Unitrunker with DSD+

[caphab1]

Hey all I understand I am dusting off an old thread but I followed these to a T and still getting no analog or digital audio out of DSD.

The pictures attached are two diff systems. I have attempted to manipulate multiple input settings on windows and in UT. any and all help greatly appreciated!!

THANKS!!

 

[AggieCon]

Caphab1, what you are trying to do is a bit different than the situation of the OP. Probably would be best for a new thread but that ship has sailed... Can you upload new pictures that are higher resolution, and please do not put stuff in front of the DSD+ window. I couldn't see your configuration.

If you are wanting to listen on your headphones, for the voice VCO, you will select your headphones for the analog output and the virtual cable for the digital output. What type of system are you monitoring (EDACS, P25, etc.)? You will need to have the appropriate boxes selected under voice or it will ignore the traffic.

In DSD+ your input should be your virtual audio cable and the output will be your headphones.

Your SIGNAL vco should not have any audio output. It should be set to unspecified for both digital and analog. That might be your problem. Private message me if you want.

 

[SCPD]

Hello CapHab;

AggieCon is sending you in the right direction.

Your voice role VCO should have the Audio Output pointed to speakers or headphones.

The digital output should be piped into DSD or DSD+ via VB Cable or VAC.

Make sure Analog and P25 are checked for the voice role VCO.

UniTrunker | Receivers

 

[dave3825]

Hey all I understand I am dusting off an old thread but I followed these to a T and still getting no analog or digital audio out of DSD.

You would only get digital audio out of dsd. Digital Speech Decoder. The analog audio would come from Unitrunker. Use the link Unitrunker posted and you should be up and running in no time.

Can you post a link to the system you are trying to monitor? I cant remember the exact setup from the original thread you woke up. Can you describe how and what you are using to monitor?

 

[caphab1]

Thanks for the feedback everyone! I will delve into those settings and let you know how it goes!!

Can you post a link to the system you are trying to monitor? I cant remember the exact setup from the original thread you woke up. Can you describe how and what you are using to monitor?

Its a couple different systems, one is the TOPAZ P25 P1which is the Mesa Simulcast site in the image: https://www.radioreference.com/apps/db/?sid=2082

And the other is Maricopa County Moto type II: https://www.radioreference.com/apps/db/?sid=848 .

Via 2 RTL-SDR Dongles, Using DSD+ 1.101, and UT. I set up UT with how the RTL/SDR blog describes (RTL-SDR Tutorial: Following Trunked Radio with Unitrunker - rtl-sdr.com), open DSD+, no addt'l setting, channels radio id's added.. still haven't quite master DSD in entirety. My problem has been when I change the settings in either UT or on the W7 settings to headphones, all i hear is the data noise..

Thanks!
Preston
Chandler AZ

 

[AggieCon]

Howdy Preston,

Let's focus on one system at a time. Since the Chandler Motorola site seems to have much better reception than the P25 site (based on your screenshots), let's stick to that one first.

As I mentioned before, I think you have the audio output from your SIGNAL vco routed to your headphones. Your SIGNAL VCO should not have any audio output. For both "Audio Output" and "Digital Output", the setting should be "Unspecified".

The following screenshot is how your SIGNAL VCO should probably look:

For the Voice VCO, the Audio Output should be to your headphones or speakers, and the Digital Output should be to your virtual audio cable input. See below.

Note: In the image above, set the Audio Output to your headphones or whichever device you want to listen on.

You will likely have to modify a couple of settings for DSD+, unless you are lucky that it defaults to the audio devices you need. If you send me a screenshot of the command prompt screen of DSD+, I can help you determine what attributes to use when executing the program. Basically, you need to tell it to listen on the virtual audio cable (-i) and play to your headphones (-o). It's probably best to tell it which type of encoding you are trying to listen to (-f).

Again, I think your main problem is you are streaming the raw control channel audio to your headphones, and your audio settings in DSD+ might not be correct. These are all easy fixes. You will be up and running in no time.

Hope this helps.

Justin

 

[caphab1]

Thanks Justin! Attached is an updated screen grab of the Maricopa system, UT settings and computer audio settings..

