Hello!
I use CentOS 6.7 droplet and Asterisk 12.8.2 + FreePBX 12.0.76.2 (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?
Thank you!
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Please use this as a droplet
Hi You can assist in the installation in my server FREEPBX? my email is adulantoc@gmail.com
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