How to set the duration of Alexa wake time? - alexa-skills-kit

I am not seeing anything in the docs (Alexa Skills Kit or Alexa Voice Service) about adjusting the time that the device waits for a command after the "wake word" (default: "Alexa") has been uttered. I was wondering if I could adjust it and how to do so, preferably programmatically.
Is it possible? How so?

This is part of the hardware device behavior. So it cannot be done via the SDKs.
The underlying service, Alexa, doesn't support the concept of a wake word. In fact, it is against the terms and conditions of people using AVS to enable hardware devices to use a wake word without explicit permission from Amazon. So it isn't the sort of thing you would expect to see available programmatically.
The same goes for the other constants that govern how the hardware reacts such as the timeout in waiting for the service to respond, how long it waits for the user to say something before a reprompt, or how long it waits after the reprompt before hanging up.

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How to acces the data from a website 50-100 times a second using raspberry Pi?

I want to fetch the data of a stock. Since the data changes very fast, is there any way to pull the data like 50-100 times a second from trading websites?
And can we implement that using a raspberry Pi 4 8gig model.
RasPi4 should be more than adequate for this task. Both the ethernet and WiFi hardware is capable of connections at these speeds. (Unless you’re running a bunch of other stuff on it.) Consider where your bottlenecks may be, likely ISP or other network traffic). Consider avoiding WiFi in favor of cat5e or cat6. Consider hanging this device off your router (edge) to keep lan traffic lower and consider QOS settings if you think this traffic may compete with other lan traffic.
This appears to be a general question with no specific platform in mind. For stocks, there are lots of platforms to choose from.
APIs for trading platforms often include a method to open a stream. Instead of a full TCP conversation for each price check, a stream tells the server to just keep on sending data. There are timeout mechanisms of course, but it is good to close that stream gracefully (It’s polite since you’re consuming server resources at a different scale. I’ve seen some financial APIs monitor and throttle stream subscribers who leave sessions open.).
For some APIs/languages you can find solid classes already built on GitHub. Although, if simply pulling and reading a stream then the platform API doc code snippets should be enough to get you going.
Be sure to find out what other overhead may be implicated. For example, if an account or API key is needed to open a stream then either a session must be opened first or the creds must be passed with the stream being opened. The API docs will say. If you’re new to this sort of thing, just be a detective and try to infer what is needed. API docs usually try to be precise and technically correct with the absolute minimum word count.
Simply checking the steam should be easy. Depending on how that steam can be handled by your code/script, it may be harder to perform logic on the stream while it is being updated. That’s usually a thread issue or a variable scope issue depending on the script/code. For what you’re doing I would consider Python or PowerShell depending on your skill-set and other design parameters.

isn't so called push really pull?

Ever since the introduction of push services in IOS I wonder how it works.
From what I found out the push function is basically an open connection that does not close serverside.
But mobiles are connecting at different points of the internet, the only way for a server to know where the mobile is connected is when the mobile tells the server where that it has changed location.
I read something about polling a connection so it stays open.
What advantage is there to manage and keep a changing open connection alive vs just checking if something is to pull?
Internally yes, push is implemented by having your phone poll for new data. The thing that makes push notifications attractive is that they are routed over Apple's service and that has many advantages.
From the phone's point of view, it only has to poll from one service, which means it can conserve bandwidth and piggyback on the normal operations of keeping a cellphone online. That means a lot less battery is used, and you can do things like set your phone to only receive push notifications every half hour, which means that 29 minutes out of every half hour you can turn off the data connection.
From the server's point of view, you no longer need to cache messages or provide quality-of-service guarantees. If you want to send a push notification to a phone that's out of range of a cell-tower for a few minutes, you may need to store the message for a considerable amount of time until the phone comes back online. Apple handles all this for you, and lets your server just be in charge of creating notifications, not storing and relaying them.

Asterisk: Originate API - Which card to use to detect busy/ringing/answer event for FXO

I want to use Originate API of Asterisk to place an outbound call on a FXO channel, for testing purpose I am using X100P card and, as expected, card is not able to detect if the number is busy/ringing or when it is answered.
I want to know which card should I use so that I can get such basic events ... I am not really interested in detailed call progress analysis for answering machine or live voice. I just need basic busy/ringing and answer events and maybe a dis-connect event.
Thanks.
The X100P cards are very old (or cheap clones). For best performance, use an analog TDM or AEX interface card from Digium, the creators of Asterisk: http://www.digium.com/en/products/analog/
Configure and tune your new card, then set busydetect=yes in chan_dahdi.conf and test for successful detection. You might also try setting progzone and turning on callprogress detection, but you also increase the likelihood of false detection of hangups and disconnected calls. Be sure to register your card -- you'll get complimentary ($0) installation assistance by email, skype, or phone.
(I'm an employee of Digium, The Asterisk Company.)

