The new VoIP Drupal Beta is out!!

leoburd's picture

Hello all,

After over 5 months in the making, we are proud to announce the release of the new VoIP Drupal betas: 6.x-1.0-beta14 and 7.x-1.0-beta3.

This release brings important new improvements to the VoIP Drupal platform, including:

  • Phone number standardization. Different VoIP providers format phone numbers in different ways, which makes it practically impossible to compare calls with one another. Starting in this version, VoIP Drupal expects call source and destination numbers to be prefixed with either a "+" and country code (ITU E.164 standard), as in "+1 617 222 3333" for US numbers, or with a "sip:" (for SIP numbers). While writing numbers without the recommended formatting might work in same cases, that's not advisable and might generate unexpected side effects.
    ** NOTE: VoIP Number 2.0 ( provides a common API for handling area codes and other features associated with the new number format adopted by VoIP Drupal

  • Advanced SMS features. The new release extends the already-existing VoIP Drupal texting capabilities in many ways. By going to "VoIP Drupal -> Default call configuration"(admin/voip/call/settings) and selecting "Advanced text message options" you will find options to:
    ** Accept incoming text messages from the SMS framework
    ** Use the SMS framework for outgoing text messages
    ** Reduce text message size and server issues by converting special Latin characters (ç, ã, é, etc.) to their ASCII equivalent (c, a, e, etc.)
    ** Define a special "text message handler" function to preprocess incoming SMS whenever they arrive. For instance, you might use a function such as voipcall_join_text_messages() (available in modules/voipcall/ to join multiple SMS message segments back into one single message before sending it to be processed by a call script.

  • Advanced hang up processing. Now it is possible to define a special "hang up handler" function to finish processing call elements as soon as a call is hang up. Have a look at the "voipscript_record" script in voipscriptsamples.module for an example in which a caller might accept an ongoing recording simply put hanging up the call.

  • A new voip_debug() function -- and associated "voip_debug" variable -- to allow admins to control the output of debugging strings generated by the VoIP Drupal framework.

  • Small bug fixes

We are very interested in your feedback and suggestions. If you have any questions, please let us know by posting them in the VoIP Drupal discussion group:

== The VoIP Drupal team



freescholar's picture

I am installing the D7 version on 3 new sites and I look forward to the new improvements. I am particularly interested in the SMS improvements... Even though VoipDrupal already totally rocked, I am sure it rocks harder now!
Thanks Leo:)

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Can't Find Callhandler File For Plivo

bdrhoa's picture

I'm trying to get voip drupal working with freeswitch via plivo.
The README.txt says to set DEFAULT_ANSWER_URL =
But there is no callhandler file or directory that I can find.

Also, I registered fro the group about a week ago, but haven't been approved yet.


Re: Can't Find Callhandler File For Plivo

leoburd's picture

Hello Brad,

What do you mean by "there is no file or directory" that you could find? What happens when you type in that URL (with your own site id) on the browser?



404 Page not found The

bdrhoa's picture


Page not found

The requested page "/voip/plivo/callhandler/" could not be found.

Are the docs just giving an example? There is nothing in my entire filesystem named callhandler. Am I really supposed to be pointing to plivohelper.php?