Voicemail works only for internal calls

I am having difficulty getting voicemail to work for inbound calls, it will simply hangup and beep twice (think that is AT&T on my cell phone, but not sure). However, if I call an extension using another extension, voicemail works perfect.

I have an inbound route set to forward all DID to a single extension (for troubleshooting). I can’t recall the commands to make the asterisk command line output the actions it is taking, so if you think that would help (as I do), can you also post that. (I have tried core set debug 10, sip set debug peer [extension])

Thanks.

Whoops. I did use the AsteriskNOW-1.7.1-i386.iso image, with FreePBX. Sorry for the confusion.

Calling in from my cell phone and answering the extension has two-way audio with essentially no lag. Calling from another internal extension also has two-way audio.

I updated everything to 2.8 and the problem is still the same.

Thanks again for the help

You need to tell us what kind of system this is, Asterisk and FreePBX version and how it was installed.

The command is ‘core set verbose 3’

It is installed on an old laptop (sorry, I don’t have it with me, VPN at the moment) using the Free PBX image. It is FreePBX 2.7.0.10 (in the bar on the page), but says 2.7.0.12 under module admin for Core. (It was installed as 2.7, just the defaults from the image). Voicemail is 2.7.0.1 and enabled. The inbound route is set to ring only the one extension, which has a ring time of 5 to speed up troubleshooting.

Thanks

The first long section is what happens when I dial in using my cell phone. I replaced my cell number with MY_CELL_PHONE and my trunk id with MY_TRUNK_NUMBER.

The second long section is what happens when I dial the same extension using another extension.

