Voximal:无法连接到使用自定义ASR插件编译的UNIMRCP

Question

我已经编写了一个自定义的unimrcp asr插件，并希望它与星号上的voximal一起使用。

我在此处遵循文档：https：//wiki.voximal.com/doku.php?id=asrproviders：unimrcp。VXML工作正常，但是当我尝试在VXML中录制时，我看不到将任何流发送到UNIMRCP服务器。我的UNIMRCP服务器和星号都在同一台计算机上。我也尝试了EC2上的VoxiBot安装安装，但遇到了同样的问题

以下是星号中的一些配置：

mrcp.conf

[general]
; Default ASR and TTS profiles.
default-asr-profile = uni2
default-tts-profile = speech-nuance5-mrcp2
log-level = DEBUG,NOTICE,INFO
max-connection-count = 100
offer-new-connection = 1
; rx-buffer-size = 1024
; tx-buffer-size = 1024
; request-timeout = 5000
; speech-channel-timeout = 30000
[uni2]
version = 2
; SIP settings
server-ip = 172.17.0.2
server-port = 8060
; SIP user agent
;client-ip = 172.17.0.2
;client-port = 25097
sip-transport = udp
; RTP factory
rtp-ip = 172.17.0.2
rtp-port-min = 4000
rtp-port-max = 5000
; Jitter buffer settings
playout-delay = 50
max-playout-delay = 200

res-Speech-unimrcp.conf

[general]
; UniMRCP named profile. Options are:
unimrcp-profile = uni2      ; UniMRCP MRCPv2 Server
log-level = DEBUG,INFO,NOTICE
; Preloaded grammars
[grammars]
;grammar-name = path-to-grammar-file
[mrcpv2-properties]
Recognition-Timeout = 20000
No-Input-Timeout = 15000
[mrcpv1-properties]
Recognition-Timeout = 20000
No-Input-Timeout = 15000

voximal.conf

[general]
autoanswer=yes
videosilence=
audiosilence=
; tried with speechprovider=unimrcp too
speechprovider=unimrcp:uni2
speechscore=50
[control]
forward=#
reverse=*
stop=123456789
pause=
restart=0
skipms=5000
;Optional local license
[license]
;max=1
;key=trial
tts=yes
speech=auto
[prompt]
uri=http://ttsf.voximal.net/tts/pico/tts.php
method=post
format=wav
maxage=-1

[recognize]
sendproperties=0
[account1]
number=8965
name=helloworld
url=file:///var/lib/voximal/record.vxml
speech=automatic

record.vxml

<!-- for testing recording -->
<?xml version="1.0" encoding="UTF-8"?>
<vxml version="2.0" xmlns="http://www.w3.org/2001/vxml" xml:lang="en-US">
    <form>
        <block>
            <prompt>
                <audio src="/var/lib/asterisk/sounds/speech_start.wav"/>
            </prompt>
        </block>
        <record  name="msg" beep="true" maxtime="10s" finalsilence="4000ms" dtmfterm="true" type="audio/x-wav">
            <prompt timeout="5s">
                <audio src="/var/lib/asterisk/sounds/speech_start.wav"/>
            </prompt>
        </record>
    </form>
</vxml>

Answer 1

我看不到ASR和记录功能之间的链接(用于录制用户的声音而没有。如果要记录发送到ASR的音频流，则可以使用属性" recordutterance"(true/false(，您将拥有一个shadow varible field_name $ .recording，field_name $ .recordingsize，field_name $ .recordingDuration。