UniMRCP integration allows Asterisk to facilitate the use of the industry standard media resource servers for implementation of speech-enabled applications in accordance with the MRCP specifications. UniMRCP supports both version 1 and 2 of the MRCP protocol.

This guide provides instructions on how to install, configure, and use the UniMRCP modules for Asterisk.

1.1 Applicable Versions

Instructions provided in this guide are applicable to the following versions.

UniMRCP modules for Asterisk 1.1.0 to 1.2.0

Check out the new/updated guide for versions starting from 1.3.0.

http://unimrcp.org/asterisk

1.2 Available Modules

Generic Speech Recognition API

The module res_speech_unimrcp.so is an implementation of the Generic Speech Recognition API of Asterisk based on the UniMRCP client library. See Section 6 for more information.

Suite of UniMRCP Applications

The module app_unimrcp.so is a suite of speech recognition and synthesis applications for Asterisk. See Section 7 for more information.

2. Licensing

Since Asterisk is distributed under the GPLv2 license, and the UniMRCP modules are loaded by and directly interface with Asterisk, the GPLv2 license applies to the UniMRCP modules too.

See the following link for more information regarding the GPLv2 license.

GNU General Public License, version 2

3. Supported Operating Systems

The following Red Hat based Linux distributions are primarily supported.

Operating System	32-bit	64-bit
RHEL / CentOS 5.x
RHEL / CentOS 6.x

Other Linux distributions such as Ubuntu, Debian should also be supported with no or minimal changes in the installation procedure.

4. Supported MRCP Servers

The UniMRCP modules for Asterisk are known to work with the following MRCP servers.

MRCP Server	ASR	TTS
UniMRCP Server
LumenVox Media Server
Nuance Speech Server

Note: installation of the MRCP server is not covered in this document.

5. Installation

5.1 Prerequisites

The development tools such as autoconf, libtool, gcc, and others must be installed on the system. The following command can be used to install all the prerequisites on a RHEL based distribution:

yum groupinstall "Development Tools"

Several 3-rd party development libraries are additionally required for the installation of Asterisk. The libraries can be installed issuing the following commands:

yum install ncurses-devel

yum install libxml2-devel

yum install sqlite-devel

yum install openssl-devel

5.2 Installing the Release Package

Package Retrieval

The latest UniMRCP release for Asterisk can be downloaded from the following location:

http://www.unimrcp.org/downloads/asterisk

The release package is a source archive file which needs to be extracted and copied to a local directory. The following software components are included in the package:

❖ Asterisk

❖ UniMRCP

❖ UniMRCP Dependencies

❖ UniMRCP Modules for Asterisk

For versions of the components included in a particular release package, see the file INSTALL.

Installation of Asterisk

If a compatible version of Asterisk has already been installed on the system, this section can be skipped. The supported versions of Asterisk are:

Asterisk 1.6, 1.8, 10, 11

Asterisk 12 is supported only by app_unimrcp.so

In order to install Asterisk included in the package, run the asterisk-install.sh script. The script may also optionally install sample configuration files.

Installation of UniMRCP

If a compatible version of UniMRCP has already been installed on the system, this section can be skipped. The supported versions of UniMRCP are:

UniMRCP r1988 and above

In order to install UniMRCP included in the package, run the unimrcp-install.sh script. The script will also prompt to install the dependencies.

Installation of Modules

In order to install the UniMRCP modules for Asterisk included in the package, run the modules-install.sh script.

5.3 Installing from the Repository

Before starting installation of the UniMRCP modules for Asterisk from repository, both Asterisk and UniMRCP must have already been installed.

Source Retrieval

Use the following command to check out the latest source:

svn checkout http://unimrcp.googlecode.com/svn/solutions/asterisk-unimrcp

Build Procedure

The bootstrap script must be run first in order to generate the configure script and other required files.

./bootstrap

The usual configure, make, make install sequence of commands should follow in order to build and install the modules from source.

./configure

make

make install

As a result, the modules res_speech_unimrcp.so and app_unimrcp.so will be installed in the modules directory of Asterisk such as /usr/lib/asterisk/modules by default. Similarly, the configuration files res-speech-unimrcp.conf and mrcp.conf will be placed in /etc/asterisk by default.

