Last active
May 28, 2021 05:04
-
-
Save worldadventurer/757bf4b10af2356d83d57a8a6bb3e4e8 to your computer and use it in GitHub Desktop.
FreePBX Asterisk 13 Install Opus Codec
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
# This is for x64 only | |
# To check x32 vs x64, run: getconf LONG_BIT | |
# Tested on Centos x64 - on a FreePBX Distro 13 x64 | |
echo ------- | |
echo This installs opus codec from the Digium website | |
echo ------- | |
read -rsp $'Press any key to continue OR CTRL-c to QUIT ...\n' -n1 key | |
cd ~ | |
wget http://downloads.digium.com/pub/telephony/codec_opus/asterisk-13.0/x86-64/codec_opus-13.0_1.1.0-x86_64.tar.gz | |
tar -zxf codec_opus-13.0_1.1.0-x86_64.tar.gz && cd codec_opus-13.0_1.1.0-x86_64 | |
cp codec_opus.so /usr/lib64/asterisk/modules/ | |
cp format_ogg_opus.so /usr/lib64/asterisk/modules/ | |
cp codec_opus_config-en_US.xml /var/lib/asterisk/documentation/ | |
# FOR x32 ONLY: | |
#axel -n 8 http://downloads.digium.com/pub/telephony/codec_opus/asterisk-13.0/x86-32/codec_opus-13.0_1.1.0-x86_32.tar.gz | |
#tar -zxf codec_opus-13.0_1.1.0-x86_32.tar.gz && cd codec_opus-13.0_1.1.0-x86_32 | |
#cp codec_opus.so /usr/lib/asterisk/modules | |
#cp format_ogg_opus.so /usr/lib/asterisk/modules | |
#cp codec_opus_config-en_US.xml /var/lib/asterisk/documentation/ | |
echo Restarting Asterisk server | |
service asterisk restart | |
echo Checking Asterisk transcoding support - waiting for Asterisk to finish restarting: | |
sleep 5 | |
asterisk -rx 'core show translation' | |
echo ------- | |
echo If you see opus above then both installed succesfully. Otherwise check the Asterisk log | |
echo ------- |
Probably better to use 'current' to get the latest release:
Except it extracts to 'codec_opus-13.0_$ver-x86_64' and not 'codec_opus-13.0_current-x86_64'
Doh....
Sign up for free
to join this conversation on GitHub.
Already have an account?
Sign in to comment
This worked great, however can't find any info on the internet about how to make extensions like directory services and conferences work. I'm getting a translation error:
channel.c: Unable to find a codec translation path: (slin) -> (opus)
Asterisk 13.17.1
FreePBX 13