Im not sure how to tell DSD what to listen on and all that. Could you direct me to which file in DSD+ Im supposed to edit?

Thanks!

 

[AggieCon]

Based on your screenshots, it looks like you have it configured properly. However, on the Windows recording devices tab, it shows audio playing on Line 1 Virtual Audio Cable. Do you have something else routed through that?

There are two ways to run DSD+. You can run it from command prompt, or you can create a batch file (.bat) with the same command prompt information saved so you don't have to reenter it every time you want to run the program.

In the DSD+ folder, there is a text file "DSDPlus.txt" with instructions for using the program.

To create your own batch file, open Notepad and configure it to something like this:

dsdplus.exe -v4 -f1 -i1 -o1

Save the file in the folder with DSD+, perhaps with the file name provoicedecoding. Make sure your Windows folder settings are set to show file extensions. In the folder, right click and rename your new provoicedecoding.txt file to a .bat file. Then you can double click on the file to launch DSD+.

The commands I listed above do the following:

• -v4 -> tells it to give you more information in the command prompt about what the program is doing
• -f1 -> tells it to decode for P25 only
• -i1 -> specifies where to listen, which based on your screen capture, -i1 is your VB Audio Cable (your DSD+ defaulted to this)
• -o1 -> specifies where to play the output, which should be the headphones in your instance

Note, since we did not specify recording options, it should be defaulting to creating an ongoing recording of the decoded traffic to the file "DSDPlus.wav" in the folder with DSD+.



Also, if you add audio devices, such as more virtual audio cables, the input and output device numbers could change. This would cause you to need to update the -i and -o parameters appropriately to listen/play on the correct devices.



There's really no reason to "park" your voice receiver on a particular frequency, unless you want to listen to or record the output when it's not following something (for instance sometimes I park mine on conventional P25 channels just to perhaps hear a little bit extra if the trunked sites are slow).


Are you hearing any of the analog traffic? Are you seeing anything in the DSD+ source audio window? How busy is the site?

 

[AggieCon]

DSD+ 1.101 User Guide (DSDPlus.txt)

DSD+ 1.101 User Guide
---------------------


Installation
------------

BACK UP YOUR FILES

Before installing this version of DSD+, make backup copies of all of your current DSD+ files,
especially the data files:

DSDPlus.networks
DSDPlus.sites
DSDPlus.groups
DSDPlus.radios
DSDPlus.frequencies

Also back up any files you have created or modified, such as configuration (.cfg) and batch files.


If this is a new installation or if you are installing to a new folder (e.g. you use separate
folders for each version of DSD+ that you install), just extract all of the zipfile contents
to your installation folder. Then copy over any data/configuration/batch files that you have
modified from your old DSD+ folder to the new one.


If you are updating an existing DSD+ installation, just extract the .exe files (the programs)
and the .txt files (the documentation / help files); also extract all of the FMPA-related files.
Doing this will leave your existing data/batch/configuration files intact.



DLL files
---------

All of the DLL files used by DSD+, FMP and FMPA can be downloaded from dsdplus.com;
copy all DLL files to your DSD+ folder.



Usage/Options Summary
---------------------

Usage:
DSDPlus [options] Decode from audio stream
DSDPlus [options] ? file Decode from .wav file
DSDPlus -h Show help

Options ( [...] = default value):

Display/Logging options:
>file Create log file
>>file Append to log file
-_<num> Minimize selected windows at startup (bitmapped, 0-15) [-_0]
-- Show command line options in console window title
-t Time stamp console log file entries
-T Time stamp console log file entries and console screen data
-E Add NAC/RAN/DCC/RAS data to event log file entries
-H<num> High contrast mode (bitmapped, 0-63) [-H0]
-v<num> Frame information verbosity (0-4) [-v2]

-wsl<v>.<h> Source audio waveform window location [-wsl10.10]
-wss<h>.<w> Source audio waveform window size (0.0 blocks) [-wss200.300]
-wsp<num> Source audio waveform window update period (10-1000) [-wsp100]

