Digital Communications and Technology Net

The Digital Communications Net is an informal net that meets every Tuesday evening at 8:00pm on the SBARC Hub repeater network.

JT65 operating on 10m

JT65 operating on 10m

We discuss and operate all kinds of digital communications and special modes such as 2m/440 Yaesu System Fusion digital radios, mesh networking, as well as HF oriented modes like PSK31, JT65, FreeDV, SSTV, Olivia, DominoEx, the Winlink 2000 system and many others. We typically discuss and operate digital modes on HF but sometimes operate on VHF/UHF as well, so everyone gets a chance to participate regardless of your station equipment or license class.

Using a radio and your computer, you can send data, voice, pictures, documents, and even email over the air. We also discuss using the various digital mode software applications and explain how to install and use them.

Arduino and Raspberry Pi

Arduino and Raspberry Pi

We also discuss using microprocessors like the Arduino as well as small micro-computers like the Raspberry Pi. Both of these devices and similar development boards are fun to learn about, and are great for creating projects you can use with amateur radio.

If you are interested in digital communications or learning about working with microprocessors and micro-computers, we are happy to help you get set up and explain how things work.

The net is hosted by Brian – K6BPM and everyone is invited to check in. We hope to see you there!

You’ll find recordings of previous nets in the SBARC Audio Archives.

Please consider joining the SBARC Digital Modes and Digital Radio Discussion Group mailing list and participate in the group off the air! It’s easy to do. Just send a blank email to

For quick access to the programs we use for digital communications and other helpful resources, visit the link below…

Digital Communications Downloads and Resources

New Packet Capabilities at TCF

We have just installed a new VHF packet node at the Talanian Communications Facility on the Mesa. The new station has a Winlink gateway that can gate mail between RF and the internet (perfect for emergencies), as well as a few other features, including a local keyboard chat room!

Check it out if you have a TNC or sound card modem for VHF packet on 145.050.

The main node is K6TZ-2 but the following aliases will connect you directly to the various services: MESA (main node), TZPBBS, TZCHAT, TZRMS (Winlink)

Note that BPQ is a little different than Kantronics. Commands like MHEARD and CONNECT (to hop) require a port number (in this case the port number to use is 2). More details in the welcome message.

Thanks to Doc, W6EW for donating the Kenwood TM-D700 to the club and to Bill, W1UUQ for leaping into action when I said I needed help tracing out the coax at his namesake TCF.

This is the first of a couple major planned packet upgrades. We expect to install an updated digipeater at the La Cumbre West Site in the coming weeks which will expand the reach of this node.

If you are interested in learning more about VHF packet radio and other digital modes, join us on Tuesday nights for the Digital Communications & Technology Net and consider joining our group mailing list.

-Levi, K6LCM
K6TZ Trustee

Accessing AllStarLink from a SIP telephone

I managed to get my Western Electric rotary phone connected to ASL using an ATA.

I have been using a SHARI AllStarLink node for some time as a means to tinker and learn about AllStarLink (ASL) and to explore the wide web of nodes around the globe that are connected to this vast network.

ASL (HamVoIP being the flavor of it on my node) is based on Asterisk, an open source PBX system that is capable of handling the most complex telephony tasks and could conceivably be used to replicate or even replace nearly all voice switching capabilities of the public switched telephone network. On the K6TZ repeater system, we have configured the repeater node to handle phone patch requests over VoIP, eliminating the need for a copper line to offer radio to telephone network connectivity. This means that members can key in a code followed by a phone number on the 446.400 repeater to initiate a phone call to any number in the USA.

I had also heard that ASL could provide for direct access to radio linking through SIP connections, sort of a reverse autopatch, whereby one could dial into the node from a SIP telephone to access and link to nodes throughout the ASL network. In setting out to configure my own SHARI node for this, I ran across a few inconsistencies on how to accomplish this so I am writing up this article to document what worked for me.

The first place I turned was the official ASL wiki documentation. It is helpful but somewhat incomplete in its explanation. Below I’ll outline the steps and try to explain them in a bit more detail, adding some additional configuration to support multiple SIP extensions.

The first step is to locate and edit the /etc/asterisk/modules.conf file. I am including the full path to all files since it can be challenging for newbies and experienced users alike to find them, especially when tinkering in ASL is not a daily task.

