stasis | Static sites | Static Site Generator library

I have an object full of blueprints. The blueprint is an array of either another blueprint or and ingredient. I'm trying to show a path to build a 'Stasis Device' but the output is missing some steps. The output displays:

Hi I want insert break line tag in javascript and tried everything from stackowerflow,but nothing helped me. Here is my github repo for.js file and I need make breakline between line 204 and 205 like this . I mean need third section under second one ,check it here turashviliguro.github.io/d2symbols I will be happy if you help me.

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 wrote a function to slide a div into the viewport horizontally in the middle of a vertically scrolling page and it mostly works. It slips when you scroll through the page quickly, though, and doesn't always finish scrolling in, horizontally. If you're scrolling fast, it sometimes stops short of the left-most edge of the browser and I'm at a bit of a loss to explain why that is.

Any takers? Codepen here: https://codepen.io/ThatWerewolfTho/pen/xxRZERv

HTML

Getting this error.

asterisk stasis app not active

Dialplan:

I am trying to wrap my head around ARI and Asterisk, my goal is to dial from an extension to another. I dialed 5001 from extension 5002. Now in the stasisStart function, I want to create a new channel, and used the dialed number (5001) and pass 'PJSIP/5001' to the endpoint. How do I get the dialed number?

Dialplan:

I've been able to setup Asterisk to log CDRs to a MySQL database using the ODBC option. The challenge I am currently facing is that only calls with the disposition ANSWERED are logged. NO ANSWER, BUSY and other calls are not logged in the database though I see the status from the logs.

I place the calls using ARI which connects to a stasis app when the call is answered.

How do I ensure asterisk logs all calls to the database, irrespective of the call status.

I am using Asterisk 16.2.1 and added a additional field to the cdr table.

I'll preface this by saying I'm a complete novice to VBA but not coding. I'm trying to write part of a VBA script that can copy data from one worksheet in another workbook to a pre-existing worksheet in another workbook.

I'm trying to make use of dynamic ranges as I know the starting point for the data every time however my issues comes when it comes to the copying. The code runs up to the point of the copy where it stagnates with no error codes. Effectively gets stuck in stasis requiring user action.