-wel<v>.<h> Event log window location [-wel50.50]
-wes<h>.<w> Event log window size (min ?) [-wes400.500]
-weh<num> Event log window font height [-weh15]

-wcl<v>.<h> Channel window location [-wcl90.90]
-wch<num> Channel window font height [-wch15]

Input/Output options:
-i<spec> Input audio device (1-255) and channel (M/L/R) [-i1M]
-i<[addr:]port> FMP TCP link IPv4 address and port number

-g<num> Output audio gain (0.001-999; 0=auto) [-g0]

-o<spec> Output audio device (1-255; 0=none) and channel (M/L/R) [-o1]
in/out channels are optional; default=in:mono, out:mode based

-Och <file> Output audio file channel count and name/type [-O DSDPlus.wav]
ch: M=mono,S=stereo,blank=auto; file: .wav or .mp3 (NUL=none)

-I<num> Create new wav/mp3 file every <num> minutes (1440=daily) [-I0]

-P<wav|mp3> Also create per-call wav or mp3 files

-M<num> MP3 ABR kbps per channel (8-32) [-M15]

Decoder options:
-rc role is control/rest channel decoder
-rv role is voice channel decoder

-p Invert signal polarity (may be required for X2-TDMA and dPMR)
-mp optimize for PSK modulation (will not decode non-PSK)

-fa Auto-detect all protocols / frame types except dPMR [-fa]
-fd Decode D-STAR (no audio)
-fn Decode NXDN4800 (Kenwood NEXEDGE and Icom IDAS)
-fN Decode NXDN9600 (Kenwood NEXEDGE)
-fr Decode DMR/MotoTRBO (TDMA inputs + both output slots)
-f1 Decode P25 Phase 1
-fx Decode X2-TDMA
-fp Decode ProVoice
-fm Enable dPMR decoding (no audio)

-1 Synthesize audio for first DMR timeslot
-2 Synthesize audio for second DMR timeslot

-UA<num> AMBE unvoiced speech level (0-100) [-UA50]
-UI<num> IMBE unvoiced speech level (0-100) [-UI50]
-u<num> Unvoiced speech quality (1-64) [-u3]
-e Auto-mute encrypted voice

Advanced decoder options:
-dr<num> Rolloff filter (1-11; 0=auto) [-dr0]
-dh<num> Hotspot size (1-8; 0=auto) [-dh0]
-ds<num> Scaling factor (55-75; not used with D-Star or ProVoice) [-ds64]
-dd<num> Damping level (1-100; not used with D-Star or ProVoice) [-dd10]
-dv<num> Viewport size (1-30; not used with D-Star or ProVoice) [-dv20]

Active keys:
? Display active keys list in event log window
1 Synthesize audio for first DMR timeslot
2 Synthesize audio for second DMR timeslot
3 Synthesize audio for both DMR timeslots
- Toggle command line options display
| Toggle symbol phase display
A/a Adjust AMBE unvoiced audio level
B Show/hide background events in event log window
E Toggle auto-muting of encrypted voice
F Forget current system information
H Cycle high contrast modes
I/i Adjust IMBE unvoiced audio level
N Reset/redisplay neighbor list
P Toggle signal polarity
R Start/stop recording of raw source audio to wav file
S Close/reopen source audio waveform display
V Toggle voice call start alerts
W Display window locations and high contrast value
Spacebar Hold on current call
Esc End program

Source Audio Window:
Right click Pause/unpause source audio waveform display

Channel Activity Window:
Left click on priority Increase traffic priority
Right click on priority Decrease traffic priority
Left click on target Increase priority override
Right click on target Decrease priority override
Left click on headings Clear all lockouts
Right click on headings Clear hold



Run Modes
---------

The program can decode live discriminator audio or recorded .wav files.
Recorded audio files must be 48 or 96 kHz 16 bit mono PCM .wav files.



Trunk Voice Following
---------------------

Consult the Trunking.txt file.



