Voicemail Audio Issue

Recently installed Asterisk 1.8 and FreePBX 2.10 on CentOS 6.2. Incoming and outgoing calls work flawlessly as does voicemail access from internal extensions. However, I am running into an issue with voicemail access when inbound.

If I dial my did from the outside and let call go to VM I hear complete silence after the last ring (no message, tone, etc.). This lasts for several seconds before the line is disconnected. However, if instead of letting it go to VM I pick up the extension, the audio is fine. I can also dial the extension from itself - if I do this I hear the busy vm message.

Looking at the Asterisk logs I see that it says it is playing unavail.slin, vm-intro.ulaw, and beep.ulaw (and it does when I call it locally, so the files exist). There is also a warning after that regarding audio being unavailable. I included relevant parts of the log below. Any help would be greatly appreciated.

– Executing [[email protected]-one:43] ExecIf(“SIP/Flowroute-0000005b”, “0?Set(DIALSTATUS=)”) in new stack
– Executing [[email protected]:44] GosubIf(“SIP/Flowroute-0000005b”, “0?s-NOANSWER,1()”) in new stack
– Executing [[email protected]:45] MacroExit(“SIP/Flowroute-0000005b”, “”) in new stack
– Executing [[email protected]:8] Set(“SIP/Flowroute-0000005b”, “SV_DIALSTATUS=NOANSWER”) in new stack
– Executing [[email protected]:9] GosubIf(“SIP/Flowroute-0000005b”, “0?docfu,1()”) in new stack
– Executing [[email protected]:10] GosubIf(“SIP/Flowroute-0000005b”, “0?docfb,1()”) in new stack
– Executing [[email protected]:11] Set(“SIP/Flowroute-0000005b”, “DIALSTATUS=NOANSWER”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/Flowroute-0000005b”, “1?MacroExit()”) in new stack
– Executing [[email protected]:3] Set(“SIP/Flowroute-0000005b”, “__PICKUPMARK=”) in new stack
– Executing [[email protected]:4] GotoIf(“SIP/Flowroute-0000005b”, “1?ext-local,vmu1000,1”) in new stack
– Goto (ext-local,vmu1000,1)
– Executing [[email protected]:1] Macro(“SIP/Flowroute-0000005b”, “vm,1000,NOANSWER,”) in new stack
– Executing [[email protected]:1] Macro(“SIP/Flowroute-0000005b”, “user-callerid,SKIPTTL”) in new stack
– Executing [[email protected]:1] Set(“SIP/Flowroute-0000005b”, “AMPUSER=+NUMBER”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/Flowroute-0000005b”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/Flowroute-0000005b”, “0?Set(REALCALLERIDNUM=+NUMBER)”) in new stack
– Executing [[email protected]:4] Set(“SIP/Flowroute-0000005b”, “AMPUSER=”) in new stack
– Executing [[email protected]:5] Set(“SIP/Flowroute-0000005b”, “AMPUSERCIDNAME=”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/Flowroute-0000005b”, “1?report”) in new stack
– Goto (macro-user-callerid,s,11)
– Executing [[email protected]:11] GotoIf(“SIP/Flowroute-0000005b”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,24)
– Executing [[email protected]:24] Set(“SIP/Flowroute-0000005b”, “CALLERID(number)=+NUMBER”) in new stack
– Executing [[email protected]:25] Set(“SIP/Flowroute-0000005b”, “CALLERID(name)=+NUMBER”) in new stack
– Executing [[email protected]:26] Set(“SIP/Flowroute-0000005b”, “CHANNEL(language)=en”) in new stack
– Executing [[email protected]:2] Set(“SIP/Flowroute-0000005b”, “VMGAIN=”"") in new stack
– Executing [[email protected]:3] Macro(“SIP/Flowroute-0000005b”, “blkvm-check,”) in new stack
– Executing [[email protected]:1] Set(“SIP/Flowroute-0000005b”, “GOSUB_RETVAL=”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/Flowroute-0000005b”, “0?Set(GOSUB_RETVAL=TRUE)”) in new stack
– Executing [[email protected]:3] MacroExit(“SIP/Flowroute-0000005b”, “”) in new stack
– Executing [[email protected]:4] GotoIf(“SIP/Flowroute-0000005b”, “1?vmx,1”) in new stack
– Goto (macro-vm,vmx,1)
– Executing [[email protected]:1] Set(“SIP/Flowroute-0000005b”, “MEXTEN=1000”) in new stack
– Executing [[email protected]:2] Set(“SIP/Flowroute-0000005b”, “MMODE=NOANSWER”) in new stack
– Executing [[email protected]:3] Set(“SIP/Flowroute-0000005b”, “RETVM=”) in new stack
– Executing [[email protected]:4] Set(“SIP/Flowroute-0000005b”, “MODE=unavail”) in new stack
– Executing [[email protected]:5] GotoIf(“SIP/Flowroute-0000005b”, “1?chknomsg”) in new stack
– Goto (macro-vm,vmx,8)
– Executing [[email protected]:8] GotoIf(“SIP/Flowroute-0000005b”, “0?s-NOANSWER,1”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/Flowroute-0000005b”, “1?notdirect”) in new stack
– Goto (macro-vm,vmx,11)
– Executing [[email protected]:11] NoOp(“SIP/Flowroute-0000005b”, “Checking if ext 1000 is enabled: “) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/Flowroute-0000005b”, “1?s-NOANSWER,1”) in new stack
– Goto (macro-vm,s-NOANSWER,1)
– Executing [[email protected]:1] Macro(“SIP/Flowroute-0000005b”, “get-vmcontext,1000”) in new stack
– Executing [[email protected]:1] Set(“SIP/Flowroute-0000005b”, “VMCONTEXT=default”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/Flowroute-0000005b”, “0?200:300”) in new stack
– Goto (macro-get-vmcontext,s,300)
– Executing [[email protected]:300] NoOp(“SIP/Flowroute-0000005b”, “”) in new stack
– Executing [[email protected]:2] VoiceMail(“SIP/Flowroute-0000005b”, “[email protected],u”””) in new stack
Audio is at 14158
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

