Differences

This shows you the differences between two versions of the page.

--- installation_guide:configuration_file_voximal:start [2016/01/18 21:57] – borja
+++ installation_guide:configuration_file_voximal:start [2016/12/02 15:20] (current) – [License Section [license]] borja
@@ Line 139: / Line 139: @@
 |audiowait | file | empty | Plays this file and waits for a free VoiceXML channel. |
 |mrcpsynthparams | string | empty | Add extra parameters for the MRCPsynth application (volume, keep session...) |
+|console | yes/no| empty | Enable the console log output of the interpreter, when you launch Asterisk with the console mode (-c), you can mix Asterisk and Voximal output logs on the console. |
 =====  License Section [license] =====
@@ Line 168: / Line 169: @@
 |max|number of channels|1|Max number of VoiceXML channels for this license.|
 |texttospeech|yes/no|yes|Allows to use Text to Speech prompts.|
-|video |yes/no|no|Allows to use Video options.|
+|speech |yes/no|no|Allows to use ASR/Speech to Text options.|
-|speech |yes/no|no|Allows to use ASR/Speech options.|
-|dialer |yes/no|no|Allows to use the VoiceXML with external Dialer.|
-|voicexml|yes/no|yes|A allows to use the VoiceXML browser execution.|
 |expiration|date YYMMDD| |Optional expiration date of the license|
 |key|License key|  |Key after purchasing a official commercial license. If not set or wrong key, you have 1 trial VoiceXML channel.|
+=====  Synthesis Section [prompt] =====
+The configuration of the Speech To Text connector.
+^Key ^Values ^Default ^Purpose |
+|api|api protocol|(empty)|HTTP protocol to use (defaule, Voximal, Microsoft, IBM/watson.|
+|uri|http uri|local PicoTTS|HTTP uri local or remote server.|
+=====  Recognize Section [recognize] =====
+^Key ^Values ^Default ^Purpose |
+|api|api protocol|(empty)|HTTP protocol to use (defaule, Voximal, Microsoft, IBM/watson.|
+|uri|http uri|local PicoTTS|HTTP uri local or remote server.|
 ===== Control Section [control] ======
@@ Line 269: / Line 281: @@
 |split |  |  |  |
-=====  Configuration example =====
-Example:
-<code>
-; VoiceXML Configuration
-;
-[general]
-wav_codec=gsm
-videosilence=;silence
-audiosilence=;silence
-debug=4
-video=yes
-[license]
-max=100
-key=…
-[account1]
-name=Test1
-url=http://localhost/vxml/index.php
-max=1
-[account2]
-name=Test2
-url=http://localhost/vxml/demo/index.vxml
-max=3
-dialformat=SIP/%s@ovh-out
-</code>
-To assign an extension to a VXML account just follow this example, where we are assigning the previous account to three extensions number in your /etc/asterisk/extension.conf asterisk:
-<code>
-[default]
-exten => 981001001,1,Voximal(Test1)
-exten => 981001002,1,Voximal(Test2)
-exten => 981001003,1,Voximal(Test3)
-</code>
-NOTE:
-When you update your voximal.conf file, remember to refresh configuration making a command “voximal reload” in your CLI*> prompt. If you have added SIP, PRI or new extensions you must launch “sip reload”, “extensions reload”, “dialplan reload” or reload asterisk/voximal processes. Use the command “voximal show accounts” to dump your accounts.
-<code>
-CLI*> diaplan reload
-CLI*> voximal reload
-CLI*> voximal show accounts
-</code>
-Example:
-Add extensions to the Asterisk dial plan /etc/asterisk/extensions.conf, if you want to set the URL in the dialplan:
-<code>
-exten => 888,1,Answer
-exten => 888,n,Wait(3)
-exten => 888,n,Voximal(file:///root/example.vxml)
-exten => 888,n,Hangup
-</code>
-You can create and edit the file /root/example.vxml with the GNU text editor, VI, for example.
-This example will work if you have text-to-speech configured.  If not, use a pre-recorded wav or gsm file to replace the “Hello world!” text by an <audio> tag.  For more information, see the format extensions supported by Asterisk.
-<code>
-<?xml version="1.0"?>
-<vxml version = "2.0" xmlns="http://www.w3.org/2001/vxml">
- <form>
-  <block><audio src="hello.wav"/></block>
- </form>
-</vxml>
-</code>
-Save the file in the same directory as the VoiceXML script (relative reference in this example).
-Reload the extensions configuration with:
-<code>
-*CLI> extensions reload
-</code>
-Call the service by calling:
-<code>
-SIP:888@<your server address>
-</code>