Analog remix part 2: Tropo

Tropo, a project of Voxeo Corporation, adds voice and short-message-communications functions to PHP, JavaScript, Ruby, Python, and Groovy through a set of classes. They host the platform and your scripts, which are then triggered through the web or by PSTN/SIP, SMS, Twitter, or chat events. You can use the system as long as you want for free for development, or pay reasonable hosting rates to go production. Oh, and their support team is terrific, even for developers who haven’t yet paid them a dime.
So what can you do with it?
  • Interactive voice-response (IVR) systems
  • Chat or Twitter robot
  • Dialer
  • Call routing based on logic in the script
  • Text-to-speech
  • Speech-to-text
  • Audio recording
  • Whatever you can normally do with the aforementioned scripting languages, but note that the stdout will be a communications outlet such as a SIP channel, a Twitter feed or conversation, a chat, or an SMS–not a web page
My first Tropo application, the “local operator,” is the topic of part two of the analog remix. The platform has so many features, though, that I look forward to spending more time developing in it.
I chose to use PHP because it’s the language with which I am most familiar and also has Curl libraries built in, which I am using to make a CGI callback to my own Asterisk server. The purpose should become clear when you read the code, but you may ask, “why not just send a SIP REFER to do the call transfer?” The answer is that you can’t with the Tropo platform. All transfers go through Tropo; there’s no option for a REFER. So if you want to use your own dialplan and not have your call going out to Tropo and back to use local resources, you have to do the transfer through the Asterisk Manager, thus the CGI call.
Number, Please?
The script answers the call (which will come in by SIP URI), asks, “Number, please?”, validates and confirms the spoken number with the caller, then uses the Curl functions to go back to the calling Asterisk server (running a web server) and transfer the call.
Getting There From Asterisk
Once you’ve written a script and uploaded it to  Tropo (or typed it directly into Tropo’s file-editing interface), you go to the Applications section and choose your script as the source. Once you’ve submitted it, you’ll be given a SIP URI (amongst other methods) by which to access the application. So as part of a FreePBX Custom Extension or in your extensions.conf file, you can make an extension for your new application by referencing SIP/ I set mine up as extension 1100, which I mentioned in the previous post.
Now, if I go off hook from one of my internal extensions and dial 1100, or simply pick up the analog extension on which I configured the Warm Line/Hot Line, I’ve got the operator ready to take my number and transfer my call. Only one piece left after this: the Asterisk Manager CGI script.

Appendix: the PHP code (Number, Please?)


// Main

// SSML tags needed for advanced text-to-speech manipulation
$ssml_start = "<?xml version='1.0'?><speak>";
$sayas_digits ="<say-as interpret-as='vxml:digits'>";
$ssml_end = "</speak>";


// These IDs are used to identify the caller and the Asterisk box
$callid = $currentCall->getHeader('x-sbc-call-id');
$hostid = trim((strstr($callid, '@')),'@');

_log("Call-ID is $callid");
_log("Host is $hostid");


$haveNumber = false;

while (!$haveNumber) {
$number = ask($ssml_start . "Number <emphasis level='reduced'>please?</emphasis>" .
array(voice => "allison",
attempts => 3,
bargein => false,
choices => "[4-16 DIGITS]",
minConfidence => 0.4,
timeout => 8,
onBadChoice => "handlerBadChoice",
onError => "handlerError",
onHangup => "handlerHangup",
onTimeout => "handlerTimeout"

// Confirm the number with the caller
// 1 and 0 are options for testing with DTMF
$yesno = ask($ssml_start . $sayas_digits . $number->value . $sayas_end .
", is this correct?" . $ssml_end,
array(voice => "allison",
attempts => 2,
bargein => false,
choices => "yes, no, 1, 0",
minConfidence => 0.5,
timeout => 3,
onBadChoice => "handlerBadChoice",
onError => "handlerError",
onHangup => "handlerHangup",
onTimeout => "handlerTimeout"

_log("Caller said $yesno->value.");

if ($yesno->value == "yes" || $yesno->value == "1") {
$haveNumber = true;
} else {
say("OK, let's try again.");

// The CGI callback. Refer to a future posting for the design
// Customize for your server. Use SSL for security if you can.
$url = "http://$hostid/tropo.cgi?f=xfer&callid=" . urlencode($callid) .
"&dest=" . $number->value;
$ch = curl_init();
curl_setopt($ch, CURLOPT_URL, $url);
curl_setopt($ch, CURLOPT_HEADER, 0);
curl_setopt($ch, CURLOPT_RETURNTRANSFER, 1);
curl_setopt($ch, CURLOPT_TIMEOUT, 10);

_log("Received from CGI: " . $result = curl_exec($ch));

if (!$result) {

// If the transfer didn't happen, the caller is still here...

if ($currentCall->isActive()) {
_log("Transfer failed and caller still on the line.");
say("I'm sorry, something went wrong. Please try again later.");


// Function definitions

function handlerBadChoice($event) {
say("I'm sorry, I didn't understand you.");
if($event->attempt >= 3) {

function handlerError($event) {
_log("Error occurred: $event->value");
say("I'm sorry, an error occurred. Goodbye");

function handlerHangup($event) {
_log("Caller hung up.");

function handlerTimeout($event) {
say("I'm sorry, I didn't hear you.");
if($event->attempt >= 3) {