-- <SIP/Flowroute-0000005b> Playing '/var/spool/asterisk/voicemail/default/1000/unavail.slin' (language 'en')
-- <SIP/Flowroute-0000005b> Playing 'vm-intro.ulaw' (language 'en')
-- <SIP/Flowroute-0000005b> Playing 'beep.ulaw' (language 'en')
-- Recording the message
-- x=0, open writing:  /var/spool/asterisk/voicemail/default/1000/tmp/F4BNOk format: wav, 0x7f39a8013b88

[2012-04-07 14:59:29] WARNING[9971]: app.c:855 __ast_play_and_record: No audio available on SIP/Flowroute-0000005b??
– User hung up
– Recording was 0 seconds long but needs to be at least 10 - abandoning
== Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on ‘SIP/Flowroute-0000005b’ in macro ‘vm’
== Spawn extension (ext-local, vmu1000, 1) exited non-zero on ‘SIP/Flowroute-0000005b’
– Executing [[email protected]:1] Macro(“SIP/Flowroute-0000005b”, “hangupcall,”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/Flowroute-0000005b”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [[email protected]:3] Hangup(“SIP/Flowroute-0000005b”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘SIP/Flowroute-0000005b’ in macro ‘hangupcall’
== Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/Flowroute-0000005b’
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: ACK)
set_destination: Parsing sip:216.115.69.144;lr for address/port to send to
set_destination: set destination to 216.115.69.144:5060
Reliably Transmitting (NAT) to 216.115.69.144:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK6511dd2a;rport
Route: sip:216.115.69.144;lr,sip:216.115.69.132;lr
Max-Forwards: 70
From: sip:[email protected];tag=as23bb4c3b
To: sip:[email protected];tag=gK02387440
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: FPBX-2.10.0(1.8.11.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0