We must edit the modules.conf to be sure the SIP module is loaded by ensuring that the following line is present:

load => ; Session Initiation Protocol (SIP)

You may see this line in your file beginning with noload. Simple change it to load and save the file.

Next locate the /etc/asterisk/sip.conf file and open it in your favorite editor. In this file we will append the following code to add one new SIP extension to our node. Note that I am calling this extension 101. Conceivably you can use whatever extension number you wish. Note that anything after a semicolon on a line is a comment instructing you to change the particular contents to something that makes sense for your setup.

[101] ; Can be practically any extension number you wish
secret=pass!!word ; set this to a SECRET PASSWORD you create
dial=SIP/12345 ; Change to your assigned ASL node number
callerid=Ham Shack ; pick your own Caller ID for this extension

You could add multiple SIP extensions with different numbers (for example: 102, 103, etc.) if you had more than one SIP handset or ATA.

Once you have completed editing the sip.conf file, save it and open /etc/asterisk/extensions.conf.

Here we will need to add a new “context” called sip-phones to define how these extensions will work. In this case, we want to dial our node number and be connected to our SHARI node via VoIP. We will also let the system know how to handle calls placed to these extensions. This will allow us to pick up one handset (e.g. 101) and dial another extension (e.g. 102) and have it ring.

Be sure to change 12345 to your node number in the lines below.

exten => 12345,1,Answer
exten => 12345,n,Wait(2)
exten => 12345,n,Playback(rpt/node)
exten => 12345,n,Playback(digits/1) ;say node digits one at a time
exten => 12345,n,Playback(digits/2)
exten => 12345,n,Playback(digits/3)
exten => 12345,n,Playback(digits/4)
exten => 12345,n,Playback(digits/5)
exten => 12345,n,Rpt,12345|P|${CALLERID(name)} ;

exten => 101,1,NoOp(SIP Phone 1)
exten => 101,n,Dial(SIP/101)

exten => 102,1,NoOp(SIP Phone 2) ; if you have a second extension, 
exten => 102,n,Dial(SIP/102) ; be sure to include these two lines.

Now that you have made the config file changes, we must restart the Asterisk Server on the node. ssh into your node and select menu item 13.

HamVoIP Main Menu Screen accessed via SSH

Example client configuration screen from a Grandstream SIP handset

The node is now ready to accept incoming connections from SIP handsets and ATAs. Configuration of your hardware will vary by vendor but basically you will want to name the SIPM account something like “AllStarLink” and point the SIP Server setting to the IP address of your node (this assumes you are on the same local area network as your node). The SIP Authentication ID is the extension number you defined in sip.conf. which in this case is 101. The Password will be whatever you changed it to (hopefully not pass!!word but it really doesn’t matter much if you are on your own network behind a firewall). If your phone or client asks for a SIP port, the default is 5060. That’s pretty much it. Your save the configuration and your SIP phone should register with your node as its PBX. You can dial into your ASL node by picking up the line and entering your node number. Once it is connected, you can dial around using the same DTMF commands you are used to, including *3 + node number to connect to outside nodes.

One important thing to note is that hanging up the line won’t disconnect links. It simply disconnects your SIP phone from your ASL node instance. If you pick the line back up and dial your node number again, your session is restored as you left it.

If you wish to dial another SIP phone extension on your new PBX, simply pick up the line and dial the extension number. In this example, picking up extension 101 and dialing 102 on the keypad will ring extension 102.

You can configure inexpensive SIP phones like the Grandstream GXP1620 (note that this particular model requires Ethernet and does not have WiFi capabilities), an ATA like the Cisco ATA 192 – for use with a standard landline phone, or even smartphone SIP apps like Zoiper (iOS / Android) as long as your phone is connected to the same network as your ASL node.

K6LCM LiFePO4 PowerBank Battery Box

The author’s battery box shown here at the K6TZ Museum of Doughnuts on a Saturday morning.

I experienced some disappointing results with a portable 12-volt jump-start lead-acid battery at Field Day this year. David, AC9AC, saved Field Day for me by bringing by his 30 amp-hour LiFePO4 battery to our operating location at Shoreline Park. Impressed with its capabilities, I decided to upgrade my portable power.

