Configuration file: vxml.conf

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 “;”








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.

General Section [general]

The following section covers general functions of VoiceXML Asterisk module.

Key Values Default Purpose
msqkey*decimal number1973 Message queue ID.
msqfactor*value in Kbytes10Size by VoiceXML channel allocated for the message queue.
msqfirsttimeout*value in milisecondes100First timeout, the next ones are set to 0.5s, setting lower value can increase the reactivity of the browser.
msqtimeout*value in secondes30Timeout when the module not receive commands from the VoiceXML browser (can stop the Asterisk if autoexit set to yes).
msqsize*auto or value in KbytesautoSize of the message queue or if auto msqfactor x number of VoiceXML channels.
msqlock*yes/nonoForce to wait and disable the timeouts.
debugyes/no/fullnoEnable or disable the logs from the Asterisk module and the VoiceXML Browser.
paramseparator*character,Separator for Asterisk application parameters.
systemnameauto/value string name(empty)Empty not change the systemname (used in the uniqueID for the CDR), auto use the hostename of the server or set a string to define the system name.
autoanswer yes/no/prompt/recognize/ringing/yesyes, answer the call with the vxml() execution, no never answer the call you need to call the Answer() application from the VoiceXML, prompt answer before first prompt/audio play, recognize answer before first recognize action, ringing no answer send a single audio frame to cancel the ringing tone.
autohangup yes/no/soft/exitnoyes execute hangup() a the end of the vxml() execution, soft execute softhangup(), exit execute hangup() and set the variable VXML_EXIT with the VoiceXML exit value.
autoexit yes/no no Stop the Asterisk with return code -1, if VXI interpreter no answer (msq timeouts).
monitor yes/no/old/1..100 no yes execute mixmonitor() (or monitor() if old) before starting the VoiceXML session. You can set a random value (using a %).
monitordirectory directorymixmonitor directorySet the mixmonitor() monitor() directory or use the default one (/var/lib/asterisk/monitor).
monitorformat wav/gsm/pcm/ulaw/alawwavSet the format for the mixmonitor recording.
defaulttimeout value in milisecondes5000Default timeout if not set in the VoiceXML syntax.
defaultinterdigittimeout value in milisecondes 3000Default interdigit timeout if not set in the VoiceXML syntax.
priorityeventsyes/noyesIf using single DTMF events and DTMF inputs/grammars, yes force to check the first DTMF to the DTMF events as soon as it is receive, no will wait full entring or a timeout.
tdd*yes/nonoyes enable TDD () to send and receive text messages.
keepcontext*yes/noyesyes force to keep the VoiceXML channel during the consultation transfer.
videoyes/nonorefer to Lience options
speechyes/no/automatic/emulationnorefer to Lience options
videoyes/nonorefer to Lience options
dialer refer to Lience options
rtmp refer to Lience options
voicexml refer to Lience options
messaging*yes/no/database noyes allows to control the messaging counters as local context value. database use the Asterisk internal database to store the counters values.
priority*value priority0if set us the priority at the end od the vxml() execution.
removeprompts*yes/noyesyes allows remove the prompts after their play or processing. no is for debugging.
autoreload yes/nonoyes Can for to automatically reload the configuration if the configuration file date has changed.
recordsendaudioyes/nonoyes force the VoiceXML module to send audio silence during the recording. Some operators can close the call is there is no audio traffic during the recording.
recordsendvideo yes/nonoyes forec the VoiceXML module to send back to the user its own video during the video recording (as an echo test).
recordsilence yes/no/value milisecondsyes(300)yes allows to record this time before the VAD detection. no force to record all the audio without waiting for the voice detection. You can set a value in milisecondes.
recordtotext*urlemptyIf record type=“text/plain”, use this URL to convert speech to text (as Google)
recorddirectory directory/tmp/cacheContentDefault temporary directory for recording
recordrewind yes/noyes Strip off the last 1/4 second of the audio records.
threshold dsp/default/number default Set the default threshold for silence detection.
videoupdate yes/no yes Request a video update before recording.
audiosilence* no/file no Play an audio file during silences.
cachedirectory directory /tmp/cacheContent Cache directory for audio memory, MRCPsynth, record, builtin grammars…
ttsdirectory directory /tmp/cacheContent/ TTS cache directory, used to purge old files (greater the maxage parameter).
videosilence* no/file no Play an video file during silences.
wavcodec pcm/gsm Codec use for record
videocodec* h263/h263p Deprecated
dialformat string %s Format string used to generate outgoing calls (allows to set the channel type and peer for example)
dialformatvideo string %s Format string used to generate outgoing video calls (allows to set the channel type and peer for example)
dialnumbersonly yes/no no Filter to suppress non numeric charaters
nomaskerade yes/no no For video calls add /nj for video calls
videoprofile string empty Set the variable VXML_VIDEO_CODEC used by the defaukt transcoder dialplan
blindapplication transfer/ none To generate a SIP REFER, set a <transfer bridge=“false” …/> and this parameter to “transfer”
localformat spawn/bridge/nooptimization/jitter spawn Set the mode to transfer to local the dialplan
localextension string vxml Set the default extension for local transfers
conferenceformat string empty Specific parameter for the conference application.
celudanurl* deprecated
speechprovider unimrcp/verbio/lumenvox/string empty Set the Speech module (ASR) provider.
speechdirectory directory /tmp/cacheContent Directory to generate temporary grammars (buitins for example).
speechscore 1-100 20 Minimal score to forward the speech result o the interpreter.
speechfactor -x/+x 0 Parameter to multiply or divise the result from the speech engine.
speechforcescore* 0-100 0 Force a speech score (debug only)
speechforceresult* string empty Force a speech value (debug only)
speechforcename* yes/no no Add a 'V' for the grammar name, only for old Vestec engines
speechhotwordscore 0-100 0 If score is lower than this value, the module restart the recognition
speechbargein yes/no/forced/enhanced no Force the prompt bargein mode for the speech.
speechmaxresults number 1 Max results returned when it use Asterisk speech module.
speechconcatenate yes/no no Concatenate multiple speech results for Verbio.
speechrecord yes/no no Record Speech sessions (wav) and results (txt) (for debug)
speechrecorddirectory directory /tmp/cacheContent Directory to generate temporary speech recordings.
speechunload yes/no yes Disable the Speech unload (some ASR engine not support to set empty grammars contents to free the loaded grammars.
speechurl* url empty not implemented
speechbeepstart* file empty not implemented
speechbeepstop* file empty not implemented
cli yes/no no Allows to execute CLI commands form the VoiceXML syntax (audio src=“cli:command”)
urldecode yes/no no Decode VoiceXML first URL.
cdrupdate yes/no no Allows to modify all the CDR fields
cdrexit yes/no no Use the VoiceXML exit value to set the CDR userfiled
cdrdial yes/no no Generate a CDR with the transfered call.
cdroverwrite yes/no no Overwrite the local/remote call numbers (with the values managed the VoiceXML VXML_LOCAL, VXML_REMOTE…)
cdrparam yes/no no Use the VXML_PARAM to set the CDR userfield
cachetimeout number 60 Execute the cache process each x seconds.
cachehour number -1 Execute the cache process at a definided hour.
cachemaxage number -1 Remove cache file older than maxage seconds.
cachemaxremoves number -1 Max removed files for each execution process.
loadlibrary* library Deprecated
loadcli string empty Execute a specific CLI command when the module is loaded.
originatedelay number 10000 Time to wait before dial in miliseconds.
originatedelay2 number 10000 Time to wait before dial the second time n miliseconds.
requesturl* URL empty URL for the billing HTTP/API.
countermax* number Deprecated
countermin* number Deprecated
counterval* number Deprecated
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…)

