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Support building a freepbx system

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@quynhngo wrote:

Can you tell me? I am building a freepbx system. Currently I am having the following problems.

  • for any number that calls out.
    -for any number called internally.
  • Prohibit calling into IP phone numbers.
  • Block numbers calling to other countries.
  • Limited time to call out.
  • record the call.

The features do not charge and charge is just for me. And the free features guide me.
Thanks very much.

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FreePBX security concern

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@vcba79 wrote:

I just opened my firewall for voip test and saw many log like showed below:

[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] pbx.c: Executing [0019146423112952@from-sip-external:1] NoOp("PJSIP/anonymous-00000164", "Received incoming SIP connection from unknown peer to 0019146423112952") in new stack
[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] pbx.c: Executing [0019146423112952@from-sip-external:2] Set("PJSIP/anonymous-00000164", "DID=0019146423112952") in new stack
[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] pbx.c: Executing [0019146423112952@from-sip-external:3] Goto("PJSIP/anonymous-00000164", "s,1") in new stack
[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] pbx_builtins.c: Goto (from-sip-external,s,1)
[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] pbx.c: Executing [s@from-sip-external:1] GotoIf("PJSIP/anonymous-00000164", "1?setlanguage:checkanon") in new stack
[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] pbx_builtins.c: Goto (from-sip-external,s,2)
[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] pbx.c: Executing [s@from-sip-external:2] Set("PJSIP/anonymous-00000164", "CHANNEL(language)=en") in new stack
[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] pbx.c: Executing [s@from-sip-external:3] GotoIf("PJSIP/anonymous-00000164", "1?noanonymous") in new stack
[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] pbx_builtins.c: Goto (from-sip-external,s,5)
[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] pbx.c: Executing [s@from-sip-external:5] Set("PJSIP/anonymous-00000164", "TIMEOUT(absolute)=15") in new stack
[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] func_timeout.c: Channel will hangup at 2020-07-23 23:11:31.040 CST.
[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] pbx.c: Executing [s@from-sip-external:6] Set("PJSIP/anonymous-00000164", "receveip=pjsip,remote_addr") in new stack
[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] pbx.c: Executing [s@from-sip-external:7] Log("PJSIP/anonymous-00000164", "WARNING,"Rejecting unknown SIP connection from 156.96.117.183:58154"") in new stack
[2020-07-23 23:11:16] WARNING[12793][C-0000015d] Ext. s: "Rejecting unknown SIP connection from 156.96.117.183:58154"
[2020-07-23 23:11:16] VERBOSE[12793][C-0000015d] pbx.c: Executing [s@from-sip-external:8] Answer("PJSIP/anonymous-00000164", "") in new stack

I have several questions:

  1. Is 156.96.117.183 tried to connect my freepbx but freebpx did not serve this ip address?
  2. I have several subnet defined in local networks under general sip settings tab. Every sip client from local network can use outbound route to reach remote sip device and any other client not listed in local network will be rejected by freebpx. Is this right?
  3. sip device must register in extensions module before that got served by freepbx. any sip device not register to freepbx will not be served even they are in local network. Is this right?
  4. what about the anonymous appeared in log? I do not see any information in freepbx wiki. Will freepbx serve anonymous sip cient?

Sorry, I am just a freepbx newbee who is trying to figure out these important security issue. Any help would be appreciated.

Thanks,

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Freepbx Gui not reloaded

Need to migrate EPM to another deployment

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@faisalkhan wrote:

HI guys,

I have EPM installed on one of my Freepbx deployment but I need to assign this module to my another Production Deployment.

How can I do that?

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FrleePBX 15 Asterisk 16.9.0 not showing any logs on asterisk console

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@faisalkhan wrote:

hi all,

I am observing something strange which I need to share with you guys.

Freepbx 15 asterisk 16.9.0 console is not showing any logs of the calls when I connect to asterisk console with the command “asterisk -rvvvv”

asterisk -rvvvv

Asterisk 16.9.0, Copyright © 1999 - 2018, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 16.9.0 currently running on 01 (pid = 37298)
01CLI>
01
CLI>
01CLI>
01
CLI>

however I have live traffic of calls and even If I tried to make a call, call went through fine but no logs appearing on the CLI.

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Restrict users to access my Freepbx admin Page

Force https - Tampered file .htaccess

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@idscomm wrote:

Hello,

I was doing so research to force freepbx to load htts instead of http (both admin and ucp) and found a topic where some suggested to add the following to /var/www/html/.htaccess

RewriteEngine On
RewriteOptions inherit
RewriteCond %{HTTPS} off
RewriteCond %{SERVER_PORT} 80
RewriteCond %{REQUEST_URI} !^/.well-known
RewriteCond %{REQUEST_URI} !^/.freepbx-known
RewriteRule ^(.*)$ https://%{HTTP_HOST}%{REQUEST_URI} [L,R=301]

Now, it works for me but FreePBX tells me that I have a tampered file and every time there is an update, my modifications are overwritten. Any suggestion on how to accomplish this without breaking FreePBX?

