WARNING: pbx.c:1344 pbx_exec: The application delimiter is now the comma,
not the pipe. Did you forget to convert your dialplan? (AGI(say.pl |'Welcome
to the Sphinx and Asterisk integration test.'))
WARNING: res_agi.c:885 launch_script: Failed to execute '/var/lib/asterisk
/agi-bin/say.pl |'Welcome to the Sphinx and Asterisk integration test.'': File
does not exist.
-- Executing SpeechActivateGrammar("SIP/6003-00000001", "ar.DMP") in new stack
-- Executing AGI("SIP/6003-00000001", "say.pl|'Please say yes or
no.'|1|yornprompt") in new stack
WARNING: pbx.c:1344 pbx_exec: The application delimiter is now the comma, not
the pipe. Did you forget to convert your dialplan? (AGI(say.pl|'Please say yes
or no.'|1|yornprompt))
WARNING: res_agi.c:885 launch_script: Failed to execute '/var/lib/asterisk
/agi-bin/say.pl|'Please say yes or no.'|1|yornprompt': File does not exist.
-- Executing SpeechStart("SIP/6003-00000001", "") in new stack
-- Executing SpeechBackground("SIP/6003-00000001", "/tmp/yornprompt|10") in
new stack
WARNING: pbx.c:1344 pbx_exec: The application delimiter is now the comma, not
the pipe. Did you forget to convert your dialplan?
(SpeechBackground(/tmp/yornprompt|10))
WARNING: file.c:650 ast_openstream_full: File /tmp/yornprompt|10 does not
exist in any format
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
NOTICE: res_speech_sphinx.c:302 sphinx_sread: Score: 0 Result: 'لَّا'
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
NOTICE: res_speech_sphinx.c:302 sphinx_sread: Score: 0 Result: 'لَّا'
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
NOTICE: res_speech_sphinx.c:302 sphinx_sread: Score: 0 Result: 'لَّا'
WARNING: dsp.c:1282 ast_dsp_silence: Can't calculate silence on a non-voice
frame
NOTICE: res_speech_sphinx.c:302 sphinx_sread: Score: 0 Result: 'لَّا'
"Can't calculate silence on a non-voice frame" when search on this word its
appear its a frame time where is the configuration to this time
It gives you warnings in the log which you need to understand and fix
[Apr 4 00:30:53] WARNING[31982]: pbx.c:1344 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (AGI(say.pl|'Please say yes or no.'|1|yornprompt)) [Apr 4 00:30:53] WARNING[31982]: res_agi.c:885 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/say.pl|'Please say yes or no.'|1|yornprompt': File does not exist.
If you would like to refer to this comment somewhere else in this project, copy and paste the following link:
exten => 5554,1,Answer()
exten => 5554,2,SpeechCreate()
exten => 5554,3,SpeechLoadGrammar(digit,/usr/local/unimrcp/data/gramma$
exten => 5554,4,SpeechActivateGrammar(digit)
exten => 5554,5,SpeechBackground(hello-world,20)
exten => 5554,6,GotoIf($?7:9)
exten => 5554,7,Playback(vm-nonumber)
exten => 5554,8,Goto(5)
exten => 5554,9,Verbose(1,The recognized input is ${SPEECH_TEXT(0)})
exten => 5554,10,Verbose(1,The score is ${SPEECH_SCORE(0)})
exten => 5554,11,Verbose(1,The matched grammar is ${SPEECH_GRAMMAR(0)})
exten => 5554,12,SpeechDeactivateGrammar(digit)
exten => 5554,13,SpeechUnloadGrammar(digit)
exten => 5554,14,SpeechDestroy()
exten => 5554,15,Hangup()
when i call from sip to 5554
i got this error
== Using SIP RTP CoS mark 5
-- Executing Answer("SIP/6006-00000000", "") in new stack
-- Executing SpeechCreate("SIP/6006-00000000", "") in new stack
NOTICE: app_unimrcp.c:4169 unimrcp_log: Create MRCP Handle 0x8770988
NOTICE: app_unimrcp.c:4169 unimrcp_log: Add Control Channel 0x8770988 new@speechrecog
ip-118-139-163-102*CLI>
all xml can be found her
any help
If you would like to refer to this comment somewhere else in this project, copy and paste the following link:
glibc detected /usr/sbin/asterisk: double free or corruption (out):
0x000000001fddd2a0
This message means there is a bug in the application. It might be a bug in
asterisk or in other modules. You need to collect stacktrace to provide more
information for developer. Without stacktrace it's hard to resolve this issue.
