client-ip = 192.168.1.9 client-port = 60820 ;client-port = 25097 sip-transport = udp ; ; RTP factory rtp-ip = 192.168.1.9 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 = 0

My extensions.conf file -->

[from-internal] exten => 6113,1,Answer() exten => 6113,n,MRCPRecog(builtin:grammar/number,p=default&t=5000&b=1&ct=0.7&spl=en-US&f=hello-world) exten => 6113,n,Verbose(1, ${RECOGSTATUS}, ${RECOG_COMPLETION_CAUSE}, ${RECOG_RESULT}) exten => 6113,n,Hangup

Then when I'm making the call using zoiper getting below error -->

== Setting global variable 'SIPDOMAIN' to '192.168.1.9' -- Executing [6113@from-internal:1] Answer("PJSIP/6001-00000007", "") in new stack

0x7f2fb8033c30 -- Strict RTP learning after remote address set to: 192.168.1.9:8000 0x7f2fb8033c30 -- Strict RTP switching to RTP target address 192.168.1.9:8000 as source -- Executing [6113@from-internal:2] MRCPRecog("PJSIP/6001-00000007", "builtin:grammar/number,p=default&t=5000&b=1&ct=0.7&spl=en-US&f=hello-world") in new stack [Sep 8 21:05:59] NOTICE[28712][C-00000008]: app_mrcprecog.c:1086 app_recog_exec: MRCPRecog() grammar: builtin:grammar/number [Sep 8 21:05:59] NOTICE[28712][C-00000008]: app_mrcprecog.c:1107 app_recogexec: MRCPRecog() options: p=default&t=5000&b=1&ct=0.7&spl=en-US&f=hello-world **> [Sep 8 21:05:59] ERROR[28712][C-00000008]: app_mrcprecog.c:1166 app_recogexec: (ASR-6) Can't find profile, default [Sep 8 21:05:59] NOTICE[28712][C-00000008]: app_mrcprecog.c:1047 mrcprecog_exit: MRCPRecog() exiting status: ERROR on PJSIP/6001-00000007** == Spawn extension (from-internal, 6113, 2) exited non-zero on 'PJSIP/6001-00000007'

Please help me with this one. Thank you in advance