[App_rpt-users] Echolink connection announcement Volume

[Ken]

Well - I expect you have seen where you can reduce all the "Allison" words
to whatever volume you want, with a single script command.  

I do not know what words might somehow be "hard coded" into the program. I
would make a list of what announcements are bothering you and try to find
those words in this directory:

/var/lib/asterisk/sounds
  or
/usr/src/astsrc/sounds/rpt/
 
and then change a few of them and see if those are what the program uses. I
would be surprised if some words are hard-coded and not changeable.  

For example: 
 	connected.gsm
 	disconnected.gsm
 	connection-failed.gsm
	remote-already-in-this-mode.gsm

I note that these words are contained in BOTH of those directories mentioned
above.  It could be that the system uses words from one directory and the
user is supposed to use words from the other directory.  

One last idea - since the second directory is a source file directory, it
may be that you have to restart or even recompile asterisk after changes are
made, before they will take effect. There are some references to sounds in
the makefile. (The system words may be cached on program startup, so that
they do not need to be repeatedly fetched from disk).

73
Ken



GL
Ken




-----Original Message-----
From: kk6ecm [mailto:kk6ecm at gmail.com] 
Sent: Tuesday, January 05, 2016 2:27 PM
To: Ken <ke2n at cs.com>
Cc: app_rpt-users at ohnosec.org
Subject: Re: [App_rpt-users] Echolink connection announcement Volume

I'll take a look at it. Our issue boils down to the fact all the volumes are
the same, announcements and user voice, and we'd like to have the
announcements about half the volume when the repeater is in use. I'm looking
at holdofftelem=1 as well as http://docs.allstarlink.org/drupal/node/126 to
see if we can achieve a workable compromise.

Thanks,
Bob
kk6ecm

Sent from iPad


On Jan 5, 2016, at 11:11 AM, Ken <ke2n at cs.com> wrote:

>> The audio was setup using the prescribed method, without the "MK-1
> calibrated ear."
>> If we raised the rx level, and reduced the tx level, then we'd have 
>> difficulty hearing the station id. I suspect the level of our custom 
>> recordings is too high, an I was trying to avoid the brute force 
>> method of
> redoing each one.
>> 
>> Thanks,
>> Bob
>> kk6ecm
> 
> Record level is tricky in Asterisk.  When seen on any editing program, 
> "full scale" is -6 dB f.s. or 0.5 on the voltage scale. I suspect the 
> reason for this is that uLaw only produces about 14 bits when 
> converted to linear and that is what asterisk expects for full scale.  
> (I guess losing those two bits is equivalent to dividing by four or 6 
> dB down).  Have a look at some of the Allison GSM files by importing 
> them into Audacity and you will see the max level is 0.5.  Even this 
> is too loud for most people (as newbie postings have indicated many 
> times). I think a reasonable level is 3 dB down from full scale - in 
> other words, minus 9 dB relative to full scale (!) or about 0.3 on the
voltage scale.
> 
> Ref: http://images.ohnosec.org/app-rpt-audacity.pdf
> 
> Regards
> Ken
> KE2N
> 
>