Configure Options

There are a number of options which can be additionally configured.

To explicitly specify where to look for Asterisk, use the option --with-asterisk. For example, if Asterisk is installed in /usr/local/asterisk-11, use:

./configure --with-asterisk=/usr/local/asterisk-11

To explicitly specify where the Asterisk configuration files are located, use the option --with-asterisk-conf. For example:

./configure --with-asterisk-conf=/usr/local/asterisk/conf

To explicitly specify the Asterisk version, use the option --with-asterisk-version. For example:

./configure --with-asterisk-version=11.2.1

To explicitly specify where to look for UniMRCP, use the option --with-unimrcp. For example, if UniMRCP is installed in /usr/local/unimrcp-1.1, use:

./configure --with-unimrcp=/usr/local/unimrcp-1.1

To exclude the module res_speech_unimrcp.so from build, use:

./configure --disable-res-speech-unimrcp

To exclude the module app_unimrcp.so from build, use:

./configure --disable-app-unimrcp

5.4 Testing the Installation

Run Asterisk and check if the UniMRCP modules are loaded normally.

*CLI> module show like res_speech_unimrcp.so

Module Description Use Count

res_speech_unimrcp.so UniMRCP Speech Engine 0

1 modules loaded

*CLI> module show like app_unimrcp.so

Module Description Use Count

app_unimrcp.so MRCP suite of applications 0

1 modules loaded

6. Generic Speech Recognition API

6.1 Overview

The module res_speech_unimrcp.so is an implementation of the Generic Speech Recognition API of Asterisk based on the UniMRCP client library.

6.2 Configuration

The module res_speech_unimrcp.so loads the UniMRCP client stack by means of the XML configuration format defined in the Client Configuration Guide. The configuration file is located in the directory /usr/local/unimrcp/unimrcpclient.xml by default.

Module-specific configuration parameters are specified in the file res-speech-unimrcp.conf, located in the directory /etc/asterisk by default. This configuration file uses the Asterisk native configuration format, where a configuration file is broken into various sections with the section name surrounded by square brackets.

General Settings

The section general contains global settings of the module.

Name

[general]

Settings

Name	Description
unimrcp-profile	Specifies the profile defined in the UniMRCP client configuration. The profiles are located by default in the directory /usr/local/unimrcp/client-profiles.
log-level	Specifies the log level, one of EMERGENCY, ALERT, CRITICAL, ERROR, WARNING, NOTICE, INFO, DEBUG.

Example

The section general contains the setting unimrcp-profile set to uni2 and the setting log-level set to DEBUG.

[general]

unimrcp-profile = uni2

log-level = DEBUG

Common Grammars

The section grammars optionally specifies a set of grammars which are applicable to and need to be loaded for all the MRCP sessions established from the client to the server. The grammars are specified by means of name-value pairs, where the name is an arbitrary (but unique) name of the grammar and the value specifies a grammar file path.

Name

[grammars]

Settings

Name	Description
“grammar-name”	Maps the grammar name to the grammar file path.

Example

The section grammar specifies two grammars which are supposed to be implicitly used with all the MRCP sessions.

[grammars]

yes-no = /usr/local/unimrcp/data/yes-no. xml

exit = /usr/local/unimrcp/data/exit. Xml

MRCPv2 Properties

The section mrcpv2-properties specifies a set of header fields to be used for all the MRCPv2 sessions established from the client to the server. The applicable header fields are defined in the section Recognizer Header Fields.

Name

[mrcpv2-properties]

Settings

Name	Description
“header-field-name”	Maps the header field name to the header field value.

Example

The section mrcpv2-properties specifies the recognition and no-input timeouts.

[mrcpv2-properties]

Recognition-Timeout = 20000

No-Input-Timeout = 15000

MRCPv1 Properties

The section mrcpv1-properties specifies a set of header fields to be used for all the MRCPv1 sessions established from the client to the server. The applicable header fields are defined in the section Recognizer Header Fields.

Name

[mrcpv1-properties]

Settings

Name	Description
“header-field-name”	Maps the header field name to the header field value.

Example

The section mrcpv1-properties specifies the recognition and no-input timeouts.

[mrcpv1-properties]

Recognition-Timeout = 20000

No-Input-Timeout = 15000

6.3 Dialplan Applications

SpeechCreate

Synopsis

Creates a recognition channel.