Calling from an external number
– Executing [[email protected]:1] Set(“SIP/VoipVoip-00000030”, “GROUP()=OUT_2”) in new stack
– Executing [[email protected]:2] Goto(“SIP/VoipVoip-00000030”, “from-trunk,MY_TRUNK_NUMBER,1”) in new stack
– Goto (from-trunk,MY_TRUNK_NUMBER,1)
– Executing [[email protected]:1] NoOp(“SIP/VoipVoip-00000030”, “Catch-All DID Match - Found MY_TRUNK_NUMBER - You probably want a DID for this.”) in new stack
– Executing [[email protected]:2] Goto(“SIP/VoipVoip-00000030”, “ext-did,s,1”) in new stack
– Goto (ext-did,s,1)
– Executing [[email protected]:1] Set(“SIP/VoipVoip-00000030”, “__FROM_DID=s”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/VoipVoip-00000030”, “1 ?Set(CALLERID(name)=+MY_CELL_PHONE)”) in new stack
– Executing [[email protected]:3] Set(“SIP/VoipVoip-00000030”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [[email protected]:4] Set(“SIP/VoipVoip-00000030”, “CALLERPRES()=allowed_not_screened”) in new stack
– Executing [[email protected]:5] Goto(“SIP/VoipVoip-00000030”, “from-did-direct,12,1”) in new stack
– Goto (from-did-direct,12,1)
– Executing [[email protected]:1] Set(“SIP/VoipVoip-00000030”, “__RINGTIMER=5”) in new stack
– Executing [[email protected]:2] Macro(“SIP/VoipVoip-00000030”, “exten-vm,12,12”) in new stack
– Executing [[email protected]:1] Macro(“SIP/VoipVoip-00000030”, “user-callerid,”) in new stack
– Executing [[email protected]:1] Set(“SIP/VoipVoip-00000030”, “AMPUSER=+MY_CELL_PHONE”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/VoipVoip-00000030”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/VoipVoip-00000030”, “1?Set(REALCALLERIDNUM=+MY_CELL_PHONE)”) in new stack
– Executing [[email protected]:4] Set(“SIP/VoipVoip-00000030”, “AMPUSER=”) in new stack
– Executing [[email protected]:5] Set(“SIP/VoipVoip-00000030”, “AMPUSERCIDNAME=”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/VoipVoip-00000030”, “1?report”) in new stack
– Goto (macro-user-callerid,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/VoipVoip-00000030”, “0?continue”) in new stack
– Executing [[email protected]:10] Set(“SIP/VoipVoip-00000030”, “__TTL=64”) in new stack
– Executing [[email protected]:11] GotoIf(“SIP/VoipVoip-00000030”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [[email protected]:18] Set(“SIP/VoipVoip-00000030”, “CALLERID(number)=+MY_CELL_PHONE”) in new stack
– Executing [[email protected]:19] Set(“SIP/VoipVoip-00000030”, “CALLERID(name)=+MY_CELL_PHONE”) in new stack
– Executing [[email protected]:20] NoOp(“SIP/VoipVoip-00000030”, “Using CallerID “+MY_CELL_PHONE” <+MY_CELL_PHONE>”) in new stack
– Executing [[email protected]:2] Set(“SIP/VoipVoip-00000030”, “RingGroupMethod=none”) in new stack
– Executing [[email protected]:3] Set(“SIP/VoipVoip-00000030”, “VMBOX=12”) in new stack
– Executing [[email protected]:4] Set(“SIP/VoipVoip-00000030”, “EXTTOCALL=12”) in new stack
– Executing [[email protected]:5] Set(“SIP/VoipVoip-00000030”, “CFUEXT=”) in new stack
– Executing [[email protected]:6] Set(“SIP/VoipVoip-00000030”, “CFBEXT=”) in new stack
– Executing [[email protected]:7] Set(“SIP/VoipVoip-00000030”, “RT=5”) in new stack
– Executing [[email protected]:8] Macro(“SIP/VoipVoip-00000030”, “record-enable,12,IN”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/VoipVoip-00000030”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] ExecIf(“SIP/VoipVoip-00000030”, “0?MacroExit()”) in new stack
– Executing [[email protected]:5] GotoIf(“SIP/VoipVoip-00000030”, “0?Group:OUT”) in new stack
– Goto (macro-record-enable,s,15)
– Executing [[email protected]:15] GotoIf(“SIP/VoipVoip-00000030”, “1?IN”) in new stack
– Goto (macro-record-enable,s,20)
– Executing [[email protected]:20] ExecIf(“SIP/VoipVoip-00000030”, “1?MacroExit()”) in new stack
– Executing [[email protected]:9] Macro(“SIP/VoipVoip-00000030”, “dial,5,tr,12”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/VoipVoip-00000030”, “1?dial”) in new stack
– Goto (macro-dial,s,3)
– Executing [[email protected]:3] AGI(“SIP/VoipVoip-00000030”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘+MY_CELL_PHONE’ number is '+MY_CELL_PHONE’
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 12 to extension map
– dialparties.agi: Extension 12 cf is disabled
– dialparties.agi: Extension 12 do not disturb is disabled
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
– dialparties.agi: dbset CALLTRACE/12 to +MY_CELL_PHONE
– dialparties.agi: Filtered ARG3: 12
– <SIP/VoipVoip-00000030>AGI Script dialparties.agi completed, returning 0
– Executing [[email protected]:7] Dial(“SIP/VoipVoip-00000030”, “SIP/12,5,tr”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called 12
– SIP/12-00000031 is ringing
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Nobody picked up in 5000 ms
– Executing [[email protected]:8] Set(“SIP/VoipVoip-00000030”, “DIALSTATUS=NOANSWER”) in new stack
– Executing [[email protected]:9] GosubIf(“SIP/VoipVoip-00000030”, “0?NOANSWER,1”) in new stack
– Executing [[email protected]:10] GotoIf(“SIP/VoipVoip-00000030”, “0?exit,return”) in new stack
– Executing [[email protected]:11] Set(“SIP/VoipVoip-00000030”, “SV_DIALSTATUS=NOANSWER”) in new stack
– Executing [[email protected]:12] GosubIf(“SIP/VoipVoip-00000030”, “0?docfu,1”) in new stack
– Executing [[email protected]:13] GosubIf(“SIP/VoipVoip-00000030”, “0?docfb,1”) in new stack
– Executing [[email protected]:14] Set(“SIP/VoipVoip-00000030”, “DIALSTATUS=NOANSWER”) in new stack
– Executing [[email protected]:15] NoOp(“SIP/VoipVoip-00000030”, “Voicemail is ‘12’”) in new stack
– Executing [[email protected]:16] GotoIf(“SIP/VoipVoip-00000030”, “0?s-NOANSWER,1”) in new stack
– Executing [[email protected]:17] NoOp(“SIP/VoipVoip-00000030”, “Sending to Voicemail box 12”) in new stack
– Executing [[email protected]:18] Macro(“SIP/VoipVoip-00000030”, “vm,12,NOANSWER,”) in new stack
– Executing [[email protected]:1] Macro(“SIP/VoipVoip-00000030”, “user-callerid,SKIPTTL”) in new stack
– Executing [[email protected]:1] Set(“SIP/VoipVoip-00000030”, “AMPUSER=+MY_CELL_PHONE”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/VoipVoip-00000030”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/VoipVoip-00000030”, “0?Set(REALCALLERIDNUM=+MY_CELL_PHONE)”) in new stack
– Executing [[email protected]:4] Set(“SIP/VoipVoip-00000030”, “AMPUSER=”) in new stack
– Executing [[email protected]:5] Set(“SIP/VoipVoip-00000030”, “AMPUSERCIDNAME=”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/VoipVoip-00000030”, “1?report”) in new stack
– Goto (macro-user-callerid,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/VoipVoip-00000030”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [[email protected]:18] Set(“SIP/VoipVoip-00000030”, “CALLERID(number)=+MY_CELL_PHONE”) in new stack
– Executing [[email protected]:19] Set(“SIP/VoipVoip-00000030”, “CALLERID(name)=+MY_CELL_PHONE”) in new stack
– Executing [[email protected]:20] NoOp(“SIP/VoipVoip-00000030”, “Using CallerID “+MY_CELL_PHONE” <+MY_CELL_PHONE>”) in new stack
– Executing [[email protected]:2] Set(“SIP/VoipVoip-00000030”, “VMGAIN=”"") in new stack
– Executing [[email protected]:3] GotoIf(“SIP/VoipVoip-00000030”, “1?vmx,1”) in new stack
– Goto (macro-vm,vmx,1)
– Executing [[email protected]:1] Set(“SIP/VoipVoip-00000030”, “MEXTEN=12”) in new stack
– Executing [[email protected]:2] Set(“SIP/VoipVoip-00000030”, “MMODE=NOANSWER”) in new stack
– Executing [[email protected]:3] Set(“SIP/VoipVoip-00000030”, “RETVM=”) in new stack
– Executing [[email protected]:4] Set(“SIP/VoipVoip-00000030”, “MODE=unavail”) in new stack
– Executing [[email protected]:5] GotoIf(“SIP/VoipVoip-00000030”, “1?chknomsg”) in new stack
– Goto (macro-vm,vmx,7)
– Executing [[email protected]:7] GotoIf(“SIP/VoipVoip-00000030”, “0?s-NOANSWER,1”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/VoipVoip-00000030”, “1?notdirect”) in new stack
– Goto (macro-vm,vmx,10)
– Executing [[email protected]:10] NoOp(“SIP/VoipVoip-00000030”, "Checking if ext 12 is enabled: ") in new stack
– Executing [[email protected]:11] GotoIf(“SIP/VoipVoip-00000030”, “1?s-NOANSWER,1”) in new stack
– Goto (macro-vm,s-NOANSWER,1)
– Executing [[email protected]:1] Macro(“SIP/VoipVoip-00000030”, “get-vmcontext,12”) in new stack
– Executing [[email protected]:1] Set(“SIP/VoipVoip-00000030”, “VMCONTEXT=default”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/VoipVoip-00000030”, “0?200:300”) in new stack
– Goto (macro-get-vmcontext,s,300)
– Executing [[email protected]:300] NoOp(“SIP/VoipVoip-00000030”, “”) in new stack
– Executing [[email protected]:2] VoiceMail(“SIP/VoipVoip-00000030”, “[email protected],u”) in new stack
== Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on ‘SIP/VoipVoip-00000030’ in macro ‘vm’
== Spawn extension (macro-exten-vm, s, 18) exited non-zero on ‘SIP/VoipVoip-00000030’ in macro ‘exten-vm’
== Spawn extension (from-did-direct, 12, 2) exited non-zero on ‘SIP/VoipVoip-00000030’