Serial Comms dies in WinXP

A bit of history: We have an application, which was originally written many years ago (1998 is the first date in PVCS but the app is about 5 years older than that as it originally was a DOS program). This application communicates with a piece of hardware via serial. When we got to Windows XP we started receiving reports of the app dying after a short time of running. It seems that the serial comms just 'died' and the app was left in a stuck state. The only way to recover from this situation was to restart the application.
The only information I can find regarding this problem was apparently the Windows Message system would miss that information was received, the buffer would fill and the system would get stuck. This snippet of information was left in a old word document, but there's no evidence to back this up. It also mentions that this is only prevalent at high baud rates (115200+).
The solution was to provide customers with USB->Serial converters along with the hardware.
Today: We are working on a new version of the hardware that will run across a network as well as serial ports. So to allow me to work on the network code, minus the actual hardware we are using a VSCOM NetCom113 device. It also installs a virtual comm port on the users (ie: mine) machine.
Now I have got the network code integrated with the app, it appears that the NetCom device exhibits the same behaviour as a physical commport. This is undesirable as I need the app to run longer than ~30 seconds.
Google turns up zero problems that we experience.
I was wondering:
Has anyone experienced this before? If so what did you do to fix/workaround the problem?
Does anyone have any suggestions as to whether the original author of the document is correct and what I can do to test the theory?
Unfortunately I can't post code as the serial code is tightly couple with the rest of the system, though if you have questions regarding it I can answer questions about it.
Updates:
The code is written using Win32 Comm routines - so I am using CreateFile, ReadFile. There's also judicious calls to GetOverlappedResult.
It's not hanging per se, it's just that the comms stops. You can access the menus, click the buttons, but nothing can interact with the connected hardware. Using realterm you can see that no data is coming in or going out.
I think the reference to the windows message is that the problem is internal to windows. Data has arrived but the kernal has missed it and thus not told the rest of the system about it.
Flow control is not used.
Writing a 'simple' test is difficult due the the fact that the code is tightly coupled and the underlying protocol is quite complex and would require a lot of work.
Are you using DOS-style serial code, or the Win32 CreateFile approach?
If the former, be very suspicious: if at all possible I'd convert to the latter.
If the latter, do you know on what kind of system call it's hanging? Are you in a blocking read call? or an overlapped I/O call? or waiting on an event? (I'm not sure I have enough experience to help, but those are the kinds of questions that come to mind)
You might also check into the queue size, which you can set with the SetupComm function.
I don't buy the "Windows Message system" stuff -- it sounds fishy; you can write good Win32 serial i/o code that never uses Windows messages.
edit: does your Overlapped I/O use events? I seem to remember something about auto-reset events occasionally missing their trigger... check your overlapped I/O calls very carefully to see whether you're handling the possible outcomes properly. Perhaps there's a way to make your code more robust by automatically cancelling the overlapped i/o and restarting another read. (I assume the problem is in the read half, not the write half?)
edit 2: A suggestion: assuming the win32 side has missed a byte or packet, and your devices are in deadlock because they're both expecting each other to respond to something, can you tweak the other side of the serial I/O to regularly send some type of "ping" packet with an incrementing counter? (and log the ping packets on the PC side; that way you can see whether you've missed any)
Are you sure you have your flow control set up correctly? DTR, RTS, etc...
-Adam
i have written apps that use usb / bluetooth serial ports and have never had an issue. with bluetooth i have seen bit rates (sustained) of 800,000 bps for long periods of time. most people don't properly implement the port.
My serial port
Not sure if this is a possibility for you, but if you could re-write the code using C#.NET you'd have access to the SerialPort class there. It might remedy your problem. I know a lot of legacy code based around the Win32 API for hardware I/O ports tended to fail in XP due to timing (had a small bit of experience with MIDI).
In addition, I don't know if you can use the Win32 method of Serial Port access in Vista, so that might shut out future MS OSes from being able to use your code.

GSM Modems, PCs, SMS and Telephone Calls

What all would be the requirements for the following scenario:
A GSM modem connected to a PC running
a web based (ASP.NET) application. In
the application the user selects a
phone number from a list of phone nos.
When he clicks on a button named the
PC should call the selected phone
number. When the person on the phone
responds he should be able to have a
conversation with the PC user.
Similarly there should be a facility
to send SMS.
Now I don't want any code listings. I just need to know what would be the requirements besides asp.net, database for storing phone numbers, and GSM modem.
Any help in terms of reference websites would be highly appreciated.
I'll pick some points of your very broad question and answer them. Note that there are other points where others may be of more help...
First, a GSM modem is probably not the way you'd want to go as they usually don't allow for concurrency. So unless you just want one user at the time to use your service, you'd probably need another solution.
Also, think about cost issues - at least where I live, providing such a service would be prohibitively expensive using a normal GSM modem and a normal contract - but this is drifting into off-topicness.
The next issue will be to get voice data from the client to the server (which will relay it to the phone system - using whatever practical means). Pure browser based functionality won't be of much help, so you would absolutely need something plugin based.
Flash may work, seeing they provide access to the microphone, but please don't ask me about the details. I've never done anything like this.
Also, privacy would be a concern. While GSM data is encrypted, the path between client and server is not per default. And even if you use SSL, you'd have to convince your users trusting you that you don't record all the conversations going on, but this too is more of a political than a coding issue.
Finally, you'd have to think of bandwidth. Voice uses a lot of it and also it requires low latency. If you use a SIP trunk, you'll need the bandwidth twice per user: Once from and to your client and once from and to the SIP trunk. Calculate with 10-64 KBit/s per user and channel.
A feasible architecture would probably be to use a SIP trunk (they optimize on using VoIP as much as possible and thus can provide much lower rates than a GSM provider generally does. Also, they allow for concurrency), an Asterisk box (http://www.asterisk.org - a free PBX), some custom made flash client and a custom made SIP client on the server.
All in all, this is quite the undertaking :-)
You'll need a GSM library. There appear to be a few of these.
e.g. http://www.wirelessdevstudio.com/eng/
Have a look at the Ekiga project at http://www.Ekiga.org.
This provides audio and or video chat between users using the standard SIP (Session Initiation Protocol) over the Internet. Like most SIP clients, it can also be used to make calls to and receive calls from the telephone network, but this requires an account with a commercial service provider (there are many, and fees are quite reasonable compared to normal phone line accounts).
Ekiga uses the open source OPAL library to implement SIP communications (OPAL has support for several VoIP and video over IP standards - see www.opalvoip.org for more info).

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