Hello!
I use CentOS 6.7 droplet and Asterisk 12.8.2 + FreePBX 12.0.76.2 (I install it according to instructions wiki.freepbx.org/display/FOP/Installing+FreePBX+12+on+CentOS+6.5). Calls are made with the use of sip-trunk from Zadarma.
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?
Thank you!
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.
Hey there,
Thank you for reaching out to us!
I’d make sure your endpoint (phone) is configured with NAT enabled if it’s behind a router/firewall which it most likely is.
Thank you and please let us know if there is anything else we can do for you.
Happy coding,
Jon Schwenn Platform Support Specialist DigitalOcean
Become a contributor for community
Get paid to write technical tutorials and select a tech-focused charity to receive a matching donation.
DigitalOcean Documentation
Full documentation for every DigitalOcean product.
Resources for startups and AI-native businesses
The Wave has everything you need to know about building a business, from raising funding to marketing your product.
The developer cloud
Scale up as you grow — whether you're running one virtual machine or ten thousand.
Start building today
From GPU-powered inference and Kubernetes to managed databases and storage, get everything you need to build, scale, and deploy intelligent applications.
