ASL3 questions - AutoConnect to another node, Change Morse ID to Voice ID & Logs

1

Here are some issues or I can’t seem to find the answer, so far :slight_smile: and am hoping someone here might know for ASL3

1 - I want to be able to connect to my other node on startup. In HamVoip there is a place in the Rpt.conf, but something is different, so I don’t know where that is.
When I try to connect my node 62030 to 61172 or the other way around, it just sits there saying connecting and will not connect. But when I connect to another node of someone it connects. Both nodes are currently at my home on the same network.

I did change the one port from 4569 to 4580 and opened the port up. Am I missing another port that I need to change and open up?

2 - how to I change from Morse code ID to Voice ID instead?

3 - Logging for connections and web connections, how and where to turn that on?

So far, I have been doing searching, but so far, no luck in getting this resolved.

Does anyone know if DVSwitch works with ASL3?

Thank you and 73

Rob - K6IRK

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2

In order of your questions:

1 - Two nodes on a local network will be always be squirrely regardless of ASL3 or HamVOIP because the system is using your public IP address as NAT’d through your router. Depending on a whole host of randomness in a given setup, it’s hard to predict how that may work. If these nodes don’t move, then I’d suggest putting custom entries into rpt.conf and hard-code your local LAN IPs. Another possibility is that your router setup is doing enough traffic handling to send your traffic to a node but it’s responding locally. In this case, you could try toggling off the firewall in cockpit to see if that resolves your issue. However that is NOT recommend as a permanent solution.

2 - Simply change your idrecording in your local node’s [62030](node-main) staza from an |i to a path to an appropriate encoded .gsm file. See Rpt.conf - AllStarLink Wiki.

3 - I’m not sure what you mean by "logging for connections and web connections? Can you clarify?

4 - By “DVSwitch” are you asking about the DVSwitch Mobile app? If so, yes it does.

3

Yes, hairpin NAT can be a problem on some routers. That’s where the [nodes] stanza the OP mentioned comes into play. ASL3 is no different in that regard than any other AllStar system.

Two nodes behind a NAT router is well documented and has not changed in ASL3. See https://wiki.allstarlink.org/images/0/0c/ASL_two_servers_one_netwrok.pdf

4

Thank you for the information.

I have my 2 allstar nodes now communicate.

For the logs, I would like to see the Connection Log, AST Log & Linux Log

Using Supermon for the Connection log log, it is looking for this:

File: /var/log/asterisk/connectlog.2

File: /var/log/asterisk/connectlog.1

File: /var/log/asterisk/connectlog

AST Log I get this
File: /var/log/asterisk/messages

Linux Log I get this
Commands: echo ‘/var/log/messages:’; /bin/cat /var/log/messages; echo; echo ‘/var/log/syslog:’; /bin/cat /var/log/syslog

/var/log/messages:

/var/log/syslog:

The connection log would be the most handy one for sure.

DVSwitch I was talking about is what would load onto the node, not the mobile app :slight_smile:

I have added info to connect to my other node from the ASL3 node on startup

62030
statpost_url = http://stats.allstarlink.org/uhandler
duplex = 1
idrecording = |iK6IRK
;;;;;;;;;;;;;;;;;;; Your node settings here ;;;;;;;;;;;;;;;;;;;
rxchannel = SimpleUSB/62030 ; SimpleUSB
startup_macro = *8161172 ; *8132000
wait_times = wait-times_hd
hangtime = 100

One example showed *8361172 and this one above shows *8161172

I have restarted the node each time, but still no luck. Am I using the wrong format?

Thank you
Rob

5

I cannot advise you on anything about Supermon. I know that /var/log/asterisk/connectlog requires installing and configuring some process specific to Supermon - the “smlogger” stuff. Have you tried managing the connections and data in Allmon3? That works out-of-the-box and does not require any special configuration or local installation files.

Regarding /var/log/* items, I assume you’re using the Pi appliance. The Pi appliance is not running syslog by default to cut down on write-wear of SD cards. I would not suggest it, but you can install the rsyslog package if you want traditional syslog logs.

Finally, what are you trying to accomplish with the startup_macro line?

6

with the startup_macro I am trying to connect this node to my other node

the asl3 node is 62030 and the node I want to connect up with is 61172 when I startup the node instead of manually connecting it

7

startup_macro=*361172

8

Perfect and thank you. the information in the file showed *8132000, so I assumed that was the format it had to be in.

works now.

Rob

9

Oh, I see what you were trying to do… *813 is the default “persistent link” command. So you could use *81361172 but that’s likely overkill for the startup_macro.

10

The 813 DTMF function is commented by default in rpt.conf. Uncomment it, change your startup macro back to 813 and restart asterisk.

Permanent connects (*813) are very useful for automatically reconnecting after a network outage or far end node going off line.

11

I should mention that when uncommenting and using 813, functions 811 and 812 should also be uncommented and used.

12

Thank you very much for the advise. I have uncommented that as well.

Would you by chance know if this script will work in ASL3? It works in HamVoip, but would like to use it on ASL3 too.

Thank you again.

#!/bin/bash

Witten By Tony Morris WRAW556 K1TNY

GMRS Live support@gmrslive.com

03/12/2023

Just call the command from you node computer

random.sh 1

Directory can be any directory on the computer. ONLY have audio files in the directory

The audio files must be anything Asterisk can play. Mix and match.

File Name doesn’t matter.

node is the node you want it to play on

Placing a 1 at the end will play accross ANYTHING connected to you. Leaving the 1 off

will play on THIS node only

Example: The command below will play a file randomly from the /var/lib/asterisk/sound/notices

directory on node 1234

random.sh /var/lib/asterisk/sound/notices 1234

Make sure you give this script permissions to run

FILES=($1/)
RANDOM_INDEX=“$(($RANDOM % ${#FILES[@]}))”
echo ${FILES[$RANDOM_INDEX]%.}

if [ -z “$3” ]
then
play=“localplay”

else
	if [ $3 -eq 1 ];then  play="playback"; fi

fi

sudo asterisk -rx “rpt $play $2 ${FILES[$RANDOM_INDEX]%.*}”

13

I’m sure we could make it work. Our sounds are in /usr/share/asterisk/sounds/ but we don’t have a notices directory. If that is something HamVoIP is willing to share, we’d be happy to consider it.

14

The sounds is something we have to make ourselves

GMRSLive Random Play.pdf (104.1 KB)

Here is the document on it.

This is the video Freddie Mac did on it for HamVoip and GMRSLive

Custom Voice ID’s for GMRSlive and HamVoip AllStar (youtube.com)

15

Oh, I see. Sure. Please remind us in a couple of weeks about adding features. We’re working on some bugs at the moment.

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