Syntax

SpeechCreate(Engine Name)

Input parameters

· Engine Name - The engine name is unimrcp. If not specified, default engine will be used

SpeechLoadGrammar

Synopsis

Loads grammar on a channel.

Syntax

SpeechLoadGrammar(Grammar Name|[Content-Type:]Path)

Input parameters

· Grammar Name - The name of the grammar to load

· Path - The path to grammar file

SpeechLoadGrammar(grammar,path) - Loads a grammar from the specified location. Grammar type will be implicitly determined.

SpeechLoadGrammar(grammar,content-type:path) - Loads a grammar from the specified location. Grammar type is explicitly specified.

SpeechLoadGrammar(grammar,builtin:grammar/digits) - Specifies a built-in grammar.

SpeechUnloadGrammar

Synopsis

Unloads grammar on a channel.

Syntax

SpeechUnloadGrammar(Grammar Name)

Input parameters

· Grammar Name - The name of the grammar to unload

SpeechActivateGrammar

Synopsis

Activates the specified grammar on a channel.

Syntax

SpeechActivateGrammar(Grammar Name)

Input parameters

· Grammar Name - The name of the grammar to activate

SpeechStart

Synopsis

Starts speech recognition on a channel.

Syntax

SpeechStart()

SpeechBackground

Synopsis

Plays a file and waits for input.

Syntax

SpeechBackground(Sound File|Timeout)

Input parameters

· Sound File - The sound file to play

· Timeout - Timeout to wait

Output results

· ${SPEECH(status)}

· ${SPEECH(spoke)}

· ${SPEECH(results)}

· ${SPEECH_SCORE(result number)}

· ${SPEECH_TEXT(result number)}

· ${SPEECH_GRAMMAR(result number)}

SpeechDeactivateGrammar

Synopsis

Deactivates the specified grammar on a channel.

Syntax

SpeechDeactivateGrammar(Grammar Name)

Input parameters

· Grammar Name - The name of the grammar to deactivate

SpeechProcessingSound

Synopsis

Changes the file that SpeechBackground is playing.

Syntax

SpeechProcessingSound(Sound File)

Input parameters

· Sound File - The sound file to play

SpeechDestroy

Synopsis

Destroys channel.

Syntax

SpeechDestroy()

7. Suite of UniMRCP Applications

7. 1 Overview

The module app_unimrcp.so is a suite of speech recognition and synthesis applications for Asterisk.

7. 2 Configuration

The module app_unimrcp.so uses the Asterisk native configuration format, where a configuration file is broken into various sections with the section name surrounded by square brackets. The configuration file

mrcp.conf is located in /etc/asterisk by default and consists of general settings and MRCP profiles.

General Settings

The section general contains global settings of the module.

Name

[general]

Settings

Name	Description
default-asr-profile	Specifies the default profile to be used for speech recognition. More than one profile can be defined in the same configuration file.
default-tts-profile	Specifies the default profile to be used for speech synthesis. More than one profile can be defined in the same configuration file.
log-level	Specifies the log level, one of EMERGENCY, ALERT, CRITICAL, ERROR, WARNING, NOTICE, INFO, DEBUG.
max-connection-count	Specifies the maximum number of TCP/MRCPv2 connections.
offer-new-connection	Specifies whether to offer the server to establish a new connection or re-use the existing one.
rx-buffer-size	The size of the buffer which temporarily holds the data received from the network socket.
tx-buffer-size	The size of the buffer which temporarily holds the data to be sent via the network socket.
request-timeout	Specifies the network timeout used for outgoing MRCPv2 and RTSP (MRCPv1) requests.

Example

The section general contains some typical settings.

[general]

default-asr-profile = unimrcpv2

default-tts-profile = unimrcpv2

log-level = DEBUG

max-connection-count = 100

offer-new-connection = 1

MRCP Profiles

One or more profiles can be specified. Each section in the configuration corresponds to a particular profile, where the section name specifies the name of the profile and the settings are profile parameters.

