Hi all.
As title says, my phone is not recording microphone input during the calls. Other party will not hear anything while call is ongoing.
This is kind of weird, as I would swear that it was working fine the first day I got the phone.
I flashed the phone with latest Byzantium image, using the info from docs, but no change whatsoever.
I found a workaround though. If I go to Settings -> Sound during the call, and change Input Device to “Analog Input - Modem”, it will start working. The problem is that I need to do this every call, and I also need to increase volume every time.
Does anyone have the same issue? I searched the forums and issues on GitLab, but haven’t found anything.
I found this in journalctl log during the calls, I believe it’s relevant:
Dec 03 16:26:14 pureos ModemManager[693]: [modem1/call9] user request to accept call
Dec 03 16:26:15 pureos ModemManager[693]: [modem1/call9] call is accepted
Dec 03 16:26:15 pureos ModemManager[693]: [modem1/call9] call state changed: ringing-in -> active (accepted)
Dec 03 16:26:15 pureos feedbackd[924]: Tried to end non-existing event 298
Dec 03 16:26:15 pureos pulseaudio[3626]: Configured maximum latency is smaller than latency, using latency instead
Dec 03 16:26:15 pureos ModemManager[693]: [modem1] unexpected incoming call to number ‘+XXXXXXXXXXXX’ reported in call list: state active
Dec 03 16:26:15 pureos pulseaudio[3626]: Configured latency of 200.00 ms is smaller than minimum latency, using minimum instead
Dec 03 16:26:15 pureos pulseaudio[3626]: Cannot set requested sink latency of 60.75 ms, adjusting to 100.00 ms
Dec 03 16:26:15 pureos pulseaudio[3626]: Cannot set requested source latency of 60.75 ms, adjusting to 100.00 ms
Dec 03 16:26:15 pureos pulseaudio[3626]: Configured maximum latency is smaller than latency, using latency instead
Dec 03 16:26:15 pureos pulseaudio[3626]: Cannot set requested sink latency of 50.00 ms, adjusting to 100.00 ms
Dec 03 16:26:15 pureos pulseaudio[3626]: Cannot set requested source latency of 50.00 ms, adjusting to 100.00 ms
Dec 03 16:26:15 pureos gnome-control-c[3811]: g_utf8_casefold: assertion ‘str != NULL’ failed
Dec 03 16:26:15 pureos gnome-control-c[3811]: g_utf8_casefold: assertion ‘str != NULL’ failed
Dec 03 16:26:15 pureos pulseaudio[3626]: Doing resync
Dec 03 16:26:15 pureos pulseaudio[3626]: Playback after capture (-10624252), drop sink 340016
Dec 03 16:26:15 pureos phosh[903]: Failed to claim proximity sensor: GDBus.Error:org.freedesktop.DBus.Error.ServiceUnknown: The name net.hadess.SensorProxy was not provided by any .service files
Dec 03 16:26:15 pureos gnome-control-c[3811]: g_utf8_casefold: assertion ‘str != NULL’ failed
Dec 03 16:26:15 pureos gnome-control-c[3811]: g_utf8_casefold: assertion ‘str != NULL’ failed
Dec 03 16:26:16 pureos pulseaudio[3626]: Playback too far ahead (78858), drop source 2520
Dec 03 16:26:17 pureos pulseaudio[3626]: Playback too far ahead (10282), drop source 328
Dec 03 16:26:18 pureos pulseaudio[3626]: Playback too far ahead (10134), drop source 324
Dec 03 16:26:19 pureos pulseaudio[3626]: Playback too far ahead (10186), drop source 324
Dec 03 16:26:20 pureos pulseaudio[3626]: Playback too far ahead (10272), drop source 328
Dec 03 16:26:21 pureos pulseaudio[3626]: Playback too far ahead (10067), drop source 320
Dec 03 16:26:29 pureos ModemManager[693]: [modem1/call9] user request to hangup call
Dec 03 16:26:29 pureos ModemManager[693]: [modem1/call9] couldn’t hangup single call with call id ‘1’: Operation not allowed
Dec 03 16:26:29 pureos ModemManager[693]: [modem1/call9] call state changed: active -> terminated (terminated)
My guess is that there’s some issue with PulseAudio configuration, but I’m not familiar enough with it to conclude anything useful.