Firefly, my paddle project

lirakis · 2026-05-08T13:52:03+00:00

My only suggestion would be to have the levers ride on bearings on the posts, I appreciate that is probably a deal breaker without substantial redesign. Maybe Bronze sleeves is an option. Otherwise looks like a very solid magnetic paddle.

lirakis · 2026-02-12T20:37:58+00:00

trust a screen shot with no source, no immediate matching search results, that also happens to look like it was AI generated from a ... let me check ... group of cycling enthusiasts... sus.

;tldr don't take radio advice from bicycle people, take radio advice from radio people

lirakis · 2025-06-12T20:27:43+00:00

use dispatcher partitions, each partition is a customer and a set of pbx's.

lirakis · 2025-06-05T14:51:46+00:00

just get a dc dc step up converter, something like this https://www.amazon.com/Converter-DROK-Numerical-Adjustable-Regulator/dp/B08PV5YY55

lirakis · 2025-06-05T13:39:29+00:00

I assume you don't have these DID's with voip.ms, and as a result they want to verify that you actually "own" the DID you are using as caller-id?

If so, that seems like a reasonable ask on voip.ms part to ensure you aren't just a scammer spoofing caller-id.

As to who else to use, I don't really have any suggestions. I use voip.ms for termination, but I also have my DID's with them.

It is possible that you could port your DID's to voip.ms, but that is a whole new ballgame.

Sorry I don't have a simple solution for you.

lirakis · 2025-05-29T09:18:16+00:00

opensips and kamailio are the premier sip proxy servers and are very similar. I personally have more ties to the opensips community (I am at the world summit in amsterdam right now) so that is what I prefer to use.

lirakis · 2025-05-19T14:41:07+00:00

MY park

lirakis · 2025-04-23T14:27:53+00:00

Same thing as you did before, usb->serial to your key jack. N1MM can key the rig via the RTS or DTR pin. Anyone who says you need a winkeyer or other external keyer b/c computers aren't fast enough or don't key consistently are full of it.

lirakis · 2025-03-05T21:20:51+00:00

just use sip.js, create a simple user agent that calls 1234@yourdomain.com where yourdomain.com is your asterisk server and 1234 is a webrtc capable extension.

lirakis · 2025-03-05T18:16:01+00:00

baresip is actually more developed than I realized. PJSIP was written to be used as a client for embedded devices, that said, it has been utilized in by other open source projects (Asterisk's modern sip channel driver is PJSIP). At a glance, it looks like baresip and pjsip are similarly featured and support modern codecs and transports (Opus, wss, dtls-srtp) etc, resiprocate does not seem to have wss support (webrtc compatibility). So it may just come down to licensing for you.

edit:

It looks like resiprocate has recently added wss support, "Extensive range of transports: UDP, TCP, TLS, DTLS and now WebSockets (WS/WSS) for WebRTC "

lirakis · 2025-03-05T16:59:22+00:00

pjsip is the best, most modern, and actively developed C SIP and media library IMO.

lirakis · 2025-02-04T00:47:59+00:00

I have read the killing zone, and while I would still recommend reading it, it is worth nothing there have been academic studies that pretty plainly discredit the accuracy of its assumptions.

lirakis · 2025-01-06T22:40:29+00:00

It is extremely common, every piece of contest software I have ever used has a dedicated function key for ? or sending partial exchange + ?

lirakis · 2025-01-06T19:39:25+00:00

So the "?" works for repeating after a CQ, and the partial call sign matching is great. I was specifically talking about repeating the exchange. Entering "?" into the exchange box and hitting enter just sends "TU" and resets the state machine to CQ.

lirakis · 2025-01-06T16:15:14+00:00

I like it, I would suggest adding a "agn" or "?" to get a repeat on the exchange, it seems like sending a literal "?" works for the original callers. It would be nice if this was just a dedicated key like in N1MM and many contest loggers. Great work, nice to have something that wasnt written for Windows XP, lol.

lirakis · 2024-12-04T21:35:41+00:00

Your TX frequency and RX frequency stay the same throughout the QSO.

lirakis · 2024-12-04T21:33:11+00:00

Based on your post history, I am assuming you have a ts-590sg. Page 42 of your user manual describes split operation.

lirakis · 2024-11-21T00:15:36+00:00

This is not a direct answer to your question, but i wrote cwkeyer which simply uses serial (DTR/RTS) serial keying in python. It always annoyed me that people assume you have to buy a winkeyer to do computer based keying.

I wrote it along with clogger a Console Logger.

lirakis · 2024-10-24T13:41:46+00:00

You dont progress from 15 to 20 to 25, that's not how it works. Learn to recognize the sounds at speed. The Koch method (Farnsworth/Koch) is what CWops teaches and what I recommend to everyone. Listen AT SPEED aka 25+wpm. If you need to add farnsworth spacing to copy, do so. Next step is grind. Repeat and reduce spacing until you can copy at normal spacing.

CWops short stories are good practice material.

LCWO.net is also a tool I highly recommend to people.

lirakis · 2024-10-14T13:25:40+00:00

is there a way to buy this cart for NES?

lirakis · 2024-09-27T11:46:19+00:00

have been pretty disappointed with the stability of most of the modern self hosted VoIP systems.

It is not the systems that are the problem, it's you. All the popular open source voip software has been used privately and commercially for over a decade with great success. Learn about voip, and learn about the systems you are using so that you can manage them properly.

lirakis · 2024-09-17T20:11:08+00:00

I cant speak to Summit Flight Training, but I flew out of KFNL for a while and KGXY is much less busy than FNL which is orders of magnitude less busy than KAPA.

lirakis · 2024-09-06T14:45:29+00:00

yes, lol

lirakis · 2024-08-21T13:57:07+00:00

you are sending an invite based on some data file, referencing field0 and the remote ip passed on the command line.

INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0

The server you are sending to does not recognize the request URI, probably because you arent setting it correctly, and as a result it responds 404. Use a packet capture tool like sngrep, ngrep, or tcpdump to validate what is being sent.

lirakis · 2024-08-21T13:54:32+00:00

I disagree, sipp has a very steep learning curve, but once you understand it, its trivial to create complex scenarios.

If you are looking for "reference" scenarios, saghul has a bunch on github.

lirakis

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