Thanks in advance!

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Some Certificates are Expiring

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@vaibhav wrote:

I’m seeing 2 errors in FreePBX Dashboard – screenshot attached.
image

My freepbx portals’s SSL certificate shows expiring in Oct 2020, so that’s fine.

I’m unable to understand what the issue is & which PBX Services do I need to restart? the whole server as in reboot or something else?

Thanks

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UDPTL.conf

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@voipgenius wrote:

Where is fax gateway getting it’s settings from? It appears to be setting it’s own datagram size and ports rather than those listed in the udptl.conf, sip.conf, res_fax.conf

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Installing centos packages

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@rilliam wrote:

I am trying to install browsh and it seems like yum is looking at the wrong place or doesn’t have the centos repository? Is there some way to include the centos repository?

                                  x86_64                         1.4.2-2.el7                                 sng-base                            134 k

libepoxy x86_64 1.5.2-1.el7 sng-base 211 k
liberation-fonts-common noarch 1:1.07.2-16.el7 sng-base 27 k
liberation-sans-fonts noarch 1:1.07.2-16.el7 sng-base 279 k
libgusb x86_64 0.2.9-1.el7 sng-base 40 k
libsoup x86_64 2.62.2-2.el7 sng-base 411 k
libwayland-cursor x86_64 1.15.0-1.el7 sng-base 20 k
libwayland-egl x86_64 1.15.0-1.el7 sng-base 13 k
libxkbcommon x86_64 0.7.1-3.el7 sng-base 108 k
mozilla-filesystem x86_64 1.9-11.el7 sng-base 5.3 k
rest x86_64 0.8.1-2.el7 sng-base 63 k
startup-notification x86_64 0.12-8.el7 sng-base 39 k
xcb-util x86_64 0.4.0-2.el7 sng-base 16 k
xkeyboard-config noarch 2.24-1.el7 sng-base 834 k
Updating for dependencies:
cairo x86_64 1.15.12-4.el7 sng-base 741 k
nspr x86_64 4.21.0-1.el7 sng-base 127 k
nss x86_64 3.44.0-7.el7_7 sng-base 854 k
nss-softokn x86_64 3.44.0-8.el7_7 sng-base 330 k
nss-softokn-freebl x86_64 3.44.0-8.el7_7 sng-base 224 k
nss-sysinit x86_64 3.44.0-7.el7_7 sng-base 65 k
nss-tools x86_64 3.44.0-7.el7_7 sng-base 528 k
nss-util x86_64 3.44.0-4.el7_7 sng-base 79 k

Transaction Summary

Install 1 Package (+25 Dependent packages)
Upgrade ( 8 Dependent packages)

Total size: 126 M
Total download size: 103 M
Is this ok [y/d/N]: y
Downloading packages:
No Presto metadata available for sng-updates
No Presto metadata available for sng-base
cairo-gobject-1.15.12-4.el7.x8 FAILED
http://sng7.com/os/7.6/os/x86_64/Packages/cairo-gobject-1.15.12-4.el7.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.
To address this issue please refer to the below wiki article

https://wiki.freepbx.org/display/PPS/YUM+Errors

If above article doesn’t help to resolve this issue please use https://issues.freepbx.org/

cairo-1.15.12-4.el7.x86_64.rpm FAILED
http://sng7.com/os/7.6/os/x86_64/Packages/cairo-1.15.12-4.el7.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.
colord-libs-1.3.4-2.el7.x86_64 FAILED
http://sng7.com/os/7.6/os/x86_64/Packages/colord-libs-1.3.4-2.el7.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.
gsettings-desktop-schemas-3.28 FAILED
http://sng7.com/os/7.6/os/x86_64/Packages/gsettings-desktop-schemas-3.28.0-3.el7.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.
firefox-68.9.0-1.el7.centos.x8 FAILED
http://sng7.com/os/7.6/updates/x86_64/Packages/firefox-68.9.0-1.el7.centos.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.
gtk3-3.22.30-5.el7.x86_64.rpm FAILED
http://sng7.com/os/7.6/os/x86_64/Packages/gtk3-3.22.30-5.el7.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.
libxkbcommon-0.7.1-3.el7.x86_6 FAILED
http://sng7.com/os/7.6/os/x86_64/Packages/libxkbcommon-0.7.1-3.el7.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.
nspr-4.21.0-1.el7.x86_64.rpm FAILED
http://sng7.com/os/7.6/os/x86_64/Packages/nspr-4.21.0-1.el7.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.
nss-3.44.0-7.el7_7.x86_64.rpm FAILED
http://sng7.com/os/7.6/os/x86_64/Packages/nss-3.44.0-7.el7_7.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.
nss-softokn-3.44.0-8.el7_7.x86 FAILED
http://sng7.com/os/7.6/os/x86_64/Packages/nss-softokn-3.44.0-8.el7_7.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.
nss-softokn-freebl-3.44.0-8.el FAILED
http://sng7.com/os/7.6/os/x86_64/Packages/nss-softokn-freebl-3.44.0-8.el7_7.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.
nss-sysinit-3.44.0-7.el7_7.x86 FAILED
http://sng7.com/os/7.6/os/x86_64/Packages/nss-sysinit-3.44.0-7.el7_7.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.
nss-tools-3.44.0-7.el7_7.x86_6 FAILED
http://sng7.com/os/7.6/os/x86_64/Packages/nss-tools-3.44.0-7.el7_7.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.
nss-util-3.44.0-4.el7_7.x86_64 FAILED
http://sng7.com/os/7.6/os/x86_64/Packages/nss-util-3.44.0-4.el7_7.x86_64.rpm: [Errno 14] HTTP Error 404 - Not Found ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.

