I use CentOS 6.7 droplet and Asterisk 12.8.2 + FreePBX 126.96.36.199 (I install it according to instructions http://wiki.freepbx.org/display/FOP/Installing+FreePBX+12+on+CentOS+6.5). Calls are made with the use of sip-trunk from https://zadarma.com/en/.
When the caller dials the extension number and forwarded to an external mobile number, sometimes occurs a one-way audio. The called party cannot hear the caller.
Do you have some experience in dealing with such problems? What do you recommend in this situation?
This textbox defaults to using Markdown to format your answer.
You can type !ref in this text area to quickly search our full set of tutorials, documentation & marketplace offerings and insert the link!
These answers are provided by our Community. If you find them useful, show some love by clicking the heart. If you run into issues leave a comment, or add your own answer to help others.
Join our DigitalOcean community of over a million developers for free! Get help and share knowledge in Q&A, subscribe to topics of interest, and get courses and tools that will help you grow as a developer and scale your project or business.
Click below to sign up and get $200 of credit to try our products over 60 days!