Configuration

The configuration file is /etc/asterisk/vxml.conf. It is editable by typing:

# vi /etc/asterisk/vxml.conf

The vxml.conf file is divided in serveral sections to set different group of parameters and functions:

; commented text starts with a “;”

[general]
...

[control]
...

[license]
...

[account1]
...

[account2]
...

[accountn]
...

NOTE:

Please note that before configuring the Asterisk Module, you should make a backup copy of vxml.conf. Vxi upgrade process keep the current configuration but some specific updates can be mandatory for future releases.

The following section covers general functions of VoiceXML Asterisk module.

The following section covers license information and keys of Vxi. To obtain a commercial valid license key for your copy of VoiceXML browser (for use more than one session), you should run asterisk and get your code by typing the following commands:

*CLI> vxml show license

Default License section (no key) allow to run any VoiceXML application with one free port.

Example:

[license]
max=1
key=

Example:

[license]
max=30
key=yourkey

The control section describes the control functions that allow you to configure DTMF commands that stream a file with fast forward, pause, reverse, restart, etc. To enable a control during a prompt, you must first set the VoiceXML property name to equal “control, with a value of “yes.”

<property name=”control” value=”yes” />

This works only with GSM and WAV files - in all other cases the property is ignored.

The Vxi VoiceXML browser is designed to manage hosted VoiceXML services and share different applications thru different Inbound / Outbound phone lines. This will allow you to manage customers’ accounts for VoiceXML hosted applications and control easily the capacity you need to assign to each one.

To create accounts, you need to add an [accountX] section (where “X” is a number 1-100) in the bottom part of the /etc/asterisk/vxml.conf file. Please find here three added accounts to manage different port capacity per application/URI:

Examples:

[account1]
name=voiceportal1
local=0099090
url=http://www.i6net.com/vxml/voiceportal1.vxml
max=25

[account2]
name=helloworld
local=0099090
url=http://www.i6net.com/vxml/helloword/index.vxml
max=10

[account3]
name=localservice
local=0099090
url=http://localhost/vxml/test.vxml
max=51

The account properties are defined within sections. Each account has a specific section. The sections are named and numbered from account1 to account99 (maximun 100 accounts). The example below shows the functions in each account section.

[account1]
name=example
number=*3
url=http://…/index.vxml
urlvideo=http://…/index.vxml
max=5
speech=no
dialformat=SIP/%s
dialformatvideo=SIP/%s

Example:

; VoiceXML Configuration
;
[general]
wav_codec=gsm
videosilence=;silence
audiosilence=;silence
debug=4
video=yes

[license]
max=100
key=…

[account1]
name=Test1
url=http://host.i6net.com/vxml/links/vxml/index.php
max=1

[account2]
name=Test2
url=http://www.mdfactory.com/mpm/camp/index.vxml
max=3
dialformat=SIP/%s@voztele-out

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:

[default]

exten => 981001001,1,Vxml(Test1)
exten => 981001002,1,Vxml(Test2)
exten => 981001003,1,Vxml(Test3)

NOTE: When you update your vxml.conf file, remember to refresh configuration making a command “vxml 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/openvxi processes. Use the command “vxml show accounts” to dump your accounts.

CLI*> extensions reload
CLI*> vxml reload
CLI*> vxml show accounts

Example:

Add extensions to the Asterisk dial plan /etc/asterisk/extensions.conf, if you want to set the URL in the dialplan:

exten => 888,1,Answer
exten => 888,n,Wait(3)
exten => 888,n,Vxml(file:///root/example.vxml)
exten => 888,n,Hangup

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.

<?xml version="1.0"?>
<vxml version = "2.0" xmlns="http://www.w3.org/2001/vxml">
 <form>
  <block><audio src="hello.wav"/></block>
 </form>
</vxml>

Save the file in the same directory as the VoiceXML script (relative reference in this example). Reload the extensions configuration with:

*CLI> extensions reload

Call the service by calling:

SIP:888@<your server address>