Trunk audio problem

I have a FreePBX Distro (latest stable version 10 with Asterisk v13 64bit) and i am using:

  1. SIP Trunk connectivity with VoIP provider (2 Trunks 2 numbers)
  2. 2 Cisco 232D ATA using the FXOs with one POTS and one FXS from VoIP router of my ISP (ZTE VoIP Router)
    and finaly i tried to use (as an alternative of CISCO 232D ATA) an OpenVOX A400E analog card.
    I use G711a codec.
    When i perform an internal call all the partisipants have perfect sound clear and loud.
    When i call an external number using any of the above alternatives i have the same issue: i can hear the other end clear and loud but the other end can’t hear me loud. There is an attenuation of about -9db at least! So i have always complaints from the other end that can’t hear me loud.
    I am using ip phones of Grandstream GXP1628 and Android Zoiper softphone with the same results.
    Is this normal? Is there a reason why the audio is about -9db down when i use any kind of SIP trunk or Dahdi trunk? Is this a hardware related issue?
    Thank you very much in advance!