There are a number of vendors on Amazon and Ebay selling high-capacity LiFePO4 batteries at low prices. LiFePO4 batteries are impressive. Without going into too much detail, the main advantages are that they are up to 70% lighter than lead acid batteries, will continuously supply 13-14 volts under high-current draw conditions and can be completely discharged without damage. Most LiFePO4 batteries include a battery management system inside the sealed plastic battery itself, making them nearly a drop-in replacement for lead-acid applications. In fact, many new 12-volt chargers include special modes for optimally charging LiFePO4 cells.

If you search online, you’ll find many pre-made power stations. The problem is that most of them are designed with general consumers in mind. Your average power-hungry electronics guru mostly needs a 5-volt USB power connection and a 120-volt AC outlet. These premade power stations typically lack higher current connections such as Anderson Powerpole sockets. The designers of these pre-made power stations were probably thinking: “Who would want to draw 20 amps at 12 volts?” Hams would of course!

My requirements when designing the K6LCM PowerBank were that it would support typical 5-volt USB connections and a 12-volt DC cigarette lighter connection for running mobile chargers. But my design added two 12-volt DC Anderson Powerpole connections for supplying up to 20 amps of current for higher draw devices like HF rigs.

Below are some photos of the build and a parts list. I mounted the binding posts inside the top compartment of the Harbor Freight ammo box as shown. Using 12 AWG wire, I connected the battery itself (using the blade fuse connector with a 20-amp fuse) and all of the power outlets to the positive and negative posts respectively. In addition to the fuse, the positive battery lead passes through the master switch on its way to the positive binding post. The only connection that does not pass through a binding post is the positive (7.5-amp fused) connection between the charging port on the back and the battery. Since I intend to use this setup as portable power, I won’t be running the charger and the radios at the same time. However, this 7.5-amp fuse protects the charger, just in case!

If you intended to use this PowerBank as an online battery backup in your shack, you could wire in your 13.8-volt, high-current shack supply instead of a battery charger. Consider the West Mountain Radio Epic PwrGate for this application.

Parts list:

The Hall Effect ammeter sensor is looped around the supply cable from the battery to the positive binding post and shows the current flow out from the battery.

Some newer Battery Tender chargers have settings for optimally charging LiFePO4 batteries. The volt meter doubles as two USB power ports. Connections from the various components to the battery are made through + / - binding posts. Be sure to install a 30-amp fuse on the main lead between the positive binding post and the battery. On this particular ammo box, the binding posts are concealed under the top compartment lid. High and low current 12-volt DC connections .An SAE connection for the Battery Tender charger is mounted on the back.

FM Simplex Node Frequencies in Santa Barbara


Over the past few months there has been a renewed interest in AllStarLink FM simplex nodes among our members. These low power devices allow users to connect to remote VoIP nodes and repeater systems using an internet connection and an HT. Most opt to buy or build a device on 70cm.

A couple years ago, during the initial DMR hotspot craze, SBARC expert consultant Matt W6XC identified a few frequencies around 431 MHz that were useful for digital hotspots like the OpenSpot and ZumSpot. HOWEVER, these frequencies are NOT appropriate for analog FM nodes. We must use a different part of the 70cm band for FM emissions.

Matt suggests the following options for low-power, analog FM usage:
440.000/445.000; 446.860/441.860; 446.880/441.880 may be used as pairs for a low-power duplex node or as separate simplex node frequencies. Please listen to these frequencies with low squelch settings and no CTCSS for a few days before permanently parking your node here to see if and how these frequencies are used. FM simplex nodes are best PL/CTCSS protected, especially if they are left connected to a system like K6TZ or WIN System.

Definitely avoid 446.000 altogether. This is the National Calling Frequency for 70cm. 446.500 and 446.520 are “General Simplex” frequencies. Others many want to use these for simplex QSOs or other itinerant purposes so please don’t park your node on any of these three frequencies. Also note that 432.000-439.999 MHz is allocated to weak signal, Amateur Television and digital emissions only. 440 is tough given the lack of simplex allocations. It’s a truly stuffed band!

Perhaps just as important as which frequency you choose for your node is setting the PL/CTCSS tones. In Santa Barbara and Ventura counties, DO NOT use 131.8 or 88.5 127.9 or 131.8 Hz as a tone for your node. Picking almost anything else will ensure that you don’t inadvertently open the receiver of a repeater on the same or nearby frequency.

