On 2/15/2017 5:22 PM, James R. Pilgram wrote:

You may want to check out http://aplaceonthe.net/blog/connecting-the-vertex-vxr-7000-to-allstar-app_rp...

That site goes through a HUGE dissertation of hacking up the radio, complaining about the RX audio quality (which there is nothing wrong with - except it's low level), and more grumblings. All they would have needed to do was set RXBOOST=1 and the radio tune would have calibrated just fine. I'm rather certain I used pin 6 of the DB25 for receive audio to the URI, but we'll see. Since there's more than one person wanting this information, I *may* go to the site this evening. Kevin

On 2/15/2017 6:08 PM, Kevin Custer wrote:

All they would have needed to do was set RXBOOST=1 and the radio tune would have calibrated just fine. I'm rather certain I used pin 6 of the DB25 for receive audio to the URI, but we'll see.

Since there's more than one person wanting this information, I *may* go to the site this evening.

Kevin

So here we are... back from the site. There's a DB25 male connector that plugs into the Vertex VXR-7000. And, there's a DB25 male on the other end that plugs into the URI. On the repeater end I have the following: Pin 1 = shield = GND Pin 6 = white = DISC OUT Pin 12 = orange = EXT PTT Pin 13 = black = GND Pin 24 = red = TXD LOW On the URI end I have the following: Pin 1 = orange = PTT Pin 13 = shield = GND Pin 19 = black = GND Pin 21 = white = MIC_AC Pin 23 = red = RIGHT_OUT That's it - no surprises. Program the repeater correctly, and turn on "Remote" by pressing the switch on the front panel of the VXR. Use the files at www.masterscommunications.com/temp/2302/ as a guide for your ASL configurations. The files shown are for DIAL. I used the right output for voice - you don't have to, just use pin 22 on the URI instead of pin 23 and change the configuration in usbradio.conf to: txmixa=voice txmixb=no For transmitter audio, I chose TXD_LOW instead of TX AF IN because it sounded better. This input is before hardware pre-emphasis and limiting so both are applied to audio that's inputted here. That's why the usbradio config file does not have preemphasis or limiting turned on. The high and low pass filter references in my usbradio.conf file refers to code that's currently in beta testing for DIAL. That code will eventually be included in the DIAL release. It's been available in XIPAR for years. Steve Zingman may choose to comment more on its availability for those wanting to try it, and I can explain what it does if someone is interested. Hope this helps... Comments or questions welcomed, Kevin - W3KKC

On 2/16/2017 3:31 AM, Eric EG. Guth wrote:

Thanks, Keven. We have a two meter VXR7000 in Safed that we want to put on to Allstar. We will follow your instructions. Do you think that USBradio will work find with Raspberry Pi3 on this system?

Yes... While in this instant case the computer is a PC, I have several installations (like 20) that use full DSP on one repeater (node), on a Raspberry Pi3, using DIAL. There are a few things I do by default to the RPi3: 1: Add the following "fix" DNS. cd /etc/asterisk wget https://github.com/N4IRS/AllStar/raw/master/configs/dnsmgr.conf restart asterisk (or just reboot). 2: Turn off Radio Relax - deselect it in the menu. Stop asterisk: service asterisk stop Verify asterisk is stopped: ps ax | grep asterisk If you see /usr/sbin/asterisk it is not stopped killall -9 asterisk ps ax | grep asterisk If you see /usr/sbin/asterisk it is not stopped Change compiler setting: cd /usr/src/astsrc-1.4.23-pre/asterisk make menuselect select 10. Compiler Flags select 6. RADIO_RELAX (use space bar to select or de-select) x make make install service asterisk start (or just reboot). 3: Add Jeff DePolo's filter code. I use WinSCP to transfer the files because I don't know any other way to do it. Download the five required files known to work: http://www.repeater-builder.com/w3kkc/DIAL/ Stop asterisk - service asterisk stop Put usbradio.conf in /etc/asterisk (or edit your current file with the added filter code) Put chan_usbradio.c in /usr/src/astsrc-1.4.23-pre/asterisk/channels Put xpmr.c, xpmr_coef.h, xpmr.h in /usr/src/astsrc-1.4.23-pre/asterisk/channels/xpmr cd /usr/src/astsrc-1.4.23-pre/asterisk make make install Start asterisk - service asterisk start (or just reboot). Hope this helps... Kevin

Hello Kevin, Thanks very much for the info. As to you question, the COR does not work at all on the VXR-7000. It show no movement with a volt meter. I guess it is bad. Again, Thanks so much for your help. Angelo PS : would love to get your phone number just in case there is any issue I may need to chat with you about. You have email me at n5uxt AT hotmail.com On 2/15/2017 7:13 PM, Kevin Custer wrote:

On 2/15/2017 6:08 PM, Kevin Custer wrote:

...