Name

[“unique-profile-name”]

Settings

Name	Description
version	Specifies the MRCP version: 1 or 2.
server-ip	Specifies the IP address of the server.
server-port	Specifies the SIP (MRCPv2) or RTSP (MRCPv1) port on the server.
client-ip [MRCPv2 only]	Specifies the IP address of the client (Asterisk) to be used for SIP signaling.
client-port [MRCPv2 only]	Specifies the local SIP port of the client.
sip-transport [MRCPv2 only]	Specifies the SIP transport: UDP, TCP
resource-location [MRCPv1 only]	Specifies the MRCP resource location on the server.
speechsynth [MRCPv1 only]	Specifies the name of the speech synthesizer resource.
speechrecog [MRCPv1 only]	Specifies the name of the speech recognizer resource.
rtp-ip	Specifies the client (Asterisk) IP address to be used for RTP streaming.
rtp-port-min	Specifies the first (inclusive) port number in the RTP port range.
rtp-port-max	Specifies the last (exclusive) port number in the RTP range.
playout-delay	Specifies the initial playout delay in the jitter buffer.
min-playout-delay	Specifies the min playout delay in the jitter buffer.
max-playout-delay	Specifies the max playout delay in the jitter buffer.
ptime	Specifies the RTP packetization time.
codecs	Specifies the RTP supported codecs.
rtcp	Specifies whether RTCP is enabled or not.

Example 1

This is a typical MRCPv1 profile, where the client (Asterisk) IP address is 192.168.0.29 and the server IP address is 192.168.0.30.

[unimrcpv1]

; MRCP settings

version = 1

;

; RTSP settings

server-ip = 192.168.0.30

server-port = 1554

resource-location = media

speechsynth = speechsynthesizer

speechrecog = speechrecognizer;

;

; RTP factory

rtp-ip = 192.168.0.29

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

Example 2

This is a typical MRCPv2 profile, where the client (Asterisk) IP address is 192.168.0.29 and the server IP address is 192.168.0.30.

[unimrcpv2]

; MRCP settings

version = 2

;

; SIP settings

server-ip = 192.168.0.30

server-port = 8060

; SIP user agent

client-ip = 192.168.0.29

client-port = 25097

sip-transport = udp

;

; RTP factory

rtp-ip = 192.168.0.29

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

7. 3 Dialplan Applications

SynthAndRecog

Synopsis

Plays a synthesized prompt and waits for speech to be recognized.

Syntax

SynthAndRecog(prompt, grammar, options)

Description

This application establishes two MRCP sessions: one for speech synthesis and the other for speech recognition. Once the user starts speaking (barge-in occurred), the synthesis session is stopped, and the recognition engine starts processing the input. Once recognition completes, the application exits and returns results to the Asterisk dialplan.

Input parameters

· prompt [required]

A prompt specified as a plain text, an SSML content, or by means of a file or URI reference.

· grammar [required]

An inline or URI grammar to be used for recognition.

· options

Additional parameters such as an MRCP profile, synthesizer and recognizer header fields to be used.

p - profile to use in mrcp.conf

t - recognition timeout (msec)

b - bargein value (no barge-in allowed=0, barge-in allowed=1)

gd – grammar delimiters

ct - confidence threshold (0.0 - 1.0)

sl - sensitivity level (0.0 - 1.0)

sva - speed vs accuracy (0.0 - 1.0)

nb - n-best list length

nit - no input timeout (msec)

sct - speech complete timeout (msec)

sint - speech incomplete timeout (msec)

dit - DTMF interdigit timeout (msec)

dtt - DTMF terminate timout (msec)

dttc - DTMF terminate characters

sw - save waveform (true/false)

nac - new audio channel (true/false)

spl - speech language (en-US/en-GB/etc.)

rm - recognition mode (normal/hotword)

hmaxd - hotword max duration (msec)

hmind - hotword min duration (msec)

cdb - clear DTMF buffer (true/false)

enm - early no match (true/false)

iwu - input waveform URI

mt - media type

pv - prosody volume (silent/x-soft/soft/medium/load/x-loud/default)

pr - prosody rate (x-slow/slow/medium/fast/x-fast/default)

vn - voice name to use (e.g. "Daniel", "Karin", etc.)

vg - voice gender to use (e.g. "male", "female")

vv - voice variant

a - voice age to use

uer - URI-encoded results (1: URI-encode NLMSL results, 0: do not encode)

Return parameters

· RECOG_STATUS

If recognition completed, the variable ${RECOG_STATUS} is set to OK. Otherwise, if recognition could not be started, the variable ${RECOG_STATUS} is set to ERROR. If the caller hung up while recognition was still in-progress, the variable ${RECOG_STATUS} is set to INTERRUPTED.