Logging
-------

-v -t -T -E
-v0 generates minimal output.
Use -v3 or -v4 for maximum data logging.
-t and -T add timestamps to console log entries
-E adds NAC/RAN/DCC/RAS data to event log entries
Program output can be sent to a log file ( DSDPlus >logfile )



Input/Output
------------

-i -o
WaveIn/Out devices are listed at program startup.
Select your devices if you don't want to use the defaults.

If raw audio is coming from FMP or FMPA, a TCP connection should be used;
just specify a high port number (like 20000) in FMP/FMPA (-o20000) and DSD+ (DSDPlus -i20000)
If FMP/FMPA and DSD+ are running on different PCs, add the FMP/FMPA PC's IP address like so:
DSDPlus -i192.168.1.123:20000

Use -o0 to disable output audio.


-O
By default, all synthesized audio is written/appended to DSDPlus.wav
Use -O name.ext to write synthesized audio to another .wav or .mp3 file.
Use -O NUL to disable recording of synthesized audio.

-I
Use -I# to start a new synthesized audio recording file every # minutes.



Per-Call Audio Files
--------------------

-Pwav

Creates separate .wav files for each voice transmission.



Decoder Options
---------------

-fa
Using -fa (or nothing) will (usually) auto-decode all supported protocols.
Polarity of signals is auto-detected.
Note: -fa does not enable dPMR detection; -fm must be used to enable dPMR

-fd -fn -fN -fr -f1 -fx -fp -fm
When monitoring a single type of traffic, locking the protocol can provide
slightly better decoding results.
Note: more than one protocol can be enabled via the command line.

-u
Lower values (slightly) reduce CPU load.

-UA<num> -UI<num>
Controls AMBE and IMBE unvoiced speech levels; lower levels reduce "underwater" sounds.

-e
Use to enable auto-muting of encrypted voice traffic



Advanced Decoder Options
------------------------

Fine tuning the advanced decoder options can greatly increase decoding rates.

Different systems, protocols, receivers and PC sound devices require unique
fine tuning values.

A 15 to 60 second recording of a target system should be made and used as the input
for tuning runs. Use the 'R' key to make recordings.
Recordings of voice, control or rest channels are all useable.

To speed up the tuning process, audio synthesis should be disabled
and the protocol should be locked correctly:

DSDPlus -o0 -O NUL -f1 ? rawAudio.wav

A decoding score will be displayed.

Pressing the up arrow will redisplay the previous command line,
which you can then edit and re-run.

Adjust a single parameter to determine which value produces the highest score:

DSDPlus ? rawAudio.wav -o0 -O NUL -f1 -dr1
DSDPlus ? rawAudio.wav -o0 -O NUL -f1 -dr2
DSDPlus ? rawAudio.wav -o0 -O NUL -f1 -dr3
DSDPlus ? rawAudio.wav -o0 -O NUL -f1 -dr4

When the optimal value for a tuning parameter is determined,
use that value (#) and add another parameter and repeat the tuning steps:

DSDPlus ? rawAudio.wav -o0 -O NUL -f1 -dr# -dh1
DSDPlus ? rawAudio.wav -o0 -O NUL -f1 -dr# -dh2
DSDPlus ? rawAudio.wav -o0 -O NUL -f1 -dr# -dh3
...

Repeat until all advanced decoding options have been fine tuned.


The recommended order for adjusting tuning parameters is:

1: Rolloff filtering (-dr)
2: Hotspot size (-dh)
3: Scaling factor (-ds)
4: Damping level (-dd)

Viewport tuning is rarely worth bothering with.

Scaling and Damping settings do not affect D-Star or ProVoice,
so don't bother tweaking them for those protocols.


You do NOT have to try every value for a tuning parameter:

Rolloff: start at 1 and go up by 1 until the score starts trending down

Hotspot: most signals prefer an even hotspot size (usually 2, sometimes 4);
a few signals prefer an odd hotspot size; testing 1, 2, 3 and 4 will
settle the odd/even question; continue until the score trends down

Scaling: test only 55, 60, 65, 70, 75; best is usually in the 60-65 range

Damping: start testing at 5; increment by 5 or 10; watch the trend...


