i’m experiencing a strange problem. I have an asterisk instance acting also as SIP client, so i can establish a call between a softphone and Asterisk. Asterisk, in this scenario, uses channel ALSA. On Asterisk side i have a bad voice quality, while the voice quality is good on sofphone side. I started to make some tests. I generated a 700HZ tone and sending it through the call, on Asterisk side i get the 700HZ tone plus a 3KHZ tone. Recording the audio in dialplan 3KHZ tone disappeared. In addition, also playing recorded audio with aplay 3KHZ tone is not there. So i think the problem is about the way Asterisk passes PCM samples to alsa layer or a missing configuration for alsa.
Asterisk version: 13.7.2
sample rate 8000