· RECOG_COMPLETION_CAUSE

The variable ${RECOG_COMPLETION_CAUSE} indicates whether recognition completed successfully with a match or an error occurred.

("000" - success, "001" - nomatch, "002" - noinput)

· RECOG_RESULT

If recognition completed successfully, the variable ${RECOG_RESULT} is set to an NLSML result received from the MRCP server. Alternatively, the recognition result data can be retrieved by using the following dialplan functions RECOG_CONFIDENCE(), RECOG_GRAMMAR(), RECOG_INPUT(), and RECOG_INSTANCE().

Example 1

This context demonstrates how to use the application SynthAndRecog() with a plain-text prompt and a built-in speech grammar.

[synthandrecog-app1]

exten => s,1,Answer

exten => s,n,SynthAndRecog(Please say a number,builtin:grammar/number,t=5000&b=1&ct=0.7&spl=en-US)

exten => s,n,Verbose(1, ${RECOG_STATUS}, ${RECOG_COMPLETION_CAUSE}, ${RECOG_RESULT})

exten => s,n,Hangup

Example 2

This context demonstrates how to use the application SynthAndRecog() with a plain-text prompt and a built-in DTMF grammar.

[synthandrecog-app2]

exten => s,1,Answer

exten => s,n,SynthAndRecog(Please input a number,builtin:dtmf/number,t=5000&b=1&ct=0.7&spl=en-US)

exten => s,n,Verbose(1, ${RECOG_STATUS}, ${RECOG_COMPLETION_CAUSE}, ${RECOG_RESULT})

exten => s,n,Hangup

Example 3

This context demonstrates how to use the application SynthAndRecog() with an SSML prompt and an inline SRGS XML speech grammar.

[synthandrecog-app3]

exten => s,1,Answer

exten => s,n,SynthAndRecog(<?xml version=\"1.0\"?><speak version=\"1.0\" xmlns=\"http://www.w3.org/2001/10/synthesis\" xmlns:xsi=\"http://www.w3.org/2001/XMLSchema-instance\" xsi:schemaLocation=\"http://www.w3.org/2001/10/synthesis http://www.w3.org/TR/speech-synthesis/synthesis.xsd\" xml:lang=\"en-US\">Please pick a color: red <break/> green <break/> or blue </speak>,<?xml version=\"1.0\"?><grammar xmlns=\"http://www.w3.org/2001/06/grammar\" xml:lang=\"en-US\" version=\"1.0\" mode=\"voice\" tag-format=\"semantics/1.0-literals\" root=\"color\"><rule id=\"color\"><one-of><item>red</item><item>green</item><item>blue</item></one-of></rule></grammar>,t=5000&b=1&ct=0.7)

exten => s,n,Verbose(1, ${RECOG_STATUS}, ${RECOG_COMPLETION_CAUSE}, ${RECOG_RESULT})

exten => s,n,Hangup

Example 4

This context demonstrates how to use the application SynthAndRecog() with an SSML prompt and an inline SRGS XML DTMF grammar.

[synthandrecog-app4]

exten => s,1,Answer

exten => s,n,SynthAndRecog(<?xml version=\"1.0\"?><speak version=\"1.0\" xmlns=\"http://www.w3.org/2001/10/synthesis\" xmlns:xsi=\"http://www.w3.org/2001/XMLSchema-instance\" xsi:schemaLocation=\"http://www.w3.org/2001/10/synthesis http://www.w3.org/TR/speech-synthesis/synthesis.xsd\" xml:lang=\"en-US\">Please pick a color: for red press 1 <break/> for green press 2 <break/> for blue press 3 </speak>,<?xml version=\"1.0\"?><grammar xmlns=\"http://www.w3.org/2001/06/grammar\" version=\"1.0\" mode=\"dtmf\" tag-format=\"semantics/1.0-literals\" root=\"color\"><rule id=\"color\"><one-of><item>1<tag>red</tag></item><item>2<tag>green</tag></item><item>3<tag>blue</tag></item></one-of></rule></grammar>,t=5000&b=1&ct=0.7)

exten => s,n,Verbose(1, ${RECOG_STATUS}, ${RECOG_COMPLETION_CAUSE}, ${RECOG_RESULT})

exten => s,n,Hangup

Example 5

This context demonstrates how to use the application SynthAndRecog() with a non-bargeinable prompt.