Check your tuning by re-enabling voice synthesis:

DSDPlus -f1 -dr# -dh# -ds# -dd# -dv# ? rawAudio.wav


-or-

Just use the third party program - dsdtune.


When the optimal values for a system + receiver + sound input is determined,
consider creating a batch file to store the settings:

Local-PD.bat:

DSDPlus -f1 -dr1 -dh3 -ds66 -dd40 -dv20 -O PD.mp3

DMR.bat:

DSDPlus -fd -d21 -dh2 -ds58 -dd5 -dv20 -O DMR.mp3

Then to monitor a specific system, run its batch file.

If you prefer, instead of batch files, you can create desktop shortcuts.


If you scan multiple systems and protocols with a single receiver,
you can run multiple copies of DSDPlus in parallel with each one
protocol-locked and fine tuned as required. Each copy of the program
should write synthesized audio to separate files.



Active Keys
-----------

?

'?' generates a list of keyboard commands in the event log window.

1
2
3

When monitoring conventional DMR systems, you may want to block voice
synthesis for one timeslot. Press 1 or 2 to enable only one timeslot.
Press 3 to enable both timeslots.

-
Display of the command line parameters in the console window title
is enabled/disabled by pressing the '-' key.

|
The symbol phase display in the console window title
is enabled/disabled by pressing the '|' key.

A/a
Adjusts AMBE unvoiced audio level; lower values reduce "underwater" sounds

B
The display of background events (like LRRP updates) in the event log window
is enabled/disabled by pressing the 'B' key.

E
Use to enable or disable auto-muting of encrypted voice traffic.

F
Use when switching from monitoring a trunking system to a conventional system,
for example from DMR Con+ to DMR conventional.

H
Press 'H' in each window to change its high contrast display mode.

I/i
Adjusts IMBE unvoiced audio level; lower values reduce "underwater" sounds

N
Press 'N' to force DSD+ to regather and display the current trunking site's
neighbor list in the event log window. Useful for when the neighbor list has
scrolled off the window.

P
Press 'P' to toggle the raw signal polarity. You may need to do this when
decoding X2-TDMA or dPMR signals.

R
'R' starts/stops recording of discriminator audio. Use 'R' to make 60
second source recordings of tuning data. Rename the files to identify
what they are.

S
If the source audio window has been closed, this key will reopen it.

V
This toggles voice call start alerting; DSD+ will beep each time a voice call starts.

W
When the source audio, event log and channel activity windows have been
placed onscreen where you want them, this key will display their current
locations in the event log window. You can copy these locations to
a batch file or shortcut.

Esc
To terminate real time decoding or .wav file processing, press Esc.



Window Title
------------

The window title area shows the command line parameters used (unless -- specified),
symbol tracking/centering, auto-scaling factor, output audio gain,
symbol rate (2400/4800/9600), and protocol.

During raw audio recording, "<REC>" is displayed.



Data files
----------

DSDPlus.networks

You can populate the DSDPlus.networks file with the network IDs and
network names for NEXEDGE, Connect Plus and Tier III trunking systems.
DSD+ will display the network names when those networks are monitored.
You can add or edit network entries in this file while DSD+ is running.

DSDPlus.sites

The names for each site on a network can be stored in this file.
Some non-networked DMR systems also broadcast system IDs,
so entries for them can also be added to this file.
DSD+ will use the contents of this file to display the name of the
currently monitored site as well as the names of sites in neighbor lists.
You can add or edit site entries in this file while DSD+ is running.

DSDPlus.groups

DSD+ will auto-populate this file with every group ID that is seen.
You can edit this file while DSD+ is running and add names/aliases to
group records.

DSDPlus.radios

DSD+ will auto-populate this file with every radio ID that is seen.
You can edit this file while DSD+ is running and add names/aliases to
radio records. This file replaces the DSDPlus.aliases file; if you
already have a large aliases file, you can use a text editor's
search/replace functionality to convert the contents of your aliases file
to match the format used in the radios file.

