[App_rpt] [Fwd: Re: USB radio interface problems]

Hwstar · January 7, 2008, 12:17am

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-------- Original Message --------
From: - Sun Jan 6 16:08:57 2008
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Date: Mon, 07 Jan 2008 01:42:15 +0200
From: Tomi Viikari <Tomi.Viikari@i-isolutions.fi>
To: Stephen Rodgers <hwstar@rodgers.sdcoxmail.com>,
app_rpt@marauder.illiana.net
Cc: Steven Henke <sph@xelatec.com>
Subject: Re: USB radio interface problems
References: <004901c85086$7bb35850$0800000a@tomi> <478134A9.8060209@rodgers.sdcoxmail.com>
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Lainaus Stephen Rodgers <hwstar@rodgers.sdcoxmail.com>:

Tomi,

Here are some things to try--

thanks for quick answer.

i tested some more....

Steve.

Tomi Viikari wrote:
> Hi
>
> i have asterisk (EVB) and appr_rpt installed with usb sound/radio
interface.
> one linksys voip phone as client (sip)
>
> when i dial from sip client to asterisk/rpt in phone mode
> - there is no rx audio to sip client
> - *6 activates PTT but i can'n deactivate it (exept hanging up the phone)
> - tx audio is ok

This is the normal way it works as you are typically use a repeater
(full duplex) for phone traffic.

my configuration is simplex (duplex=0)

-- however--

In the latest app_rpt.c (Version 0.91) there is an experimental simplex
autopatch which is voice-activated (VOX) from the both the autopatch and
the phone control features.

To set it up, set duplex = 0 in rpt.conf for the node, and then place
these values in the node stanza:

;The following parameters are for vox operation(s)
voxtimeout=5000 ; time to let vox talk in ms
voxrecover=500 ; time to take from vox one

timed out in ms

simplexpatchdelay=25 ; # of 20 ms frames to delay audio
simplexphonedelay=25 ; # of 20 ms frames to delay audio

For phone mode, append a v on the end of the rpt call in extensions.conf
like this:

exten=1999,n,Rpt,1999|Pv

This activates the VOX when in phone mode.

Vox works fine.(with no receive audio) (very useful feature to my setup)

in P mode(no vox), when dialin *6 (6=cop,6) it does not activate PTT
until timeout or hangup.
When i enable debug in rpt, and dial first *6 from sip
rpt detects *6 from sip and prints
...
@@@ action cop,6
...
<<radio tx on >>
and ptt activates and so on... (i think "radio tx on" is from chan_usbradio)
when i dial second *6 from sip (to deactivate)
it prints same @@@ action cop,6 but does not print << radio tx off >>
when i dial randomly *,6,# numbers from sip in random order many times
i occasionally get
<< radio tx off>>
<< radio tx on>>
always in pairs so that PTT remains on.

in S mode i can't activate PTT at all.
dialing * from sip i get
<< radio tx on>>
<< radio tx off>>
always in pairs so that PTT remains off

>
>
> some debug
> - app_rpt detects and hears dtmf commands from remote radio
> - test signal (milliwatt) from sip works fine

Are you sure this isn't chan_usb radio reporting the DTMF tones?

you are right it is chan_usb radio that reports DTMF tones

Are you using the same CTCSS tone used during setup?

i'm not using CTCSS at all

Please note that a bug has been reported on losing receive audio on some
systems with the USB FOB after running a system for a while, but in your
case it seems like the audio was never there to start with. This is not
an issue on the pciradio radio interface. Do you initially have receive
audio when you first start Asterisk? This I'd like to know.

no receive audio at all.
i started asterisk always fully when tested (not reload) and i have
never heard
receive audio

> - app_rpt test signal works. ( activates PTT, plays signal (in boths sip
and
> radio), and deactivates ptt)
> - hidthread detects squels (vol dn) (when manualy pressing the vol dn,
> nowired squels to radio)
> - i have tested this with modified usb-adabter and original with
headphones
> (both same result)
> - i tryed configure rpt as dupled and simplex (usbradio.conf) with no
> change

If you are running chan_usbradio with a half-duplex radio. Set it to 0,
otherwise set it to 1. If you set up half-duplex in usbradio.conf, be
sure to use duplex mode 0 or 1 in rpt.conf as well. Both conf files need
to match each other in settings.

half-duplex radio. dublex=0 on both

> - rpt detects dtmf's from sip
> - txtimeout works ( cut's tx after 30 sec)
>
> any ideas to try?