Asterisk + UniMRCP + Pocketsphinx help

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roko

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Jan 18, 2011, 5:02:34 PM1/18/11
to UniMRCP
Hi,

I'm configuring asterisk with UniMRCP and PocketSphinx as my ASR.

All seem to be installed but I'm a little bit confuse with configuring
Asterisk and UniMRCP to use PocketSphinx's ASR (I will use Nuance
Vocalizer TTS, I tested it and it is working fine).

Can anybody send me which files/properties do I need to configure.

I have read a lot of post and documentation at thw wiki, but as I
said, I'm a little bit confused.

Thanks in advance !

roko

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Jan 19, 2011, 3:05:21 PM1/19/11
to uni...@googlegroups.com
Hi Guys.

I'm configuring Asterisk+UniMRCP+PocketSPhinx (ASR)+Nuance (TTS) but when I just want to try the ASR (I haven't try TTS yet) I get this error in Asterisk's CLI:

[Jan 19 14:53:49] WARNING[19111]: app_unimrcp.c:4166 unimrcp_log: No Such Profile [uni2]


I read some posts and Client/Server deployment guides, but still don't get the problem solve.

Any help would be appreciate !


My architecture is:

O.S.: CentOS

Server 1: Asterisk, uniMRCP (uni-ast-package-0.3.0) and pocketSPhinx (pocketsphinx-0.6.1 & sphinxbase-0.6.1)

Server 2
: Nuance Vocalizer (TTS v8).

Firewall: UDP and TCP wide open between servers.


I just put back all configuration files and tried it again but it doesn't work.

This is the Dialplan:

exten => 6700,1,Answer()
exten => 6700,2,SpeechCreate()
exten => 6700,3,SpeechLoadGrammar(digit,/usr/local/unimrcp/data/grammar.jsgf)
exten => 6700,4,SpeechActivateGrammar(digit)
exten => 6700,5,SpeechBackground(hello-world,20)
exten => 6700,6,GotoIf($["${SPEECH(results)}" = "0"]?7:9)
exten => 6700,7,Playback(vm-nonumber)
exten => 6700,8,Goto(5)
exten => 6700,9,Verbose(1,The recognized input is ${SPEECH_TEXT(0)})
exten => 6700,10,Verbose(1,The score is ${SPEECH_SCORE(0)})
exten => 6700,11,Verbose(1,The matched grammar is ${SPEECH_GRAMMAR(0)})
exten => 6700,12,SpeechDeactivateGrammar(digit)
exten => 6700,13,SpeechUnloadGrammar(digit)
exten => 6700,14,SpeechDestroy()
exten => 6700,15,Hangup()



My Config. files (compared to the original config files using diff)

unimrcpserver.xml (original vs modified)

diff /mnt/src/uni-ast-package-0.3.0/unimrcp/conf/unimrcpserver.xml /usr/local/unimrcp/conf/unimrcpserver.xml
8,9c8,9
<     <ip type="auto"/>
<     <!-- <ip>10.10.0.1</ip> -->
---
>     <!-- <ip type="auto"/> -_>
>     <ip>10.244.19.143</ip>
87c87
<       <engine id="PocketSphinx-1" name="mrcppocketsphinx" enable="false"/>
---
>       <engine id="PocketSphinx-1" name="mrcppocketsphinx" enable="1"/>



unimrcpclient.xml (original vs modified)

diff /mnt/src/uni-ast-package-0.3.0/unimrcp/conf/unimrcpclient.xml /usr/local/unimrcp/conf/unimrcpclient.xml
11,12c11,12
<     <ip type="auto"/>
<     <!-- <ip>10.10.0.1</ip> -->
---
>     <!-- <ip type="auto"/> -->
>     <ip>10.244.19.143</ip>
99c99,112
<   </settings> 
---
>   </settings>
>   <profiles>
>
>     <!- PocketSphinx MRCPv2 profile ->
>     <mrcpv2-profile id="uni2">
>       <sip-uac>SIP-Agent-1</sip-uac>
>       <mrcpv2-uac>MRCPv2-Agent-1</mrcpv2-uac>
>       <media-engine>Media-Engine-1</media-engine>
>       <rtp-factory>RTP-Factory-1</rtp-factory>
>       <sip-settings>SIP-Settings-1</sip-settings>
>       <rtp-settings>RTP-Settings-1</rtp-settings>
>     </mrcpv2-profile>
>   </profiles>  
>



res-speech-unimrcp.conf

diff /mnt/src/uni-ast-package-0.3.0/asterisk-unimrcp/conf/res-speech-unimrcp.conf /mnt/.ub/etc/asterisk/res-speech-unimrcp.conf

NO DIFFERENCES



mrcp.conf

diff /mnt/src/uni-ast-package-0.3.0/asterisk-unimrcp/conf/mrcp.conf /mnt/.ub/etc/asterisk/mrcp.conf
2c2
< default-asr-profile = speech-nuance5-mrcp2
---
> default-asr-profile = asr-pocketsphinx-mrcp2
54c54,55
< server-ip = 192.168.1.3
---
> ;server-ip = 192.168.1.3
> server-ip = 10.202.163.112
60c61
< client-ip = 192.168.1.2
---
> client-ip = 10.202.163.112
70c71
< rtp-ip = 192.168.1.2
---
> rtp-ip = 10.202.163.112
87a89,129
> [asr-pocketsphinx-mrcp2]
> ; +++ MRCP settings +++
> version = 2
> ;
> ; +++ SIP +++
> ; === SIP settings ===
> server-ip = 10.244.19.143
> server-port = 5060
> ; server-username = test
> force-destination = 1
> ; === SIP agent ===
> ; client-ip = 0.0.0.0
> ;client-ip = 192.168.1.2
> client-ip = 10.244.19.143
> ; 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 = 192.168.1.2
> rtp-ip = 10.244.19.143
> ; 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


Any ideas ?

Thanks a lot !

roko

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Jan 19, 2011, 6:31:09 PM1/19/11
to uni...@googlegroups.com
Hi guys,

just to let you know that unimrcpclient.xml and unimrcpserver.xml failed to load so I copied these files again from the source and unimrcpclient and unimrcpserver binaries now work !

I'm configuring these files again step by step...
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