hangups | first third-party instant messaging client

I have a Raspberry Pi connected via UART to a microcontroller. The code on the RPI is trying to read incoming non-canonical UART data, but randomly receives POLLHUP. I have been able to recover by closing and reopening the file, but this is less than ideal.

Is there a way to disable the disconnect detection behavior of termios in Linux? I am not sure why the POLLHUP is being raised in the first place. I suspect that some control characters are still being interpreted despite my call to cfmakeraw(). The cable is unlikely to be the problem as canonical debug output works fine (admittedly over different pins, but same baud and same type of cable).

Sample code, setup:

i'm developing a VoIP app and having problem with background handler. What i want is: sending push with type "call" then the app will show Call UI, later if receive push type "hangup" , the app will close that call UI.

In background handler i'm using a singleton global class to inform hang up event with a StreamController and CallScreen widget will listen to that stream to close itself.

Then i found out that flutterfire will start another isolate for background handler (turn screen off) , so it will create another singleton class -> i can't close my CallScreen UI with this new singleton class.

Is it possible to do something like this with flutterfire's background isolate ?

Example pseudo code:

I have a test that says "After approx X concurrent connections, I should see socket hangups since my service will stop answering someone hammering me."

This works pretty well after a lot of painful trial and error, but because I am using await Promise.all(myrequests) the first time I see a hangup it throws a socket hang up exception.

This works, but causes some error messages, since my routine for hanging up does some debug logging, and the test is over at this point.

What's the best way to say: "wait for all of these, even when they throw errors?"

My jest/supertest problem block looks something like:

I have a .csv import from excel that has formula hangups that I am trying to remove. A simple version of the data is below.

I am looking to replace my older Asterisk 15 VoIP Server with Asterisk 17. Currently I have an application that works with Asterisk over ARI (i.e. Asterisk Rest Interface) + WS (for events). On my current version everything works fine, but after conncting my app to Asterisk 17 it did not receive ChannelHangupRequest event and the following events in time (i.e. only after ~30 seconds). So I've decided to take a look on the Asterisk logs and I've found out that:

1. The problem is not caused by my app
2. The ChannelHangupRequest event is send, as I said before, after about 30-50 seconds. The interesting part is that it's cause is equal to 1 and there is a field soft equal to true. You can see below the event in JSON format. Official Asterisk 17 documentation does not give an obvious description for it or at least it is not obvious to me: From soft: boolean (optional) - Whether the hangup request was a soft hangup request. Also, official hangup cause mappings indicate that 1 is mapped to AST_CAUSE_UNALLOCATED. On my older version (i.e. Asterisk 15), where the event fires immediately, case is equal to 16 = AST_CAUSE_NORMAL_CLEARING.

<{"cause":1,"soft":true,"type":"ChannelHangupRequest","timestamp":"2021-03-23T17:10:19.817+0000","channel":{"id":"1616519370.38","name":"PJSIP/1000-00000026","state":"Up","caller":{"name":"Cristi","number":"1000"},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"from-internal","exten":"s","priority":1,"app_name":"Stasis","app_data":"my-app"},"creationtime":"2021-03-23T17:09:30.364+0000","language":"en"},"asterisk_id":"02:42:ac:13:00:03","application":"my-app"} < {"type":"StasisEnd","timestamp":"2021-03-23T17:10:19.817+0000","channel":{"id":"1616519370.38","name":"PJSIP/1000-00000026","state":"Up","caller":{"name":"Cristi","number":"1000"},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"from-internal","exten":"s","priority":1,"app_name":"Stasis","app_data":"my-app"},"creationtime":"2021-03-23T17:09:30.364+0000","language":"en"},"asterisk_id":"02:42:ac:13:00:03","application":"my-app"}

1. Asterisk log shows that channel was disconnected for lack of audio RTP activity in 49 seconds. It explains the behaviour to some extent, specifically, it explains where the ChannelHangupRequest event comes from (after ~50 seconds), but it does not explain why isn't it sent properly immediately after user hangups.

13373[2021-03-23 17:09:28] VERBOSE[6273] stasis/app.c: Creating Stasis app 'my-app' 13374[2021-03-23 17:09:28] VERBOSE[6273] res_http_websocket.c: WebSocket connection from '172.19.0.1:54308' for protocol '' accepted using version '13' 13375[2021-03-23 17:09:29] VERBOSE[6276] stasis/app.c: Replacing Stasis app 'my-app' 13376[2021-03-23 17:09:29] VERBOSE[6276] res_http_websocket.c: WebSocket connection from '172.19.0.1:54312' for protocol '' accepted using version '13' 13377[2021-03-23 17:09:30] VERBOSE[6278] stasis/app.c: Replacing Stasis app 'my-app' 13378[2021-03-23 17:09:30] VERBOSE[6278] res_http_websocket.c: WebSocket connection from '172.19.0.1:54322' for protocol '' accepted using version '13' 13379[2021-03-23 17:09:30] VERBOSE[6280] dial.c: Called 1000 13380[2021-03-23 17:09:30] VERBOSE[1601] netsock2.c: Using SIP RTP Audio TOS bits 184 13381[2021-03-23 17:09:30] VERBOSE[1601] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field. 13382[2021-03-23 17:09:30] VERBOSE[1601] netsock2.c: Using SIP RTP Audio CoS mark 5 13383[2021-03-23 17:09:30] VERBOSE[6280] dial.c: PJSIP/1000-00000026 is ringing 13384[2021-03-23 17:09:30] VERBOSE[6280] dial.c: PJSIP/1000-00000026 is ringing 13385[2021-03-23 17:09:44] VERBOSE[6276] res_http_websocket.c: WebSocket connection from '172.19.0.1:54312' closed 13386[2021-03-23 17:09:45] VERBOSE[6282] stasis/app.c: Replacing Stasis app 'my-app' 13387[2021-03-23 17:09:45] VERBOSE[6282] res_http_websocket.c: WebSocket connection from '172.19.0.1:54334' for protocol '' accepted using version '13' 13388[2021-03-23 17:09:49] VERBOSE[1601] res_rtp_asterisk.c: 0x7f8c1c02bf90 -- Strict RTP learning after remote address set to: 192.168.10.172:10096 13389[2021-03-23 17:09:49] VERBOSE[6280] dial.c: PJSIP/1000-00000026 answered 13390[2021-03-23 17:09:49] WARNING[1601] channel.c: Unable to find a codec translation path: (ulaw) -> (g723) 13391[2021-03-23 17:09:49] WARNING[1601] channel.c: Unable to find a codec translation path: (g723) -> (ulaw) 13392[2021-03-23 17:09:49] VERBOSE[6280] ari/resource_channels.c: Launching Stasis(my-app) on PJSIP/1000-00000026 13393[2021-03-23 17:10:19] NOTICE[1108] res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/1000-00000026' for lack of audio RTP activity in 49 seconds

1. The above described behaviour applies only when user answers and, then, drops. If the user rejects call without answering, everything works fine.

One solution would be to reduce the RTP inactivity timeout as a workaround, but this wouldn't be preferable. Do you have any suggestions?

I am trying to find a way of leaving my python program running continuously, but ensuring that the main script is only 'active' during nightime hours (e.g. between 21:30 and 04:30 for example)

Background: My program records motion activated video & activates a floodlight.

The main program segment is below:

...I am currently running this each evening and then killing in the morning with Ctrl-C..!

I tried to replicate the same example given in the following question.