I have a local installation of Voxeo's Prophecy platform, and a voice xml application that runs on the voicexml browser of the platform. How can i call the application to listen to the message, without having a sip phone, and without using the voxeo's hosting capability? I just want to call the application at the local installation without using a phone. When i try this from the browser, i just get the xml file containing the dialog.
Why are you trying to start the application without a SIP phone? The developer version of Prophecy comes with a SIP phone that works great for testing and debugging. Are you trying to access the application from a regular telephone or POTS. If that is the case you will need additional hardware, such as a Cisco VoIP Gateway to translate the land-line signal to SIP. There are a number of VoIP Gateways available. Another option is to put Dialogic cards in your server and use a software that translates between SIP and the Dialogic API. If you purchase Prophecy from Voxeo as an appliance there are options they can sell you to achieve this. The advantage of getting this from Voxeo is that they will help configure it, which is not trivial depending upon the type type of telephone lines you are using and whether they are behind a PBX or not.
The telephony/communications interface to Prophecy is SIP so that is the only way to communicate with it. You could use any open source SIP stacks to develop something yourself, but the easiest thing to do is to use a software based SIP phone. It is not like you could just start talking and the VXML app would know what you intended, or if the if the voice input was really intended for the VXML app. You still need to maintain things like sessions and routing to the correct application which is easily handled by a SIP phone. Prophecy is designed for a telephony environment. If you are trying to use it for something like a home automation system, which some people have, I would think you would have to provide some type of SIP front end that is voice activated.
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I want to load test my application and I'm deciding b/w choosing a load testing service or simulating the clients on my own.
My question is whether I can accomplish this user scenario using any of the load testing tools available.
My test will have two types of users, A & B. Communication would be between A and B. A-A communication and B-B communication is not there.
I want Users to login to my application through mobile or web. Either type of users could log in through either type of platform.
Make multiple GET/PUT/POST requests to my application server with specific parameters.
Be able to exchange video or text messages by integrating through services like Pusher and Vidyo.
'3' is probably the most important requirement for me, and I'm wondering if I could simulate the dummy users so that it listens to a pusher channel and can open up a Vidyo iframe in the emulated browser / mobile device for a specific pusher message?
I got on call with Blazemeter and turns out they or anyone in the market doesn't support plugins on emulated browsers. I finally simulated the clients on my own replicating all the server calls a typical browser or mobile device makes for my application.
I haven't yet been able to load test Vidyo and don't know if there's a way to do it except by opening up conferences manually.
I was able to load test services like Pusher and SQS successfully.
I am working in a project where we need to setup an IVRS Application over FreePBX. Idea is End user will dial an extension from the SIP Phone to access the IVRS Application hosted through TTS Engine and Tomcat. So i need a call to be forwarded to the VXML server through FreePBX and continue with application over voice through SIP Phone. It can be done with Voxeo by URL Mapping once it is set to particular extension.
So looking for a Integration trick between FreePBX + Asterisk and IVRS Application to setup the particular inward call routing to the Application. In other word, how to register an IVRS Application to the FreePBX with Extension so that End User will dial the extension and gets connected with that IVRS Application and continue until hang up.Or Please let me know if i m missing the working concept of the FreePBX and Asterisk.
Thanks
You have consult your TTS server documentation for interfaces it support.
For examples you can read app_swift.c code, designed for Cepstral TTS.
https://github.com/darrensessions/app_swift
I am developing metro style app which should read data from external device using serial communication but unfortunately metro apps does not support "Serial and parallel port API ". So i thought to use hyperterminal to read data from serial port and use it in my metro application. Is there any way to access data directly from hyperterminal using c# in metro style app? If no is there any other way to access data from serial port in metro style app?
You cannot make this work with Hyperterminal either. The sandbox in which Store apps operate explicitly defeats any attempt a Store app might make to use a inter-process communication mechanism.
Do ask yourself if it makes sense to publish such an application in the Store. The odds that a user that visits the store will have the required hardware to make your app useful are extremely slim. These kind of apps will stay desktop apps for a long time to come. If the manufacturer of the device sees a market opportunity to get his device operating in store apps then he'll create a Store compatible driver for it. But that's up to them, you cannot do this yourself.
I need to implement a Notification Application in Mitel IP Phones (mainly 5330 and 5360). Where can I start? I tried to have a look at the applications samples provided by the Mitel HTML Toolkit but there is no sample for the Notification App since the web server is needed.
Any guidance (tutorial, advice, sample, etc.) you could give me will be greatly appreciated.
Depends on if you're trying to do a popup on the phone, or on a computer client. The HTML Toolkit is used typically to load apps on the phones, but there's some issues depending on what phone system the 5330 and the 5360 are connected to, the 3300 for instances has better support for apps on the phones, whereas the 5000 cp does not. The phones can be used in SIP mode which allows you the ability to do html redirect and apps easier, but then you have to pay for the sip licenses, and loose the minet functionality. An alternative is to use the UCA client SDK, it notifies a computer of events, e.g. ringing, hang up, dialed etc. and you can then use those in dot net to integrate to other sw. Hope this helps.
I'm looking at developing a VoIP application running on Sailfin (https://sailfin.dev.java.net/).
I have the server up and running and a sample app installed. What I want to be able to do is:
Receive calls from my landline in my VoIP application.
Make calls from the application to other numbers in the PSTN.
The part I'm not sure about is what hardware do I need between the landline jack and my dev box?
Thanks!
You need to look at Asterisk and hardware used to support it.