Sip incoming call voice silence

Hi here.
Help me please to solve one issue with my pbx.
I have FreePBX, on which I have configured 3 external sip trunks from sip providers. Two of them are working correctly everything but in one I can not hear the voice caller (meaning incaming call). I have also configured one IVR, when a caller calls to one of my 3 numbers it is going to IVR and then connecting to for a specific department by pressing specific buttion. But for this unworking number when pressing the butten during IVR speach nothing goes change, even when I am changing inbound route to the local phone, I can receive call but when I’m picking up the phone the voice of caller doesnt heraring.
Can anyone help me solving this.
Thank you.

Is the PBX behind a NAT router, and if so are you port forwarding the entire RTP port range as defined in Asterisk SIP Settings?

Thank you for your responce.
The PBX is in the local network, I mean behind a NAT router.
I didn’t any port forwarding in my router about PBX NAT never, everything worked correctly.
Please see the attached picture

Is there any suggestions.

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