All they would have needed to do was set RXBOOST=1 and the radio tune would have calibrated just fine. I'm rather certain I used pin 6 of the DB25 for receive audio to the URI, but we'll see.

Since there's more than one person wanting this information, I *may* go to the site this evening.

Kevin

So here we are... back from the site.

There's a DB25 male connector that plugs into the Vertex VXR-7000. And, there's a DB25 male on the other end that plugs into the URI.

On the repeater end I have the following: Pin 1 = shield = GND Pin 6 = white = DISC OUT Pin 12 = orange = EXT PTT Pin 13 = black = GND Pin 24 = red = TXD LOW

On the URI end I have the following: Pin 1 = orange = PTT Pin 13 = shield = GND Pin 19 = black = GND Pin 21 = white = MIC_AC Pin 23 = red = RIGHT_OUT

That's it - no surprises. Program the repeater correctly, and turn on "Remote" by pressing the switch on the front panel of the VXR. Use the files at www.masterscommunications.com/temp/2302/ as a guide for your ASL configurations. The files shown are for DIAL.

I used the right output for voice - you don't have to, just use pin 22 on the URI instead of pin 23 and change the configuration in usbradio.conf to: txmixa=voice txmixb=no

For transmitter audio, I chose TXD_LOW instead of TX AF IN because it sounded better. This input is before hardware pre-emphasis and limiting so both are applied to audio that's inputted here. That's why the usbradio config file does not have preemphasis or limiting turned on.

The high and low pass filter references in my usbradio.conf file refers to code that's currently in beta testing for DIAL. That code will eventually be included in the DIAL release. It's been available in XIPAR for years. Steve Zingman may choose to comment more on its availability for those wanting to try it, and I can explain what it does if someone is interested.

Hope this helps... Comments or questions welcomed, Kevin - W3KKC

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Thanks Steve for all you hard work on this. I am involved with a couple other projects that use git and I find using the issues section to track bugs and feature requests to be very effective. Is using git issues for ASL a viable idea? Thanks, Steve On Feb 16, 2017 12:23 PM, "Steve Zingman" <szingman@msgstor.com> wrote:

Along with the multitude of things we are doing to shore up the ASL (AllStarLink) infrastructure, we have started to apply a number of fixes and add some new features to the next version of asterisk used in ASL.

I am not going to try to detail every change, it's moving much too fast for that. If you want to follow the changes, it's time to start learning your way around git. We have moved any new fixes or development away from the SVN to a public repository in git. Everything ASL related is now in < https://github.com/AllStarLink> You do NOT need a git account to look at or even download the code from git.

I am starting to move the repository from my own github account to the ASL repository. There are a number of updates I need to apply to the Debian and DIAL scripts. I will copy the cleaned up code into the ASL/DIAL repository. Stay tuned.

We are using a very CONTROLLED method of development to guarantee each fix and feature is vetted before it makes it way into the master release. One of the many upsides is that you can see the changes as they are applied to the different branches as they make their way towards the master. As members of the development team start committing their work you will see it.

The master branch is as it was plus one hotfix that needed to be applied. You will not see any change there until we are ready for a release. The develop branch is where all the fixes and features will start to appear. it is at <https://github.com/AllStarLink/Asterisk/tree/develop>

The first of those features is something the WANRS < http://www.wanrepeater.net/> people have been asking for. For quite a while I might add. They needed the ability to tailor the receive and transmit audio characteristics of the chan_usbradio channel driver. This feature has been available for quite a while in XIPAR and they felt if they were going to update their existing installations, they had to have that and at least one other feature. Late last summer Mike N4IRR added the feature to a custom version of DIAL that the WAN people have been testing. The testing has been done both on the air and with some very high quality test equipment. We have the thumbs up on it, so it is now in the develop branch.

I'll say this again, you will not see any new feature in the version of asterisk used in ASL, today, tomorrow or next week. I do not have a feature freeze date let alone a release date. There is a TON of work to do. The teams are small (on purpose) and they all have day jobs. But the upside is you will be able to watch the progress.

I want to encourage everyone to do a few things:

1: If you see a question on the list, Answer it. The more people answering questions the better for all.

2: If you have a skill you think can be helpful to the overall project, let me know. Not everyone is a software architect, We need people who can write (unlike me). People that can script (bash). And let's not forget the people that spend quite a bit of time interfacing ASL to repeaters. There are a lot more tasks that can be filled. No, not everyone can contribute, but it's worth a shot.

3: If you have a idea, post it here. I read every message and after a few back and for discussions I may capture the idea and stick it in a log for review and discussion.

For AllStarLink Inc. A non-profit

73, Steve N4IRS

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