NOTE:

Radio aliases are auto-generated on NEXEDGE systems. DSD+ marks auto-generated
NEXEDGE radio aliases in the DSDPlus.radios file by prepending an asterisk like so:

NEXEDGE, ... yyyy/mm/dd hh:mm, *"aliastext"

If you edit a NEXEDGE alias, you must remove the asterisk; this tells DSD+ that
the new alias text is NOT auto-generated and DSD+ will not replace it with OTA alias text.


DSDPlus.frequencies

DSD+ uses this file to display frequency information when DSD+ is monitoring
a rest channel or control channel. The frequency records will also be
used to control channel steering for trunked voice following.

Note:

DSD+ uses two channel numbers for each DMR RF channel:

Channel #1 = first RF channel, timeslot 1
Channel #2 = first RF channel, timeslot 2
Channel #3 = second RF channel, timeslot 1
Channel #4 = second RF channel, timeslot 2
Channel #5 = third RF channel, timeslot 1
Channel #6 = third RF channel, timeslot 2
...

For all DMR systems (DMR, Cap+, Con+, TIII), only one channel record has
to be added to the DSDPlus.frequencies file for each RF channel.
You can use the channel number that corresponds to timeslot 1 or 2 and
DSD+ will use the same frequency information for the other timeslot.


All of the records in these data files have a protocol field;
DSD+ recognizes the following protocol name strings:

D-Star
IDAS
NEXEDGE48
NEXEDGE96
dPMR
DMR
Cap+
Con+
TIII
XPT
P25
ProVoice



DMR TIII handling
-----------------

Tier III control channels broadcast a 14 bit identifier that indicates
the network model (tiny/small/large/huge), network ID, service area
and site number for the current site and for neighboring sites.

Many TIII DMR systems are set up using these programming defaults:

large network (uses a 4 bit NID field)

NID = 13

Service area field length: 5 bits

Site number field length: 3 bits

Physical sites are typically assigned unique area numbers (1, 2, 3, ...)
while their site numbers are all set to 1. So odds are good that any network
you find will have sites with area.site values of 1.1, 2.1, 3.1, etc.

When -v3 or higher is used, DSD+ will display a site's 14 bit "SysCode" in binary.

Example:

CSBK Aloha SysCode=10.1100.00010000

The first two bits encode the network model value. Here, 10 = large model.

The next set of digits encodes the network ID. A zero value is used for NID 1,
so here, 1100 = 12 = NID 13.

The last set of digits encodes the area number and the site number.
Since these two fields do not have a fixed length, DSD+ cannot automatically
decode them. The dividing line between the two fields is selected when the
network is created. This is similar to the variability found in Motorola Type I
fleetmaps.

To determine the correct field sizes, gather as many SysCodes as possible
by monitoring system sites and examining their neighbor lists.

Example:

CSBK Bcast SysCode=10.1100.00010000 Neighbor SysCode=10.1100.00000000, CC=600
CSBK Bcast SysCode=10.1100.00010000 Neighbor SysCode=10.1100.00001000, CC=622

From this small sample we have these SysCodes:

10.1100.00000000
10.1100.00001000
10.1100.00010000

It becomes clear that the SysCodes should be decoded as:

10.1100.00000 000 Model=large NID=13 Area=1 Site=1
10.1100.00001 000 Model=large NID=13 Area=2 Site=1
10.1100.00010 000 Model=large NID=13 Area=3 Site=1

So here, the area length is 5 and the site length is 3.

In the DSDPlus.networks file, an area length value can be appended to TIII
network records, so if the following line is added

TIII, 13, "network name goes here", 5

DSD+ will use the supplied area length value to properly decode this system's
SysCode fields.

These TIII sites can be added to the DSDPlus.sites file as:

TIII, 13, 1.1, "site name goes here"
TIII, 13, 2.1, "site name goes here"
TIII, 13, 3.1, "site name goes here"

The records in the DSDPlus.frequencies file also reference site numbers;
for TIII sites, use the same area.site format:

TIII, 13, 1.1, 600, 462.0, 0.0, 0
TIII, 13, 2.1, 622, 462.3, 0.0, 0



DSD+ Fast Lane
--------------

Early access to features is being offered through the DSD+ Fast Lane program.