Error downloading packages:
cairo-gobject-1.15.12-4.el7.x86_64: [Errno 256] No more mirrors to try.
gtk3-3.22.30-5.el7.x86_64: [Errno 256] No more mirrors to try.
nss-util-3.44.0-4.el7_7.x86_64: [Errno 256] No more mirrors to try.
nss-3.44.0-7.el7_7.x86_64: [Errno 256] No more mirrors to try.
colord-libs-1.3.4-2.el7.x86_64: [Errno 256] No more mirrors to try.
libxkbcommon-0.7.1-3.el7.x86_64: [Errno 256] No more mirrors to try.
firefox-68.9.0-1.el7.centos.x86_64: [Errno 256] No more mirrors to try.
nss-softokn-freebl-3.44.0-8.el7_7.x86_64: [Errno 256] No more mirrors to try.
gsettings-desktop-schemas-3.28.0-3.el7.x86_64: [Errno 256] No more mirrors to try.
nss-softokn-3.44.0-8.el7_7.x86_64: [Errno 256] No more mirrors to try.
nss-sysinit-3.44.0-7.el7_7.x86_64: [Errno 256] No more mirrors to try.
nss-tools-3.44.0-7.el7_7.x86_64: [Errno 256] No more mirrors to try.
cairo-1.15.12-4.el7.x86_64: [Errno 256] No more mirrors to try.
nspr-4.21.0-1.el7.x86_64: [Errno 256] No more mirrors to try.

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Phone Registers but FreePBX wont

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@brian4 wrote:

In visiting Mynetfone (Australian VSP) I discover that they have an outage beginning 8:00am yesterday and the site has only just been update stating that the ETA on the fix will be 4:00pm today.

I noticed our trunks lost registration yesterday and I assume that MNF outage has caused this however I am not 100% sure.

We have made no modifications to our FreePBX settings, we just lost registration however I can register a trunk directly via 1 of our Yealink T46G’s.

Registration via the FPBX times out.

Question: Does it make sense or is it possible that I can register a trunk via the phone itself but not be able to register via the FreePBX box (which of course was registered prior to yesterdays reported outage.)

Cheers

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Some Certificates are Expiring

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@vaibhav wrote:

I’m seeing 2 errors in FreePBX Dashboard – screenshot attached.
image

My freepbx portals’s SSL certificate shows expiring in Oct 2020, so that’s fine.

I’m unable to understand what the issue is & which PBX Services do I need to restart? the whole server as in reboot or something else?

Thanks

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Inound calls on IVR

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@amardeeprana wrote:

Hi

I monitor my inbound calls after weekend. I see 1-2 calls land to IVR and end. these are long duration calls. like 15 mins, 45 mins. But i dont see any Desination.

It is my main number which lands to IVR. no more options.

Thanks
Amardeep

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No call progress output to CLI

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@aroundi wrote:

Hello,

I am no longer seeing call progress messages in the CLI, for example when making an inbound call to the system. This happened after bulk updating the Freepbx modules. I don’t have the complete module list, I would say at 30 or so were updated.

The update appeared to be successful and the system appears to be working properly. I am able to launch commands from the CLI, i.e. ‘sip show channels’. Call info does appear in the asterisk logs.GUI appears okay.

Asterisk ver 16.8.0
Freepbx ver 15.0.16.72.1

I’m not sure where to begin troubleshooting. Does anyone have experience with a problem like this?

I appreciate any help / pointers.

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Freepbx Arabic Sound and language

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@csambec wrote:

Dear All,

Anyone can share Arabic Sound and language package for Freepbx/Asterisk
If not available, is there any plan to develop such feature?

