Cannot hear anything when placing calls

OK Well I do not know how to explain this but here we go. My Asterisk 11 server was working great I have not made any changes other than the afterhours recording. But now no matter what I do I place call to any extension or to an external number I can not hear anything at all. I watch the CLI and it shows no errors and is not showing any warnings. below is a sample before I hang up . But Problem is I have can not hear a thing.

Using SIP RTP CoS mark 5 -- Executing [400@default:1] Answer("SIP/417-00000016", "") in new stack > 0x7f1290026820 -- Probation passed - setting RTP source address to 71.200.200.200:4008 -- Executing [400@default:2] Goto("SIP/417-00000016", "default,s,1,3") in new stack -- Goto (default,s,1) -- Executing [s@default:1] NoOp("SIP/417-00000016", "") in new stack -- Executing [s@default:2] Answer("SIP/417-00000016", "") in new stack -- Executing [s@default:3] GotoIfTime("SIP/417-00000016", "08:00-17:00,mon-fri,*,*?open:closed") in new stack -- Goto (default,s,8) -- Executing [s@default:8] BackGround("SIP/417-00000016", "afterhours") in new stack -- <SIP/417-00000016> Playing 'afterhours.gsm' (language 'en') -- Executing [s@default:9] WaitExten("SIP/417-00000016", "3") in new stack -- Timeout on SIP/417-00000016, continuing... -- Executing [s@default:10] Goto("SIP/417-00000016", "closed,s,1,3") in new stack -- Goto (closed,s,1) -- Executing [s@closed:1] Answer("SIP/417-00000016", "") in new stack -- Executing [s@closed:2] WaitExten("SIP/417-00000016", "2") in new stack -- Timeout on SIP/417-00000016, continuing... -- Executing [s@closed:3] Dial("SIP/417-00000016", "SIP/400,20") in new stack [Dec 10 18:09:33] WARNING[2924][C-00000015]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@closed:4] Directory("SIP/417-00000016", "VoiceMail,default") in new stack == Parsing '/etc/asterisk/voicemail.conf': Found == Parsing '/etc/asterisk/users.conf': Found -- <SIP/417-00000016> Playing 'dir-welcome.gsm' (language 'en') -- <SIP/417-00000016> Playing 'dir-pls-enter.gsm' (language 'en') == Spawn extension (closed, s, 4) exited non-zero on 'SIP/417-00000016'

That is a failed call, but you should hear the voice announcements.

No I cannot hear the voice prompts at all. I have restarted my server and also I have done a core reload,sip reload,dialplan reload. Still nothing