Fast Lane updates are expected to be released about once per month.

Some Fast Lane updates WILL have issues/bugs. That is the nature of alpha software.

Fully tested public releases will continue to be released, approximately every 6 months.


In light of the extra workload the Fast Lane program will create,
the DSD+ team is asking Fast Laners for:

US$10 for one year of Fast Lane updates

US$25 for unlimited Fast Lane updates

Donations above these amounts are welcomed, but it's up to you.
We're not looking to get rich here. Funds will be used for things like
needed hardware upgrades.

Funds can be sent our Paypal account (dsdplusfastlane@gmail.com)

Please include a comment that specifies the email address that
your Fast Lane updates should be sent to.

NOTE: IF YOU DO NOT INCLUDE A COMMENT, PAYPAL DOES NOT SEND US A NOTIFICATION.
THIS *WILL* DELAY YOUR FIRST FAST LANE UPDATE.

--

 

[caphab1]

Based on your screenshots, it looks like you have it configured properly. However, on the Windows recording devices tab, it shows audio playing on Line 1 Virtual Audio Cable. Do you have something else routed through that?

No not that I am aware of. It would appear that CC audio is going thro that because sometimes the CC audio will intermitently go thro my headphones and the audio bar from the line 1 will go to zero but will resume when the audio stops in my headphones.

Are you hearing any of the analog traffic? Are you seeing anything in the DSD+ source audio window? How busy is the site?

Not hearing analog when I have UT up, no source audio. Very.. I see the data, whos tx'ing on what channel but never hearing any voice.. just CC.

 

[AggieCon]

Is your Voice VCO parked on the control channel? If so, that explains the control channel output you are hearing. Your screen grab was still a little difficult to see. If you could post it in full resolution next time, that would be of use.

I still do not understand what you have set to output to Line 1 Virtual Audio Cable.

Also, I'm interested in how you obtained your operating system. Perhaps that is part of the problem.

 

[caphab1]

This system came from someone i knew years ago, this just started.. I disabled the Line 1 all together. I am receiving the analog stuff on this system but none of the digital stuff, just getting CC audio. I set up the BAT file like you said and removed the park freq from the Voice VCO. Im not getting any movement on the source audio from the configured BAT file ither.

 

[AggieCon]

Can you upload a new screen grab with DSD+ command prompt, the Call History Window, and the Info and VCO tabs for each of the receivers?

Have your tried the P25 system again? See if you can get it to work. Just change your Signal VCO park frequency.

 

[caphab1]

Can you upload a new screen grab with DSD+ command prompt, the Call History Window, and the Info and VCO tabs for each of the receivers?

Have your tried the P25 system again? See if you can get it to work. Just change your Signal VCO park frequency.

So Im dumb founded. I silenced the two analogs from the moto system, and shortly there after the digital for MCSO Disp started working. I went over to a couple other control freqs on the other p25 sys. and those worked flawlessly as well. I was working intermittently with SDR#, with the DSD plugin, so I couldve messed with a dsd setting then and made it work... I am not sure.

Either way, I am exuberantly thankful for your wealth of knowledge and willingness to assist!!

I will post an updated screen grab when I get home this eve. I apologize for the quality. I print screen and put it in paint jpg or an outside app that does the same thing.. but seems to have better quality..

 

[AggieCon]

The screen grab quality is related to the RR attachment function. Post the image to Facebook, Dropbox, Twitter, Google, or whatever and then use the URL to post it here.

Looking forward to your update.

 

[caphab1]

The screen grab quality is related to the RR attachment function. Post the image to Facebook, Dropbox, Twitter, Google, or whatever and then use the URL to post it here.

Looking forward to your update.

https://www.dropbox.com/s/p4uf2tiorn6z4qc/Image 6.jpg?dl=0

 

[AggieCon]

Looks good! Up your Voice VCO squelch to ~60. You'll appreciate it if you catch an analog call.

 

[caphab1]

Thanks again Aggie!! You have been a great help and this forum is thankful for providing such knowledge so willingly and friendly like! See you around RR cyber space!

Preston