Using Tropo, it’s incredibly easy to add voice applications to your sip:provider CE (SPCE) or PRO system. Let’s assume that we have an SPCE installation serving a subscriber sip:email@example.com, which is mapped to the phone number +4312345. By either calling +4312345 from your mobile or fixed line or calling sip:firstname.lastname@example.org from a SIP client, you’d like to have a voice application answer the call and serve the caller (e.g. like an auto attendant, or a voice recorder, or whatever).
AGI and DSM madness
If you wanted to develop such voice applications in the past, you usually fired up an editor and digged deep into the Asterisk AGI or SEMS DSM docs for days even for the most simple applications, especially if you were new to these things. You needed to reconfigure Asterisk or SEMS to serve your application, change the SIP routing to use it, and develop the application using complicated syntax or APIs. To sum it up: it was difficult, time consuming and frustrating for beginners.
Tropo to the rescue
By using your SPCE platform together with Tropo, you can jump right into building your voice applications using nothing but your browser. From start to testing your first simple app, it takes less then 5 minutes. And the good thing is – you can connect to Tropo via SIP, so it’s actually free! So here’s how you do it in 8 simple steps:
1. Sign up for a free account on http://www.tropo.com
2. Create a new application:
3. Choose “Tropo Scripting”:
4. Name your application, then “create a new hosted file for this application”:
5. Choose a file name for your script, and provide your application logic. For this test, I copy/pasted an example script taken from https://www.tropo.com/docs/scripting/asking_digits.htm . Then click “Create File”:
6. Click “Create Application”:
7. Copy the SIP URI shown in “SIP Voice”, we’ll need that in the next step:
8. Finally, log into your sip:provider Admin Panel and edit the Preferences for your subscriber, in our case email@example.com:
In this case, we set a simple Call-Forward-Unconditional to the SIP URI provided by Tropo, so every call to either +4312345 or sip:firstname.lastname@example.org will end up in your voice application. You can also set a Call-Forward-Busy or No-Answer, if you like your application to kick in only in these scenarios.
I guess it can’t get much easier than that.