Fusionpbx. Can't hear my voice on incoming call. Why?

Question

FreeSWITCH

Setting up Fusionpbx for the office.
Only incoming from the beeline is needed.
Trunk rose without problems.
With an incoming call through the trunk, I hear the subscriber, but he does not hear me.
Beeline technical support sent a response:

A check was carried out and a conclusion was given by the operation on the dump taken during the testing process

: in this call scenario, it is directed to the SIP device and to the mobile. network, an invite with a=sendonly to SDP is sent to the SIP device. Further, after the call is accepted on the SIP device, a re-invite without an offer (SDP) is sent to its side, in the 200 OK response to which a=sendrecv is expected in SDP according to RFC6337 (see below).

Instead, a=recvonly comes from the client equipment in this 200 OK. Because of this, the call hangs in hold.

It is necessary that in response to a re-invite without an offer, 200 OK should come with a=sendrecv in SDP.

Where this parameter is configured I can not understand.
Found this line on google:

<param name="NDLB-sendrecv-in-session" value="true"/>

Answer 1

Solution:
In the incoming route, write sip_unhold_nosdp=true

Answer 2

It looks like your freeswitch is behind NAT
Static IP
https://docs.fusionpbx.com/en/latest/additional_in...