You need to contact developers about this issue, it's not CMUSphinx problem.
Hopefully you will not report any BSOD you meet here.
If you would like to refer to this comment somewhere else in this project, copy and paste the following link:
hello everyone , i'm beginner in asterisk and speech recognition and i want to make a project in university about IVR with speech recognition , well i learned a lot of doumentation and i have installed asterisk but i have a problem while i'm installing asterisk-unimrcp , i have asterisk 14.2.1 and unimrcp 1.4.0
and this is the error :
root@moemen-HP-15-Notebook-PC:/home/moemen/Téléchargements/asterisk-unimrcp-1.3.0# make
Making all in res-speech-unimrcp
make[1] : on entre dans le répertoire « /home/moemen/Téléchargements/asterisk-unimrcp-1.3.0/res-speech-unimrcp »
CC res_speech_unimrcp.lo
In file included from ../include/ast_compat_defs.h:30:0,
from res_speech_unimrcp.c:28:
/usr/include/asterisk.h:300:2: error: #error "Externally compiled modules must declare AST_MODULE_SELF_SYM."
#error "Externally compiled modules must declare AST_MODULE_SELF_SYM."
^
res_speech_unimrcp.c:31:23: error: expected declaration specifiers or ‘...’ before string constant
ASTERISK_FILE_VERSION(FILE, "$Revision: $")
^
res_speech_unimrcp.c:31:33: error: expected declaration specifiers or ‘...’ before string constant
ASTERISK_FILE_VERSION(FILE, "$Revision: $")
^
Makefile:444 : la recette pour la cible « res_speech_unimrcp.lo » a échouée
make[1]: [res_speech_unimrcp.lo] Erreur 1
make[1] : on quitte le répertoire « /home/moemen/Téléchargements/asterisk-unimrcp-1.3.0/res-speech-unimrcp »
Makefile:422 : la recette pour la cible « all-recursive » a échouée
make: [all-recursive] Erreur 1
If you would like to refer to this comment somewhere else in this project, copy and paste the following link:
Those are pretty self-explaining messages not really related to pocketsphinx.
The module is not compatible with asterisk version. You need to try older asterisk version or you can fix the module yourself. Usually it is not hard to add missing defines.
If you would like to refer to this comment somewhere else in this project, copy and paste the following link:
we integrate asterisk with pocket sphinx
when tested using using xlite soft-phone http://download.cnet.com/X-Lite/300
0-2349_4-10547103.html
it give excellent result
but when i tried to connect via my Samsung galaxy s1 using
sipdroid https://play.google.com/store/apps/details?id=org.sipdroid.sipua
i got this error
full log in asterisk :
this is Arabic word 'لَّا'
and the full log in astsphinx
Current configuration:
-agc none none
-agcthresh 2.0 2.000000e+00
-alpha 0.97 9.700000e-01
-ascale 20.0 2.000000e+01
-aw 1 1
-backtrace no no
-beam 1e-48 1.000000e-48
-bestpath yes yes
-bestpathlw 9.5 9.500000e+00
-bghist no no
-ceplen 13 13
-cmn current current
-cmninit 8.0 8.0
-compallsen no no
-debug 0
-dict dict
-dictcase no no
-dither no no
-doublebw no no
-ds 1 1
-fdict
-feat 1s_c_d_dd 1s_c_d_dd
-featparams
-fillprob 1e-8 1.000000e-08
-frate 100 50
-fsg
-fsgusealtpron yes yes
-fsgusefiller yes yes
-fwdflat yes yes
-fwdflatbeam 1e-64 1.000000e-64
-fwdflatefwid 4 4
-fwdflatlw 8.5 8.500000e+00
-fwdflatsfwin 25 25
-fwdflatwbeam 7e-29 7.000000e-29
-fwdtree yes yes
-hmm /home/hiyassat/tutorial/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k/
-input_endian little little
-jsgf
-kdmaxbbi -1 -1
-kdmaxdepth 0 0
-kdtree
-latsize 5000 5000
-lda
-ldadim 0 0
-lextreedump 0 0
-lifter 0 0
-lm yesno
-lmctl
