Calls hang up after 10 min

Hi i have a running PBX with Asterisk 11.5.1 and FreePBX 2.11.0.11,

it works just perfect but i have one problem when I make or receive a external call the call hang up after 10:35 minutes

I have been reading forums about this issue and I find the could be the NAT or the session-timers, soo because I have my PBX directly attached to internet i set up NAT=never and session-timers=refuse.

but… I’m steel having the problem after 10:35 minutes the call hang up, please I will appreciate any help thanks for your time.

here is the last part of the log, if need more info please let me know.

thanks again…

2013-10-07 17:44:07] VERBOSE[18912][C-000003ab] pbx.c: – Executing [[email protected]:1] Macro(“SIP/204-000007ef”, “hangupcall,”) in new stack
[2013-10-07 17:44:07] VERBOSE[18912][C-000003ab] pbx.c: – Executing [s[email protected]:1] GotoIf(“SIP/204-000007ef”, “1?theend”) in new stack
[2013-10-07 17:44:07] VERBOSE[18912][C-000003ab] pbx.c: – Goto (macro-hangupcall,s,3)
[2013-10-07 17:44:07] VERBOSE[18912][C-000003ab] pbx.c: – Executing [[email protected]:3] ExecIf(“SIP/204-000007ef”, “0?Set(CDR(recordingfile)=)”) in new stack
[2013-10-07 17:44:07] VERBOSE[18912][C-000003ab] pbx.c: – Executing [[email protected]:4] Hangup(“SIP/204-000007ef”, “”) in new stack
[2013-10-07 17:44:07] VERBOSE[18912][C-000003ab] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/204-000007ef’ in macro ‘hangupcall’
[2013-10-07 17:44:07] VERBOSE[18912][C-000003ab] pbx.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/204-000007ef’
[2013-10-07 17:44:07] VERBOSE[18912][C-000003ab] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on ‘SIP/204-000007ef’ in macro ‘dialout-trunk’
[2013-10-07 17:44:07] VERBOSE[18912][C-000003ab] pbx.c: == Spawn extension (from-internal, 17134108415, 6) exited non-zero on ‘SIP/204-000007ef’

sorry it happens just when I’m making a external call when I’m receiving an externall call its ok

well just in case that some one wants to know how I fix it…

I just set NAT=yes on SIP configuration module and I did the same as well in the SIP trunk configuration “nat=yes”

now I’m able to talk more than 10 min in a call