License Section [license]

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.




Key Values Default Purpose
maxnumber of channels1Max number of VoiceXML channels for this license.
texttospeechyes/noyesAllows to use Text to Speech prompts.
video yes/nonoAllows to use Video options.
speech yes/nonoAllows to use ASR/Speech options.
dialer yes/nonoAllows to use the VoiceXML with external Dialer.
voicexmlyes/noyesA allows to use the VoiceXML browser execution.
expirationdate YYMMDD Optional expiration date of the license
keyLicense key Key after purchasing a official commercial license. If not set or wrong key, you have 1 trial VoiceXML channel.

Control Section [control]

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.

Key Values Default Purpose
forward dtmf # Fast-forward when this DTMF digit is received.
reverse dtmf empty Rewind when this DTMF digit is received.
stop dtmf 0123456789 Stop when this DTMF digit is received.
pause dtmf empty Pause playback when this DTMF digit is received.
restart dtmf empty Restart playback when this DTMF digit is received.
skipms number 5000 This is the number of milliseconds to skip when rewinding or fast-forwarding.

Account Section [account_]

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:





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.

Key Values Default Purpose
name string The name or designation of the account. This reference can be used to identify the account used by the VXML application, Vxml(name of the account).
number number/expression The identification of an account with the called number, signaling information from the ANI function, CALLERID(). You can use the “*”, the wildcard character, to specify a substring such as *03 or 014612* , that the number function will contain. You can start the number with “_” to use Asterisk Diaplan Patterns too (see : function can also be used to identify the account, like the name, Vxml(number of the account). You can start the number with “@” to match the caller number to the account number (use the same syntax as the called number There is a way to )
url URL The VoiceXML URL of the account.
urlvideo URL The VoiceXML video URL of the account.
max number max The maximum number of sessions allowed to this account. If there are not enough sessions then the VoiceXML application will generate an error.
dialformat string empty This is similar to the general function, but for the account only. If not set, use the general value.
dialformatvideo string empty The same purpose as the dialformat, only for video sessions.
dialnumbersonly yes/no no
force yes/nno/video/audio/ulaw/alaw/flash/ringing/vxml/h263/h263p/h264 If set to video, set the Transfercapability to VIDEO (and enable the h324m processing). If set to audio, the Vxml application execute directly the account URL and bypass the redirection execution (parameter with '@').
speech yes/no/emulation/automatic/debug general As the general function, but for the account only. If not set, use the general value.
speechprovider string general As the general function, but for the account only. If not set, use the general value.
speechscore As the general function, but for the account only. If not set, use the general value.
mark string empty Set a string mark in the VoiceXML browser traces. The session ID and this string will be added to the channel number column (3rd) in the traces (Example : …33… → …33_1_user1… ). Four redirection exist : @remote : @remote : caller number, local : called number, @id : VoieXML id parameter value, @param : VoiceXML parameter value
durationlimit number 0 The maximum duration in seconds for the session using this account. The call will be hangup after this maximal duration.

LogsSection [logs] (deprecated)

Key Values Default Purpose

Configuration example


; VoiceXML Configuration




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:


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


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="">
  <block><audio src="hello.wav"/></block>

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>
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