-lmname default default
-logbase 1.0001 1.000100e+00
-logfn
-logspec no no
-lowerf 133.33334 1.333333e+02
-lpbeam 1e-40 1.000000e-40
-lponlybeam 7e-29 7.000000e-29
-lw 6.5 6.500000e+00
-maxhmmpf -1 -1
-maxnewoov 20 20
-maxwpf -1 -1
-mdef
-mean
-mfclogdir
-min_endfr 0 0
-mixw
-mixwfloor 0.0000001 1.000000e-07
-mllr
-mmap yes yes
-ncep 13 13
-nfft 512 512
-nfilt 40 40
-nwpen 1.0 1.000000e+00
-pbeam 1e-48 1.000000e-48
-pip 1.0 1.000000e+00
-pl_beam 1e-10 1.000000e-10
-pl_pbeam 1e-5 1.000000e-05
-pl_window 0 0
-rawlogdir
-remove_dc no no
-round_filters yes yes
-samprate 16000 8.000000e+03
-seed -1 -1
-sendump
-senlogdir
-senmgau
-silprob 0.005 5.000000e-03
-smoothspec no no
-svspec
-tmat
-tmatfloor 0.0001 1.000000e-04
-topn 4 4
-topn_beam 0 0
-toprule
-transform legacy legacy
-unit_area yes yes
-upperf 6855.4976 6.855498e+03
-usewdphones no no
-uw 1.0 1.000000e+00
-var
-varfloor 0.0001 1.000000e-04
-varnorm no no
-verbose no no
-warp_params
-warp_type inverse_linear inverse_linear
-wbeam 7e-29 7.000000e-29
-wip 0.65 6.500000e-01
-wlen 0.025625 2.562500e-02
INFO: cmd_ln.c(691): Parsing command line:
\
-nfilt 31 \
-lowerf 200 \
-upperf 3500 \
-wlen 0.0256 \
-feat s2_4x \
-agc none \
-cmn current \
-varnorm no \
-alpha 0.97 \
-doublebw no \
-transform legacy \
-ncep 13 \
-nfft 512
Current configuration:
-agc none none
-agcthresh 2.0 2.000000e+00
-alpha 0.97 9.700000e-01
-ceplen 13 13
-cmn current current
-cmninit 8.0 8.0
-dither no no
-doublebw no no
-feat 1s_c_d_dd s2_4x
-frate 100 50
-input_endian little little
-lda
-ldadim 0 0
-lifter 0 0
-logspec no no
-lowerf 133.33334 2.000000e+02
-ncep 13 13
-nfft 512 512
-nfilt 40 31
-remove_dc no no
-round_filters yes yes
-samprate 16000 8.000000e+03
-seed -1 -1
-smoothspec no no
-svspec
-transform legacy legacy
-unit_area yes yes
-upperf 6855.4976 3.500000e+03
-varnorm no no
-verbose no no
-warp_params
-warp_type inverse_linear inverse_linear
-wlen 0.025625 2.560000e-02
INFO: acmod.c(242): Parsed model-specific feature parameters from /home/hiyass
at/tutorial/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k//feat.params
INFO: feat.c(684): Initializing feature stream to type: 's2_4x', ceplen=13,
CMN='current', VARNORM='no', AGC='none'
INFO: cmn.c(142): mean= 12.00, mean= 0.0
INFO: mdef.c(520): Reading model definition:
/home/hiyassat/tutorial/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k//mdef
INFO: bin_mdef.c(173): Allocating 17583 * 8 bytes (137 KiB) for CD tree
INFO: tmat.c(205): Reading HMM transition probability matrices: /home/hiyassat
/tutorial/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k//transition_matrices
INFO: acmod.c(117): Attempting to use SCHMM computation module
INFO: ms_gauden.c(198): Reading mixture gaussian parameter:
/home/hiyassat/tutorial/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k//means
INFO: ms_gauden.c(292): 1 codebook, 4 feature, size:
INFO: ms_gauden.c(294): 256x12
INFO: ms_gauden.c(294): 256x24
INFO: ms_gauden.c(294): 256x3
INFO: ms_gauden.c(294): 256x12
INFO: ms_gauden.c(198): Reading mixture gaussian parameter:
/home/hiyassat/tutorial/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k//variances
INFO: ms_gauden.c(292): 1 codebook, 4 feature, size:
INFO: ms_gauden.c(294): 256x12
INFO: ms_gauden.c(294): 256x24
INFO: ms_gauden.c(294): 256x3
INFO: ms_gauden.c(294): 256x12
INFO: ms_gauden.c(354): 144 variance values floored
INFO: s2_semi_mgau.c(908): Loading senones from dump file
/home/hiyassat/tutorial/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k//sendump
