Any calls between extensions hang up after 31 seconds

Hi all,
I installed a FreePbx whitout changes to the standard conifguration, I have just conifgured two internal extensions with 2 phone Grandstream GXP1615. When I try to call the other phone everythings is ok but the call was drop in 30 seconds. I disabled firewall, configuring local LAN address in Sip settings>>NAT Setting as I read in a other post, bu nothing. I think the phone doesn’t receive 200 ACK. Any Idea? Here the log:
[2020-07-31 10:07:23] VERBOSE[7681][C-0000000d] app_dial.c: Called PJSIP/200/sip:[email protected]:5060;user=phone
[2020-07-31 10:07:23] VERBOSE[5391] netsock2.c: Using SIP RTP Audio TOS bits 184
[2020-07-31 10:07:23] VERBOSE[5391] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2020-07-31 10:07:23] VERBOSE[5391] netsock2.c: Using SIP RTP Audio CoS mark 5
[2020-07-31 10:07:24] VERBOSE[7681][C-0000000d] app_dial.c: PJSIP/200-00000019 is ringing
[2020-07-31 10:07:24] VERBOSE[7681][C-0000000d] app_dial.c: PJSIP/200-00000019 is ringing
[2020-07-31 10:07:27] VERBOSE[7681][C-0000000d] app_dial.c: PJSIP/200-00000019 answered PJSIP/301-00000018
[2020-07-31 10:07:27] VERBOSE[7682][C-0000000d] bridge_channel.c: Channel PJSIP/200-00000019 joined ‘simple_bridge’ basic-bridge <68625808-2b32-46d9-8ea7-529d78104c3b>
[2020-07-31 10:07:27] VERBOSE[7681][C-0000000d] bridge_channel.c: Channel PJSIP/301-00000018 joined ‘simple_bridge’ basic-bridge <68625808-2b32-46d9-8ea7-529d78104c3b>
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] bridge_channel.c: Channel PJSIP/301-00000018 left ‘simple_bridge’ basic-bridge <68625808-2b32-46d9-8ea7-529d78104c3b>
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] app_macro.c: Spawn extension (macro-dial-one, s, 56) exited non-zero on ‘PJSIP/301-00000018’ in macro ‘dial-one’
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] app_macro.c: Spawn extension (macro-exten-vm, s, 26) exited non-zero on ‘PJSIP/301-00000018’ in macro ‘exten-vm’
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] pbx.c: Spawn extension (ext-local, 200, 2) exited non-zero on ‘PJSIP/301-00000018’
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] pbx.c: Executing [h@ext-local:1] Macro(“PJSIP/301-00000018”, “hangupcall,”) in new stack
[2020-07-31 10:07:59] VERBOSE[7682][C-0000000d] bridge_channel.c: Channel PJSIP/200-00000019 left ‘simple_bridge’ basic-bridge <68625808-2b32-46d9-8ea7-529d78104c3b>
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/301-00000018”, “1?theend”) in new stack
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/301-00000018”, “0?Set(CDR(recordingfile)=)”) in new stack
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/301-00000018”, "PJSIP/200-00000019 montior file= ") in new stack
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/301-00000018”, “1?skipagi”) in new stack
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“PJSIP/301-00000018”, “”) in new stack
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/301-00000018’ in macro ‘hangupcall’
[2020-07-31 10:07:59] VERBOSE[7681][C-0000000d] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/301-00000018’

At the Asterisk command prompt, type
pjsip set logger on
make a failing test call, paste the relevant section of the Asterisk log at pastebin.freepbx.org and post the link here.

This is the standard behaviour of the Asterisk when there is no communication on ports 10000 to 20000 (simply said - no voice).
So check if those ports are open between clients and FreePBX box.
I have the very same problem and found that the router in intranet and the firewall do not allow transmition trough those ports. After I oppened them - everything goes smoot.

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