PJSUA sampling issues with ALSA

I have been trying to make a VOIP calling device with a raspberry pi3 and MIC+ using PJSUA. There seems to be a bug when attempting to make/receive a call.

pjsua-armv7l-unknown-linux-gnueabihf: ../src/pjmedia/conference.c:1895: get_frame: Assertion `frame->size == conf->samples_per_frame * conf->bits_per_sample / 8' failed.

To fix this: change /root/.asoundrc

options snd_rpi_googlevoicehat_soundcard index=0

pcm.softvol {
type softvol
slave.pcm dmix
period_size 320
buffer_size 10240
control{
name Master
card 0
}
}

pcm.micboost{
type softvol
slave.pcm dsnoop
period_size 320
buffer_size 10240
control {
name Micro
card 0
}
min_dB -10.0
max_dB 50.0
resolution 256}

pcm.!default {
type asym
playback.pcm "plug:softvol"
capture.pcm "plug:micboost"
}

ctl.!default {
type hw
card 0
}

If it still does not work then refer to this:
https://community.volumio.org/t/change-the-default-alsa-sample-rate-to-44100/116
http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2010-April/010721.html

PJSIP on Raspberry Pi via ALSA

Download and extract

apt−get install build−essential automake autoconf libasound2-dev
wget https://github.com/pjsip/pjproject/archive/2.10.tar.gz
tar -xvf 2.10.tar.gz
cd pjproject-2.10

Modify config_site.h

#define PJMEDIA_AUDIO_DEV_HAS_ALSA 1

Compile

./configure && make dep && make
/root/pjproject-2.10/pjsip-apps/bin/pjsua-armv7l-unknown-linux-gnueabihf  --config /root/pjproject-2.10/pjsip-apps/bin/pjsip.conf