INFO: s2_semi_mgau.c(932): BEGIN FILE FORMAT DESCRIPTION
INFO: s2_semi_mgau.c(995): Rows: 256, Columns: 1150
INFO: s2_semi_mgau.c(1027): Using memory-mapped I/O for senones
INFO: s2_semi_mgau.c(1304): Maximum top-N: 4 Top-N beams: 0 0 0 0
INFO: dict.c(306): Allocating 6003 * 20 bytes (117 KiB) for word entries
INFO: dict.c(321): Reading main dictionary: dict
INFO: dict.c(212): Allocated 34 KiB for strings, 29 KiB for phones
INFO: dict.c(324): 1904 words read
INFO: dict.c(330): Reading filler dictionary:
/home/hiyassat/tutorial/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k//noisedict
INFO: dict.c(212): Allocated 0 KiB for strings, 0 KiB for phones
INFO: dict.c(333): 3 words read
INFO: dict2pid.c(396): Building PID tables for dictionary
INFO: dict2pid.c(404): Allocating 50^3 * 2 bytes (244 KiB) for word-initial
triphones
INFO: dict2pid.c(131): Allocated 30200 bytes (29 KiB) for word-final triphones
INFO: dict2pid.c(195): Allocated 30200 bytes (29 KiB) for single-phone word
triphones
INFO: ngram_model_arpa.c(477): ngrams 1=1430, 2=2411, 3=3011
INFO: ngram_model_arpa.c(135): Reading unigrams
INFO: ngram_model_arpa.c(516): 1430 = #unigrams created
INFO: ngram_model_arpa.c(195): Reading bigrams
INFO: ngram_model_arpa.c(533): 2411 = #bigrams created
INFO: ngram_model_arpa.c(534): 80 = #prob2 entries
INFO: ngram_model_arpa.c(542): 122 = #bo_wt2 entries
INFO: ngram_model_arpa.c(292): Reading trigrams
INFO: ngram_model_arpa.c(555): 3011 = #trigrams created
INFO: ngram_model_arpa.c(556): 55 = #prob3 entries
INFO: ngram_search_fwdtree.c(99): 150 unique initial diphones
INFO: ngram_search_fwdtree.c(147): 0 root, 0 non-root channels, 5 single-phone
words
INFO: ngram_search_fwdtree.c(186): Creating search tree
INFO: ngram_search_fwdtree.c(191): before: 0 root, 0 non-root channels, 5
single-phone words
INFO: ngram_search_fwdtree.c(326): after: max nonroot chan increased to 6986
INFO: ngram_search_fwdtree.c(338): after: 147 root, 6858 non-root channels, 4
single-phone words
INFO: ngram_search_fwdflat.c(156): fwdflat: min_ef_width = 4, max_sf_win = 25
INFO: ngram_model_arpa.c(477): ngrams 1=1430, 2=2411, 3=3011
INFO: ngram_model_arpa.c(135): Reading unigrams
INFO: ngram_model_arpa.c(516): 1430 = #unigrams created
INFO: ngram_model_arpa.c(195): Reading bigrams
INFO: ngram_model_arpa.c(533): 2411 = #bigrams created
INFO: ngram_model_arpa.c(534): 80 = #prob2 entries
INFO: ngram_model_arpa.c(542): 122 = #bo_wt2 entries
INFO: ngram_model_arpa.c(292): Reading trigrams
INFO: ngram_model_arpa.c(555): 3011 = #trigrams created
INFO: ngram_model_arpa.c(556): 55 = #prob3 entries
INFO: ngram_search_fwdtree.c(99): 150 unique initial diphones
INFO: ngram_search_fwdtree.c(147): 0 root, 0 non-root channels, 5 single-phone
words
INFO: ngram_search_fwdtree.c(186): Creating search tree
INFO: ngram_search_fwdtree.c(191): before: 0 root, 0 non-root channels, 5
single-phone words
INFO: ngram_search_fwdtree.c(338): after: 147 root, 6858 non-root channels, 4
single-phone words
any help is appreciated
It gives you warnings in the log which you need to understand and fix
Thanks Nickolay
we use asterisk-unimrcp
this my dial paln
exten => 5554,1,Answer()
exten => 5554,2,SpeechCreate()
exten => 5554,3,SpeechLoadGrammar(digit,/usr/local/unimrcp/data/gramma$
exten => 5554,4,SpeechActivateGrammar(digit)
exten => 5554,5,SpeechBackground(hello-world,20)
exten => 5554,6,GotoIf($?7:9)
exten => 5554,7,Playback(vm-nonumber)
exten => 5554,8,Goto(5)
