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!
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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
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