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.?

TURN candidates not present in SIP headers when using ICE on Asterisk 13