[synthandrecog-app5]

exten => s,1,Answer

exten => s,n,SynthAndRecog("This is a non-bargeinable prompt. When the prompt is finished, say a number",builtin:grammar/number,t=5000&b=0&ct=0.7&spl=en-US)

exten => s,n,Verbose(1, ${RECOG_STATUS}, ${RECOG_COMPLETION_CAUSE}, ${RECOG_RESULT})

exten => s,n,Hangup

Example 6

This context demonstrates how to use the application SynthAndRecog() with a prompt and a grammar referenced from files.

[synthandrecog-app6]

exten => s,1,Answer

exten => s,n,SynthAndRecog(/usr/local/unimrcp/data/speak.xml,/usr/local/unimrcp/data/grammar.xml,t=5000&b=1&ct=0.7&spl=en-US)

exten => s,n,Verbose(1, ${RECOG_STATUS}, ${RECOG_COMPLETION_CAUSE}, ${RECOG_RESULT})

exten => s,n,Hangup

Example 7

This context demonstrates how to use the application SynthAndRecog() with a grammar referenced by an HTTP URI.

[synthandrecog-app7]

exten => s,1,Answer

exten => s,n,SynthAndRecog(Please say a digit,http://unimrcp.net/data/grammar.xml,t=5000&b=1&ct=0.7&spl=en-US)

exten => s,n,Verbose(1, ${RECOG_STATUS}, ${RECOG_COMPLETION_CAUSE}, ${RECOG_RESULT})

exten => s,n,Hangup

Example 8

This context demonstrates how to use the application SynthAndRecog() with multiple grammars specified using a comma-separated list.

[synthandrecog-app8]

exten => s,1,Answer

exten => s,n,SynthAndRecog(Please say a number,"builtin:grammar/number,builtin:dtmf/number",t=5000&b=1&ct=0.7&spl=en-US)

exten => s,n,Verbose(1, ${RECOG_STATUS}, ${RECOG_COMPLETION_CAUSE}, ${RECOG_RESULT})

exten => s,n,Hangup

MRCPRecog

Synopsis

Performs MRCP recognition.

Syntax

MRCPRecog(grammar, options)

Description

This application establishes an MRCP session for speech recognition and optionally plays a prompt file to the user. Once recognition completes, the application exits and returns results to the Asterisk dialplan.

Input parameters

· grammar [required]

An inline or URI grammar to be used for recognition.

· options

Additional parameters such as an MRCP profile, recognizer header fields to be used.

p - profile to use in mrcp.conf

i - digits to allow recognition to be interrupted with (by default DTMFs are sent to the MRCP server to recognize; otherwise, if "any" or other digits are specified, recognition will be interrupted)

f - filename on play (if empty or not specified, no file is played)

t - recognition timeout (msec)

b - bargein value (no barge-in allowed=0, barge-in allowed=1)

gd – grammar delimiters

ct - confidence threshold (0.0 - 1.0)

sl - sensitivity level (0.0 - 1.0)

sva - speed vs accuracy (0.0 - 1.0)

nb - n-best list length

nit - no input timeout (msec)

sit - start input timers (true/false)

sct - speech complete timeout (msec)

sint - speech incomplete timeout (msec)

dit - DTMF interdigit timeout (msec)

dtt - DTMF terminate timout (msec)

dttc - DTMF terminate characters

sw - save waveform (true/false)

nac - new audio channel (true/false)

spl - speech language (en-US/en-GB/etc.)

rm - recognition mode (normal/hotword)

hmaxd - hotword max duration (msec)

hmind - hotword min duration (msec)

cdb - clear DTMF buffer (true/false)

enm - early no match (true/false)

iwu - input waveform URI

mt - media type

epe: exit on play error (1: terminate recognition on file play error, 0: continue even if file play fails)

uer: URI-encoded results (1: URI-encode NLMSL results, 0: do not encode)

Return parameters

· RECOGSTATUS

If recognition completed, the variable ${RECOGSTATUS} is set to OK. Otherwise, if recognition could not be started, the variable ${RECOGSTATUS} is set to ERROR. If the caller hung up while recognition was still in-progress, the variable ${RECOGSTATUS} is set to INTERRUPTED.

