I searched everywhere on the internet and I didn’t find any solution to fix this ugly issue
My Nat settings are all local I don’t use anything over the internet so the firewall of the router is not the problem and I already disabled the firewall in FreePBX settings
So please if anyone knows the real fix to this problem help me to wake up from this bad dream thank you 
The screenshot shows no SIP traffic. If your gateway trunk is using chan_sip, use
sip set debug on
to enable SIP tracing on chan_sip.
Or, the time range posted did not include the failing call.
After enabling pjsip logger or sip debug, make a failing call and paste the Asterisk log for the call (which will include the SIP trace) at pastebin.freepbx.org. As you are too new to post links, just post the last eight hex characters of the link.
Last eight hex of the link ==> d17b6395
192.168.0.70 is the FreePBX server
192.168.0.122:5060 is the DInstar gateway that I use to make the calls
192.168.0.13 ,192.168.0.6 and 192.168.0.10 is the IP-phone’s I used to make the calls
The number that I called is 0915555336 and the call dropped after 33 seconds I think
There are no calls attempted in that log (no INVITE requests).
So I need to repeat the call again and check the logs until I get an INVITE request ?
If the call connected, Asterisk must have sent an INVITE to the gateway, so if it’s not there, either your logging was not set up correctly (note that both pjsip logger and sip debug are turned off by Apply Config and you would need to reissue them), or you were not looking at the proper time range.
Check this log maybe you will find the INVITE request
https://pastebin.freepbx.org/view/66c70a9b
DESTINATION: 0944423847
Call duration 40 seconds and dropped on 43 seconds
Sorry, but the only occurrences of 0944423847 in your log are on lines 421 and 437, related to a BYE request after the call had ended. Please paste a more complete log.
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