In this page, We will look into the one way audio issue in Asterisk.

Description

We will be face a one side audio issue , When we are doing a outbound calls. While making  a phone calls from our application, We will be able to hear the prospect voice but That prospect will not be able to to hear us. Maximum of asterisk users will be face this kind of issues.

Fixes 

This issue is happening bcoz of some audio encoder modules are not loaded properly or we didn't allowed the module to the end points.

1. Open the asterisk server console

2. Run this command - modules show like codec

 Above command will be list down the Codec modules. Find Codec Opus module whether it's loaded or not.

If the module is in the list:

• In this case module is loaded properly but it's not used to the endpoints.
• Open /etc/asterisk/pjsip.conf
• allow opus encoder to the endpoint

Example:

allow=opus,ulaw,gsm,alaw

If the module is not in the list:

• In this case, Codec opus module is not loaded. We have to run it manually.
• Run the below commands

1. sudo  wget http://downloads.digium.com/pub/telephony/codec_opus/asterisk-17.0/x86-64/codec_opus-17.0_current-x86_64.tar.gz 
2. sudo tar -xvzf codec_opus-17.0_current-x86_64.tar.gz
3. cd codec_opus-17.0_current-x86_64
4. cp codec_opus.so /usr/lib/asterisk/modules/
5. cp codec_opus_config-en_US.xml var/lib/asterisk/documentation/thirdparty
6. Open Asterisk CLI - core load module codec_opus

The codec module will be loaded. This will resolve the One way audio issue.

Posts related to Asterisk:

1. Asterisk server setup  -  https://e2ehiring.com/blogs/asterisk%20server%20setup-151

2. How to make the phone calls from your own application using JS and Asterisk  -  https://e2ehiring.com/blogs/how%20to%20make%20the%20phone%20calls%20from%20your%20own%20application%20using%20js-619