· RECOG_COMPLETION_CAUSE

The variable ${RECOG_COMPLETION_CAUSE} indicates whether recognition completed successfully with a match or an error occurred.

("000" - success, "001" - nomatch, "002" - noinput)

· RECOG_RESULT

Example

This context demonstrates how to use the application MRCPRecog() with a pre-recorded prompt file and a built-in speech grammar.

[mrcprecog-app1]

exten => s,1,Answer

exten => s,n,MRCPRecog(builtin:grammar/number,p=default&t=5000&b=1&ct=0.7&spl=en-US&f=hello-world)

exten => s,n,Verbose(1, ${RECOGSTATUS}, ${RECOG_COMPLETION_CAUSE}, ${RECOG_RESULT})

exten => s,n,Hangup

MRCPSynth

Synopsis

Performs MRCP synthesis.

Syntax

MRCPSynth (prompt, grammar, options)

Description

This application establishes an MRCP session for speech synthesis.

Input parameters

· prompt [required]

A prompt specified as a plain text, an SSML content, or by means of a file or URI reference.

· options

Additional parameters such as an MRCP profile, synthesizer header fields to be used.

p - profile to use in mrcp.conf

i - digits to allow the TTS to be interrupted with

f - filename on disk to store audio to (audio not stored if not specified or empty)

l - language to use (e.g. "en-GB", "en-US", "en-AU", etc.)

v - voice name to use (e.g. "Daniel", "Karin", etc.)

g - voice gender to use (e.g. "male", "female")

a - voice age to use

pv - prosody volume (silent/x-soft/soft/medium/load/x-loud/default)

pr - prosody rate (x-slow/slow/medium/fast/x-fast/default)

ll - load lexicon (true/false)

vv - voice variant

Return parameters

· SYNTHSTATUS

If synthesis completed successfully, the variable ${SYNTHSTATUS} is set to OK; otherwise, if an error occurred, the variable ${SYNTHSTATUS} is set to ERROR. If the caller hung up while the synthesis was in-progress, the variable ${SYNTHSTATUS} is set to INTERRUPTED.

Example

This context demonstrates how to use the application MRCPSynth() with a plain-text prompt.

[mrcpsynth-app1]

exten => s,1,Answer

exten => s,n,MRCPSynth(Hello world!,p=default)

exten => s,n,Verbose(1, ${SYNTHSTATUS})

exten => s,n,Hangup

7. 4 Dialplan Functions

RECOG_CONFIDENCE

Synopsis

Get the confidence score of an interpretation.

Syntax

RECOG_CONFIDENCE(nbest_number)

Description

This function returns the confidence score of the specified interpretation.

Input parameters

· nbest_number

The parameter nbest_number specifies the index in the list of interpretations sorted best-first. This parameter defaults to 0, if not specified.

RECOG_GRAMMAR

Synopsis

Gets the matched grammar of an interpretation.

Syntax

RECOG_GRAMMAR(nbest_number)

Description

This function returns the matched grammar of the specified interpretation.

Input parameters

· nbest_number

The parameter nbest_number specifies the index in the list of interpretations sorted best-first. This parameter defaults to 0, if not specified.

RECOG_INPUT

Synopsis

Gets the spoken input.

Syntax

RECOG_INPUT(nbest_number)

Description

This function returns the spoken input.

Input parameters

· nbest_number

The parameter nbest_number specifies the index in the list of interpretations sorted best-first. This parameter defaults to 0, if not specified.

RECOG_INSTANCE

Synopsis

Gets the interpreted instance.

Syntax

RECOG_INSTANCE(nbest_number/instance_number)

Description

This function returns the interpreted instance.

Input parameters

· nbest_number

The parameter nbest_number specifies the index in the list of interpretations sorted best-first. This parameter defaults to 0, if not specified.

· instance_number

The parameter instance_number specifies the index in the list of instances for a particular interpretation. This parameter defaults to 0, if not specified.