I wrote this API :
https://github.com/prp-e/iot-api-temp-humid
and when I tested it, I used this command :
curl -b cookies.txt http://localhost:8000/login/username/password
and each time I wanted to check the data in the "Enviroment" table, I use
curl -c cookies.txt http://localhost:8000/env/username
I need the cookies to be stored somewhere, or regenerate each time ESP8266 sends data to the API. is there any way?
If the cookie data is small (fewer than 4096 bytes), you might store it using the EEPROM class. Note that the ESP8266 doesn't really have an EEPROM (Arduinos generally do), so this is just writing the data to a reserved area of its flash storage. Be sure to call EEPROM.commit() after you write or your changes won't actually be saved. The EEPROM documentation includes links to some examples of how to use it.
If the cookie data is larger, you can store it in a file using SPIFFS. SPIFFS lets you use part of the ESP8266's flash storage as a simple filesystem.
ESP8266 boards usually have low quality flash storage which can only handle at most a few hundred thousand writes, so you don't want to write to the flash very frequently. For instance, if you updated the cookies in flash once per second, in just one day you'd write to the flash 86,400 times. Within two days you'd quite possibly wear out the sector of flash that was being used to store cookie values. So be careful with how often you change the values of the cookies and how often you write to the flash memory.
The ESP8266 also has 512 bytes of RAM associated with its real time clock (RTC). Data stored here will persist across reboots but will be lost if power is removed from the chip. Because it's normal RAM and not flash, it doesn't suffer from wear problems and can be rewritten safely. Here's an example of how to use it.
I'm working with 2 little machines with limited unix tools. Both are conected between each other via serial. I'm transfering binary data, so the devices are on raw mode. The sending machine is sending files to the other one and between there's a delay of X ms (specified as parameter). I would like to know if it's possible to measure those delays on destination machine in order to identify how many files are coming. Till now i was using cat < /dev/ttyS5, but this is not a option to my purpose.
Any idea?
Thanks
IMHO the easiest way is to write a little program which is waiting for bytes on the serial line.
Everytime a character arrives some sort of timer/timestamp is reset.
Another thread could be evaluation this timer/timestamp in a loop and increment a counter if it's larger than a defined value.
But please be aware that you might experience delays from the serial line as there's the kernel and its scheduler "in between". Furthermore you'll need appropriate locking of course!
Is that possible to achieve Inter process communication using any terminal or serial ports in AIX or Unix?
I would like to achieve this by using commands/scripting only where one process writes a string on terminal and another process reads same terminal and processes that string. I know that using pipe also this is possible but I do not have enough idea on that.
Also is there a way we can determine which all ports/terminals are available in AIX machine?
Or is it possible to create new terminal at run time (not the boot time) that will be used by only above two processes?
I think what you want are pty's? Or, another option would be unix domain sockets.
The answer to your first question is "no"... not really. When you write out to a tty, that output is sent out to the real device and not available to be read back.
The list of tty's on a system is: lsdev -Cctty
Creating tty's at run time is possible but not really what you want either. A tty is a child of a serial port and you can not add serial ports arbitrarily. They are real things. With AIX and Power systems, you can add devices while the system is up (hot swap) but that is getting (I'm assuming) way far off your original topic.
The basic different between a pty and a unix domain socket is a pty mimics the output and input process of a real tty in one direction. This is what telnet, rlogin, ssh, and many other daemons use when connections come in. It is easy to make ksh believe that it has a real tty by using pty's. If you don't need that, then they are added trouble that you don't need. Find a link on how to create and use a Unix domain socket and you will have what you need (or a pipe but a pipe requires a parent / child relationship which I assume you do not have).
As someone who is very new to the opensource PBX projects like Asterisk and FreeSWITCH, I am grappling with some information overload. Have read the basic FreeSWITCH docs on Wiki, but still have few questions. Since I am not very familiar with the terminology, I will try to use close approximations.
Trying to create a small/minimalistic build of FreeSWITCH, that needs to run on an rather old laptop (Celeron 1GHz, 512MB RAM, 20GB HDD, already running Debian "Wheezy"), and set it up as a 6-port GSM-SIP/Jabber gateway. So, by "small" and "minimalistic", I mean one which doesn't have modules/optional-software that is not absolutely necessary (e.g. no need for IVR announcements, or Skype integration) -- to keep memory footprint smallest, and occupy less hard-disk real-estate.
The rough idea is to have 6 GSM ports (via 'GSM-open module', similar to chan_dongle) towards public telephony network, and about 60 SIP extension, and support upto 6 calls involving GSM ports, and about 6 SIP-SIP calls (intra PBX), on this setup. I have read that the CPU overhead of GSMopen module is pretty low, so I am guessing this is possible.
Can someone confirm this to be a realistic goal?
What might be the minimum set of modules to select for minimalistic build?
For modules not chosen during initial build, can those be added later? If so, would it require me to rebuild FreeSWITCH completely, only the modules, or that everything would be built, but only configuration changes would be required to ensure that modules are loaded, and configure?