exten => 5554,9,Verbose(1,The recognized input is ${SPEECH_TEXT(0)})
exten => 5554,10,Verbose(1,The score is ${SPEECH_SCORE(0)})
exten => 5554,11,Verbose(1,The matched grammar is ${SPEECH_GRAMMAR(0)})
exten => 5554,12,SpeechDeactivateGrammar(digit)
exten => 5554,13,SpeechUnloadGrammar(digit)
exten => 5554,14,SpeechDestroy()
exten => 5554,15,Hangup()
when i call from sip to 5554
i got this error
== Using SIP RTP CoS mark 5
-- Executing Answer("SIP/6006-00000000", "") in new stack
-- Executing SpeechCreate("SIP/6006-00000000", "") in new stack
NOTICE: app_unimrcp.c:4169 unimrcp_log: Create MRCP Handle 0x8770988
NOTICE: app_unimrcp.c:4169 unimrcp_log: Add Control Channel 0x8770988
new@speechrecog
ip-118-139-163-102*CLI>
all xml can be found her
any help
This line is not correct, don't you think so?
sorry nshmyrev
but error in copying line this line is correct
any help ?
hi Nickolay
now i got this error
when we call from ZOIPER
the log of this asterisk is :
and the log of ./unimrcpserver is :
Unimrcp server logs describes the reason of the failure
You need to submit grammar in jsgf format.
hello nsh
we solve all problem and the unimrcp give us good result .
bu there is a small problem when hangup the unimrcpserver give an error
and this is my configuration
thanks for your help :)
unimrcpserver.xml
<components>
<resource-factory>
<resource id="speechsynth" enable="false"/>
<resource id="speechrecog" enable="true"/>
<resource id="recorder" enable="false"/>
</resource-factory>
<sip-uas id="SIP-Agent-1" type="SofiaSIP">
<sip-port>8070</sip-port>
<sip-transport>udp,tcp</sip-transport>
<ua-name>UniMRCP SofiaSIP</ua-name>
<sdp-origin>UniMRCPServer</sdp-origin>
</sip-uas>
<mrcpv2-uas id="MRCPv2-Agent-1">
<mrcp-port>1544</mrcp-port>
<max-connection-count>100</max-connection-count>
<force-new-connection>true</force-new-connection>
<rx-buffer-size>2048</rx-buffer-size>
<tx-buffer-size>2048</tx-buffer-size>
</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>
<engine id="Cepstral-Swift-1" name="mrcpcepstral" enable="false"/>
<engine id="PocketSphinx-1" name="mrcppocketsphinx" enable="true"/>
<engine id="Flite-1" name="mrcpflite" enable="false"/>
<engine id="Demo-Synth-1" name="demosynth" enable="false"/>
<engine id="Demo-Recog-1" name="demorecog" enable="true"/>
<engine id="Recorder-1" name="mrcprecorder" enable="true"/>
</plugin-factory>
</components>
<settings>
<rtp-settings id="RTP-Settings-1">
<jitter-buffer>
<playout-delay>50</playout-delay>
<max-playout-delay>200</max-playout-delay>
</jitter-buffer>
<ptime>20</ptime>
<codecs own-preference="true">GSM PCMU PCMA L16/96/64000 PCMU/97/64000
PCMA/98/16000 L16/99/64000 </codecs>
<rtcp enable="true">
<rtcp-bye>2</rtcp-bye>
<tx-interval>5000</tx-interval>
<rx-resolution>1000</rx-resolution>
</rtcp>
</rtp-settings>
</settings>
<profiles>
<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>
<resource-engine-map>
<param name="speechrecog" value="PocketSphinx-1"/>
</resource-engine-map>
</mrcpv2-profile>
</profiles>
</unimrcpserver>
mrcp.conf
; +++ MRCP settings +++
version = 1
;
; +++ RTSP +++
; === RSTP settings ===
server-ip = 118.139.163.102
server-port = 8070
; force-destination = 1
resource-location = media
speechsynth = speechsynthesizer
speechrecog = speechrecognizer
;
; +++ RTP +++
; === RTP factory ===
; rtp-ip = 0.0.0.0
rtp-ip = 118.139.163.102
; rtp-ext-ip = auto
rtp-port-min = 4000
rtp-port-max = 5000
; === RTP settings ===
; --- Jitter buffer settings ---
playout-delay = 50
; min-playout-delay = 20
