Idiot's guide for SIP on ASL3?

G4ZWH · August 27, 2024, 3:17pm

Hi all

Has anyone written an “idiot’s guide” to setting up an incoming SIP call with ASL3?

I am struggling with PJSIP etc

Thanks & 73
Robert G4ZWH

k6ian · August 27, 2024, 6:36pm

@Mason10198 Wrote up an excellent guide, but it uses iax protocol. I was banging my head with Autopatch and Reverse Autopatch setup, but with this guide it was seamless.

gist.github.com

https://gist.github.com/Mason10198/89e9d7e6bbae8a321b45286c78c5783b

ASL3Autopatch.md

# Giving Your ASL3 Node a Phone Number - Setting Up Autopatch and Reverse Autopatch using VoIP.ms
**Author:** N5LSN

> [!IMPORTANT]
> As of 8/24/24, ASL3 half-duplex (duplex=0,1) autopatch still has a few bugs to work out before it is fully functional. Incoming calls (reverse autopatch) work fine, but outgoing (autopatch) calls are unable to pass phone audio to the ASL3 node.

## Introduction

This guide will walk you through setting up both Autopatch and Reverse Autopatch functionality on your ASL3 node using VoIP.ms as the service provider. This allows your repeater to make and receive traditional phone calls. This is particularly useful for emergency contact of people outside of the repeater range or otherwise unable to directly use the repeater for communications. It also is great for allowing system administrators remote access to execute DTMF functions from any active phone line.

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tsalzer · August 27, 2024, 7:25pm

Don’t know if this is what you need. I wrote up my notes on PJSIP and ASL3 at Random Wire Review 98: July 12, 2024 - by Tom Salzer

Mason10198 · August 28, 2024, 12:01pm

G4ZWH · August 28, 2024, 1:03pm

Thanks, this actually looks like just the ticket. I will have a go upgrading and updating.

Many thanks & 73

G4ZWH · October 16, 2024, 10:04pm

Sorry to bump an old thread but I have followed the guide line by line, and I cannot get the SIP connection to register with PJSIP.

Is there a command I can use to view the live log to try and solve this?

Thanks & 73!

Edit, I forgot to say the error code zoiper gives is 408, request timeout.

lattuga_tartaruga · November 28, 2024, 2:49pm

Awesome article and thank you Tom! I follow the SIP configuration, however, I did not see any info on how to configure the Cisco 525 (or screenshot), I understand this is supposed to be pretty straightforward, however, for some reason, I am unable to get that phone extension to register on the 525 (failed - no response error)… it just won’t go green any tips/help (from anyone) would be greatly appreciated!

Mason10198 · November 28, 2024, 5:37pm

AFAIK, Cisco phones only work with patched Asterisk. Or maybe that’s just the “series” of Cisco phone I use.

https://usecallmanager.nz/

I have a Cisco 8845 to which I flashed the 3PCC/Multiplatform firmware, and I set up a second (“Master”) vanilla Asterisk server in my network that I then patched according to the above link, and then created an IAX trunk to my ASL3 server. Then the Cisco registers with “Master”, and extensions are routed to/from ASL3 accordingly.

Mike · November 28, 2024, 9:09pm

As @Mason10198 stated, ‘some’ Cisco phones have particular issues, and I think you can locate that issue with a google search as it is standard asterisk stuff. Nothing unique for app_rpt/ASL purpose.

I have a multitude of sysco phones but most of them are 608G 8 line phones and have no issue nor it’s 4 line equivalent model.

I suggest if you bought them used, reset them to factory and start from there.
Upgrade firmware if available. Then set them up.

lattuga_tartaruga · November 29, 2024, 12:28am

Mason & Mike: thank you both for the feedback/tips. I have tried before and after the fw updates (including full factory resets) for my 525g2 to no avail. I was just curious about @tsalzer’s 525 settings because it’s the same phone mentioned in the article (nothing about patching or any additional steps were mentioned but perhaps there were). The name of this topic fits the bill in my case, as I’m admittedly an idiot about all this stuff. I will read up a bit more. Thanks again!

Mike · November 29, 2024, 12:53am

lattuga, give us some additional info so we might understand how you are sitting.
It may be telling or not. Perhaps some pointers.

Are you using asl3 ?
Are you trying to do this under sip or pjsip ?

lattuga_tartaruga · November 29, 2024, 1:18am

@Mike sure thing and thank you! i followed this guide (asl3) and used the same exact settings… which use pjsip:

the only differences are:

a. i obviously used my own allstar node extension
b. asterisk is running on standard 5060 vs his 5062
c. running on a raspberry pi 4 vs a deb12 vm

my 525g2 is the Bluetooth version and I’m now on the latest fw. ext1 is hoip, ext2 is hhops and I’m trying to use ext5 for this connection. my internal network is very basic… nothing special, no vlans, no open ports except for my wireguard.

Mike · November 29, 2024, 1:42am

Well, I always hate doing sip stuff on here. LOL and it’s more complicated now than ever and I use a non standard set-up.
So, I will try to stay with the program.

But start with the basics since you have no connection.
It seems first there is a network issue.
Inside of that, did you open port 5060/or 5061 in your firewall ?

Perhaps open your local lan (if this is on local lan)
sudo ufw allow from 192.168.1.0/24 - using your local lan ip or you could use the guide and be specific to UDP port 5060/5061

sudo ufw status

lattuga_tartaruga · November 29, 2024, 1:56am

@Mike: that’s exactly what the issue was! i added the “sip 5060” service in the raspberry pi networking asl3 cockpit and the light went green! you’re the bee’s knees!

THANK YOU!!!

ps. mind you, now that i have this configured, i have zero idea what to do with it

tsalzer · November 30, 2024, 9:36pm

I probably should have said that my 525G was set up to run my Hamshack Hotline number, so the firmware comes from HH: https://wiki.hamshackhotline.com/doku.php?id=fu:spa-525 (there are two versions, one for the 525G with Bluetooth and one without). I don’t know if this makes a difference.

lattuga_tartaruga · November 30, 2024, 10:41pm

all good, @tsalzer ! thank you! got it working beautifully on my local node connecting to BM DMR via usrp (to dvswitch)… it’s really neat (and kinda funny) that it works so well… I appreciate the response and the guide very much as I would not have figured this out on my own!

cheers!