I use CentOS 6.7 droplet and Asterisk 12.8.2 + FreePBX (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!

  • typically :) I went into that problem too, my problem the UDP port was QOS from the ISP side,
    i had not enough bandwidth for u(a)law, normal calls, u/alaw need something like 100kb/sec workaround was, I’ve bought G729 codec ( 35kb/sec) and changed the ISP, no problem at all any-more.

    Check the logs on the server, located at /var/log/asterisk/full, if there isnt anything bad ( i could be a NAT problem too) I would say, not enough bandwidth.

  • My problem was on the side of the voip-provider as it turned out in the end. Thank you! :)

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1 answer

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

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