Is there any rough estimate of what might be the maximum call-rate that could be supported in such a configuration? For SIP-SIP calls? Given the underpowered processor, and little RAM (as per modern standards), I am guessing that both shall be bottlenecks, but adding RAM might still be possible (even if costly and difficult).
I have read that "hooks" can be created using Lua/Python/Java etc.. However if someone share share few examples of what-all is possible using such hooks, it would make the concept clearer. Can one hope to write an application like "missed call log" or "redirect on no answer" using these hooks?
Can someone confirm this to be a realistic goal?
Yes, this is quite realistic. You need to target as little as possible transcoding, because that's where CPU resources are needed. But even with a 1Ghz Celeron, 6 transcoded sessions seem quite realistic. But it needs testing :)
What might be the minimum set of modules to select for minimalistic build?
Just start with the default list of modules, and add gsmopen (I have no experience with gsm gateways, can't help with that part). The memory footprint is pretty low, and you may need some of those modules later.
For modules not chosen during initial build, can those be added later?
as far as I remember, Wiki describes this process. You edit modules.conf and make the specific module.
Is there any rough estimate of what might be the maximum call-rate that could be supported in such a configuration? For SIP-SIP calls? Given the underpowered processor, and little RAM (as per modern standards), I am guessing that both shall be bottlenecks, but adding RAM might still be possible (even if costly and difficult).
It really depends on complexity of your dialplan. Each context consists of a number of conditions, which are doing regexp match on channel variables. So, the more complex your dialplan is, the less CPS you get. But for a 6-channel gateway, I don't see this a problem. GSM network will be much slower than your box :)
I have read that "hooks" can be created using Lua/Python/Java etc.. However if someone share share few examples of what-all is possible using such hooks, it would make the concept clearer. Can one hope to write an application like "missed call log" or "redirect on no answer" using these hooks?
You can control every aspect of FreeSWITCH behavior with FreeSWITCH. There are even examples when the complete dialplan is re-implemented by an external program (Kazoo does that).
The simplest mode of operation is when your Lua/JS/Perl/Python script is launched from within the dialplan: then it receives a "session" object, and you can do whatever you want with the call: play sounds, bridge, forward, make a new call and bridge them together, and so on. Here in my blog there's a little practical example.
Then, you can build an external application which connects to the FS socket and monitors the events and performs actions on active calls.
Also, it can be done in the opposite direction: you run a server, and FS connects to it with its socket library.
Also, you can have an HTTP service which delivers pieces of XML configuration to FreeSWITCH, and it requests those on every call (this would be the most CPU-intensive application). This way, you can feed FS from some internal database, and build fault-tolerant systems.
I hope this helps :)
You can also find me in skype if needed.
FreeSWITCH is not really memory-hungry, and you can simply start with the default set of modules (the best is to use the prebuilt Debian packages). For example, on my 64bit machine, the FreeSWIITH process occupies only 35MB of memory.
freeswitch#vx03:~$ uname -a
Linux vx03 2.6.32-5-xen-amd64 #1 SMP Thu Nov 3 05:42:31 UTC 2011 x86_64 GNU/Linux
freeswitch#vx03:~$ ps -p 11873 v
PID TTY STAT TIME MAJFL TRS DRS RSS %MEM COMMAND
11873 ? S<l 10:29 0 0 258136 36852 2.3 /opt/freeswitch/bin/freeswitch -nc -rp -nonat -u freeswitch -g freeswitch
I will go through the rest of your questions later today
I'm trying to record a DVB-Channel with a DVB-T Tuner.
I already did much research on this topic but I don't get really "information" what to do.
Basically I'm already able to create a own Graph with the default GraphEdit, make a tune request and watch a channel. Converting the Graph to C# Code with the DirectShowLib or to C++ isn't a big problem for me.
But what I don't know, what is the right approach to record the movie. (Without decode it to mpeg / avi and so on.)
The most important parts of the graph are some tuning related filters, they connect to the demultiplexer (demux), and the demux will output a video and audio stream.
The easiest way to get the mpeg stream is putting a filter before the demux. For example a samplegrabber. There you will receive the complete transport stream as it is broadcasted. But that normally contains multiple programs which are multiplexed on the same frequency. If you only need one program, you need to filter the other programs out of the stream.
If you only need a single program, it is probably easier to directly connect the audio and video stream coming out of the demultiplexer, to a multiplexer, and write it's output to a file. You need to make sure there is no decoder or any other filter between the demux and the mux. The problem is that you need to find a directshow multiplexer, as windows does not contain a standard multiplexer. I don't know any free multiplexer.
What you also can do is write the audio and video directly to a file. (again without decoding, or anything else). Then use for example ffmpeg to join the audio and video to a single file.
C:\> ffmpeg -i input.m2v -i input.mp2 -vcodec copy -acodec copy output.mpg
You probably also need to delay the audio or video stream to get them in sync.
One addition, of course you can also use ffmpeg to convert the multi program transport stream to a single program stream.