max-playout-delay = 200
; --- RTP settings ---
ptime = 20
codecs = PCMU PCMA L16/96/8000
; --- RTCP settings ---
rtcp = 1
rtcp-bye = 2
rtcp-tx-interval = 5000
rtcp-rx-resolution = 1000
; +++ MRCP settings +++
version = 2
;
; +++ SIP +++
; === SIP settings ===
server-ip = 118.139.163.102
server-port = 8070
; server-username = test
force-destination = 1
; === SIP agent ===
; client-ip = 0.0.0.0
client-ip = 118.139.163.102
; client-ext-ip = auto
client-port = 5093
sip-transport = udp
; ua-name = Asterisk
; sdp-origin = Asterisk
;
; +++ RTP +++
; === RTP factory ===
; rtp-ip = 0.0.0.0
rtp-ip = 118.139.163.102
; rtp-ext-ip = auto
rtp-port-min = 4000
rtp-port-max = 5000
; === RTP settings ===
; --- Jitter buffer settings ---
playout-delay = 50
; min-playout-delay = 20
max-playout-delay = 200
; --- RTP settings ---
ptime = 20
codecs = PCMU PCMA L16/96/8000
; --- RTCP settings ---
rtcp = 1
rtcp-bye = 2
rtcp-tx-interval = 5000
rtcp-rx-resolution = 1000
pocketsphinx.xml
<timers noinput-timeout="10000" recognition-timeout="15"/>
<model dir="/home/hiyassat/tutorial/pocketsphinx/model/hmm/ar_hafiz/an4.cd_sem i_1000/" narrowband="/home/hiyassat/tutorial/pocketsphinx/model/hmm/ar_hafiz/a n4.cd_semi_1000/" wideband="wsj1" dictionary="/home/hiyassat/tutorial/pocketsphinx/model/lm/ar_hafiz/an4.dic" preferred="narrowband"/>
<save-waveform dir="/home/hiyassat/online_project/record" enable="1"/>
</pocketsphinx>
This message means there is a bug in the application. It might be a bug in
asterisk or in other modules. You need to collect stacktrace to provide more
information for developer. Without stacktrace it's hard to resolve this issue.
You need to contact developers about this issue, it's not CMUSphinx problem.
Hopefully you will not report any BSOD you meet here.
hello everyone , i'm beginner in asterisk and speech recognition and i want to make a project in university about IVR with speech recognition , well i learned a lot of doumentation and i have installed asterisk but i have a problem while i'm installing asterisk-unimrcp , i have asterisk 14.2.1 and unimrcp 1.4.0
and this is the error :
root@moemen-HP-15-Notebook-PC:/home/moemen/Téléchargements/asterisk-unimrcp-1.3.0# make
Making all in res-speech-unimrcp
make[1] : on entre dans le répertoire « /home/moemen/Téléchargements/asterisk-unimrcp-1.3.0/res-speech-unimrcp »
CC res_speech_unimrcp.lo
In file included from ../include/ast_compat_defs.h:30:0,
from res_speech_unimrcp.c:28:
/usr/include/asterisk.h:300:2: error: #error "Externally compiled modules must declare AST_MODULE_SELF_SYM."
#error "Externally compiled modules must declare AST_MODULE_SELF_SYM."
^
res_speech_unimrcp.c:31:23: error: expected declaration specifiers or ‘...’ before string constant
ASTERISK_FILE_VERSION(FILE, "$Revision: $")
^
res_speech_unimrcp.c:31:33: error: expected declaration specifiers or ‘...’ before string constant
ASTERISK_FILE_VERSION(FILE, "$Revision: $")
^
Makefile:444 : la recette pour la cible « res_speech_unimrcp.lo » a échouée
make[1]: [res_speech_unimrcp.lo] Erreur 1
make[1] : on quitte le répertoire « /home/moemen/Téléchargements/asterisk-unimrcp-1.3.0/res-speech-unimrcp »
Makefile:422 : la recette pour la cible « all-recursive » a échouée
make: [all-recursive] Erreur 1
Those are pretty self-explaining messages not really related to pocketsphinx.
The module is not compatible with asterisk version. You need to try older asterisk version or you can fix the module yourself. Usually it is not hard to add missing defines.