ICE is not working with Asterisk 13.13.1

I am using an Asterisk 13.13.1 server behind a NAT firewall with FreePBX and I have two chan_sip extensions.
And these extensions are also registered from behind NAT of different networks. I was able to register my extensions in Asterisk.
In order to successful voice communication over RTP, I noticed that it is essential to configure ICE for achieving my needs.
And I did the following steps
I have enabled ICE support for both extensions with FreePBX GUI.
And in RTP configuration file (rtp_custom.conf) I have added STUN and TURN Server Details as follows.


And in res_stun_monitor.conf file I added following configuration


And when I testing the stun status with asterisk CLI (using command: stun show status) , I got the following

Hostname                  Port  Period  Retries  Status  ExternAddr       ExternPort    3478  30      3        OK    37254

When I tested by making phone calls between the extensions I got the sip INVITE packages with only two ice candidates. There is no server reflexive candidate present. The body of request as follows

Content-Length: 527
o=root 788094571 788094571 IN IP4
s=Asterisk PBX 13.13.1
c=IN IP4
t=0 0
m=audio 11498 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=candidate:H22c66e1f 1 UDP 2130706431 11498 typ host
a=candidate:H22c66e1f 2 UDP 2130706430 11499 typ host

And my question is, How to properly configure ICE in order to get all necessary candidates.?