If you are interested in these FM VoIP nodes, consider the ClearNode and SHARI projects.


Levi, K6LCM
K6TZ Trustee

November General Club Meeting: Tracking Transpacific Airliners

0:00 Pre-meeting chatter
26:30 Meeting Start and Intros
53:30 Tracking Transpacific Airliners Presentation Start
1:27:10 Questions & Answers

Meeting Presentation Slides (pdf)

Our November Club Meeting was host to our Board of Directors election as well as a presentation by Levi C. Maaia – K6LCM on using your ham equipment and/or computer or smartphone to listen in on transpacific airline traffic on the VHF and HF bands.

Hams aren’t the only ones using HF on a daily basis for reliable, long-distance communications. Airline pilots use HF frequencies from 2800 kHz to 22 MHz as their primary means of communication with shore stations during oceanic flights. These comms can be received by anyone with an HF SSB tuner and provide interesting data points for HF propagation. Aircraft are even sending PSK over HF! Levi showed us a tracking demonstration of a flight from Los Angeles (LAX) to Sydney (SYD) as well as pointed us toward some resources for tracking and listening in on our own, with or without a radio!

DMR QSOs from your Android

I just tried out a cool app for Android users …

DROID-Star lets users login to the Brandmeister DMR system using your DMR ID and presents an interface that is operationally similar to Echolink. You can connect to any TG and then use the on-screen PTT to have a QSO. Works decently well for a beta version. Despite the “D-Star” branding it is a DMR app. Haven’t figure that part out.

Download it from the Google Play App Store.

-Levi, K6LCM

Yaesu System Fusion-DMR (YSF2DMR) Cross-Mode Repeater

Over the past two years, there has been an explosion of interest in DMR amateur radio. Many SBARC members have been bitten by the DMR bug and they are chatting around the world on global Brandmeister talkgroups using hotspots and repeaters. While much of the attraction of DMR is the ability to work DX on a handheld transceiver, many local operators hang out on the local SBARC Brandmeister DMR Talkgroup (TG 31073).  In fact, every Tuesday night, the Digital Communications and Technology Net moves from 2m FM to TG 31073 at 21:00 Pacific Time.

If you aren’t on the air with DMR yet, not to worry.  You may be able to chat on TG 31073 with a radio you already own! K6TZ operates a multimode digital repeater at La Vigia on the Mesa in Santa Barbara. We have recently reconfigured this repeater to bridge traffic from Yaesu System Fusion radios to TG 31073 using a new protocol called YSF2DMR. If you have a newer Yaesu radio, you may be digital-ready right now. Many new Yaesu amateur radio models available support System Fusion and are capable of connecting to the K6TZ digital repeater in order to bridge from System Fusion to DMR and TG 31073.

Joining the local digital chatter on TG 31073 via System Fusion on the Santa Barbara South Coast is fairly straightforward. Just follow these steps:

  1. Obtain a DMR ID.
    If you have not already, click here to register your callsign with the DMR network and receive a unique DMR ID number. Your DMR ID is paired to your callsign on the DMR system and used to identify your transmissions. It takes a day or so to get a new DMR ID assigned and you only need to register once. Once you receive the confirmation email, keep it. You won’t need the number now using System Fusion but if you get bitten by the DMR bug in the future and want to explore further you will use this same DMR ID to configure a DMR radio.
  2. Set your amateur radio callsign in your System Fusion radio.
    Each radio model handles this differently. You must enter into your Yause radio the exact same callsign you registered to your DMR ID. The SBARC repeater will only bridge properly identified transmissions from System Fusion to DMR. So be sure you have input your callsign correctly and that you have received confirmation of your DMR ID registration by email before attempting to move on to the final step.
  3. Set your Yaesu System Fusion radio to the K6TZ digital repeater frequency pair.
    Tune 445.480- (negative offset) on your Yaesu System Fusion Radio.  Be sure you set the radio to transmit in Digital Narrow (DN) mode. Voice Wide (VW) and FM transmissions will not be bridged to DMR.

Hope to hear you on DMR/System Fusion. If this interests you and you want to learn more, join us every Tuesday night at 20:00 (8:00pm) on the K6TZ 146.79 FM repeater for the Digital Communications Net. Then QSY with us to DMR TG 31073 at 21:00 (9:00pm).