no, a pulse config directory is created in /var/lib/asterisk/.config/
the entry for asterisk in /etc/passwd is:
asterisk:x:124:130:Asterisk PBX
daemon,,,:/var/lib/asterisk:/usr/sbin/nologin
Em 08/04/2020 08:49, Iain Mott escreveu:
I suppose a likely cause for the problem is that the asterisk user has
no ~.config/pulse directory. How can I get around that?
Em 08/04/2020 08:38, Iain Mott escreveu:
Hi,
Another Asterisk PBX related question. I have the Asterisk service
periodically sending System commands on the host machine when
particular phone extensions are called. The Asterisk service is run
by the user asterisk. I made asterisk a member of the 'audio' group
so that it can use 'aplay' with the System command. eg.
System(aplay -c 1 -D pulse /home/iain/sounds/line1.wav)
This plays OK but plays on the laptop's internal hardware rather via
PulseAudio. Jack is running and is configured to use an external USB
sound card and the PulseAudio Jack Sink.
If I run the following as user iain, it plays correctly through the USB:
aplay -c 1 -D pulse /home/iain/sounds/line1.wav
However via the Aterisk service, or by running the command in a
console as asterisk, it plays through the internal speakers. eg.
sudo -H -u asterisk bash -c 'aplay -c 1 -D pulse
/home/iain/sounds/line1.wav'
I don't know if the "-D pulse" option is necessary, I'm trying lots
of things...
As well as the audio group, I've tried adding asterisk to the pulse
and pulse-access groups, but it doesn't help (and yes, I logged out
and in again).
If i run the last code given above with aplay in verbose mode (-vvv),
it throws the error:
xcb_connection_has_error() returned true
But this Is not error because adding "unset DISPLAY ; " to the start
of the command removes the error but it continues to play through the
external speakers. Here is the print for the command:
iain@samsung:~$ sudo -H -u asterisk bash -c 'unset DISPLAY ; aplay
-vvvv -c 1 -D pulse /home/iain/sounds/line1.wav'
Tocando WAVE '/home/iain/sounds/line1.wav' : Signed 16 bit Little
Endian, Taxa 48000 Hz, Mono
ALSA <-> PulseAudio PCM I/O Plugin
Its setup is:
stream : PLAYBACK
access : RW_INTERLEAVED
format : S16_LE
subformat : STD
channels : 1
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 24000
period_size : 6000
period_time : 125000
tstamp_mode : NONE
tstamp_type : GETTIMEOFDAY
period_step : 1
avail_min : 6000
period_event : 0
start_threshold : 24000
stop_threshold : 24000
silence_threshold: 0
silence_size : 0
boundary : 6755399441055744000
iain@samsung:~$
Any suggestions please?
Thanks,
Iain
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