BR
Samir

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Problem adding a recording

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@covici wrote:

Hi. I am trying to add a system recording to freepbx – I have version15.0.16.38 and this is a pretty new install under Debian 10.

When I try to add a system recording the system complains that sound language is not set and when I go to sound languages I get the following error:
Unable to get online sound packages. Error was: [2] Use of undefined constant CURLOPT_RETURNTRANSFER - assumed ‘CURLOPT_RETURNTRANSFER’ (this will throw an Error in a future version of PHP)

I have asterisk and it does have sound files, so I am a bit confused as to what to do here.

Thanks in advance for any suggestion.

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SBC SIP timeouts for several minutes

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@oli170 wrote:

Hi,

since a few days we have massive problems with our Sangoma SBC: From time to time we get timeouts and calls fail, in and out. It happens very often - it works for a few minutes, then we get timeouts for some minutes, then it works again.
With tcpdump I see, that the SBC is sending “Status 100 Trying” but there is no following “Status 200 OK” within time, so the trunk peer sends a CANCEL with reason q.850 (cause 27, destination out of order) after 14 sec and the call is aborted.
After some minutes without any change it works again, “STATUS 200 OK” is sent right after “Status 100 Trying”.

Don’t know what happens here, because I don’t see any packet with “STATUS 200 OK” in wireshark I think it has to be a problem in the SBC itself, not regarding to any possible network issues?

Thanks.

Best regards
Oliver

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ERROR[3482]: manager.c:6552 process_message: Unable to process manager action 'Filter'. Asterisk is shutting down

Sound work in both directions, but no announcement or ivr audio

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@z3usone wrote:

Hi

I have freepbx 15 on raspberry pi 4.

All seems to work fine, I’m on messagenet sip trunk configured like they says on official website.

I can receive and make call, I have audio, extension work, i can record and play systems recordings.

But… Randomly, for no reason, if I try to call the pbx number, there are two situations

Sometimes nothing appens, the call stay on old, I don’t know how to explain. Like the call don’t arrive to trunk. Because nothing appens to log file, no new lines, no record, no nothing. I can’t find the source of the problem and any solution. After a while, playing whit phone or reloading the configuration, it start to work again.

The other problem is in most of the case related, or appens in the same times. When i call from external, the pbx respond, the log file start to “log”, and some announcement start to play, but… No sound! The strange thing is the log file say nothing wrong, and (I know the times and the ivr choose) if I press the right buttons, the extension ring normally and I can talk!

I already play a bit with codes, but as a newbie, maybe somting are missing.

Thanks to all that can help me to solve this problem.

Joe

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[bug?, workaround] MIXMON_POST - [GUI] variables/parameters not passing

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@POTShead wrote:

I’m a ‘prosumer’ home user with an updated v15.0.16.72 system for my single-trunk, 6-xtn setup, living a set-and-forget life (except for yesterday when I forced all module and system updates, before confirming my observed problem.)

On May 21 at ~18h00 EDT [-0400] something oddish happened - my MIXMON_POST script stopped being given its usual helping of parameters. (I use this post-call script to upload the newly-created call recording to a storage server.)

In fact it worked properly at 18:01 of that day, and 6 minutes later it was then getting no parameters… and this no parameters problem has persisted through multiple reboots and until yesterday when I got down to finally fixing it.

I normally would set my desired script and its parameters via the GUI - Settings/AdvancedSetttings and then actual setting: PostCallRecordingScript [MIXMON_POST variable], and thusly:

/usr/local/bin/upload_newrec ‘^{ARG2}’ ‘^{ARG3}’ ‘^{FROMEXTEN}’ ‘^{TIMESTR}’ ‘^{UNIQUEID}’ ‘^{CALLERID(name)}’ ‘^{CALLFILENAME}’ ‘^{MIXMON_FORMAT}’

but no matter what fiddling I did with carets, dollarsigns, single-or-doublequotes I could not get parameters passed to the script (the script would fire but always with no parameters).

So my apparently-observed bug is that updating this param to send variables does not seem to send along the variables.

I eventually cottoned on to the fact that I could just attack the cfgfiles directly (but doubtless at my own risk… next time I hit Submit on the GUI likely), and so I was able to restore my original and desired behaviour via ensuring that each of two files has this in its config for the MIXMON_POST variable:

MIXMON_POST = {{see above, right after ‘thusly’}}

and those two files being:

- /etc/amportal.conf
- /etc/asterisk/extensions_additional.conf

Curiously, even when those two files have the desired setting, what is displayed in the GUI as being the MIXMON_POST setting is unrelated (seems to be a one-way write, from GUI to cfgfiles?)

So before I go and log an ‘issue’ over this apparent problem I thought that I’d see if it was unique to me or if anyone had any other insight on what I might feed the GUI to get my desired results?

Cheers & thx!

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