And below is when I dial the same extension from an internal extension.

-- Executing [[email protected]:1] Set("SIP/14-00000032", "__RINGTIMER=5") in new stack
-- Executing [[email protected]:2] Macro("SIP/14-00000032", "exten-vm,12,12") in new stack
-- Executing [[email protected]:1] Macro("SIP/14-00000032", "user-callerid,") in new stack
-- Executing [[email protected]:1] Set("SIP/14-00000032", "AMPUSER=14") in new stack
-- Executing [[email protected]:2] GotoIf("SIP/14-00000032", "0?report") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/14-00000032", "1?Set(REALCALLERIDNUM=14)") in new stack
-- Executing [[email protected]:4] Set("SIP/14-00000032", "AMPUSER=14") in new stack
-- Executing [[email protected]:5] Set("SIP/14-00000032", "AMPUSERCIDNAME=Xlite") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/14-00000032", "0?report") in new stack
-- Executing [[email protected]:7] Set("SIP/14-00000032", "AMPUSERCID=14") in new stack
-- Executing [[email protected]:8] Set("SIP/14-00000032", "CALLERID(all)="Xlite" <14>") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/14-00000032", "0?continue") in new stack
-- Executing [[email protected]:10] Set("SIP/14-00000032", "__TTL=64") in new stack
-- Executing [[email protected]:11] GotoIf("SIP/14-00000032", "1?continue") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [[email protected]:18] Set("SIP/14-00000032", "CALLERID(number)=14") in new stack
-- Executing [[email protected]:19] Set("SIP/14-00000032", "CALLERID(name)=Xlite") in new stack
-- Executing [[email protected]:20] NoOp("SIP/14-00000032", "Using CallerID "Xlite" <14>") in new stack
-- Executing [[email protected]:2] Set("SIP/14-00000032", "RingGroupMethod=none") in new stack
-- Executing [[email protected]:3] Set("SIP/14-00000032", "VMBOX=12") in new stack
-- Executing [[email protected]:4] Set("SIP/14-00000032", "EXTTOCALL=12") in new stack
-- Executing [[email protected]:5] Set("SIP/14-00000032", "CFUEXT=") in new stack
-- Executing [[email protected]:6] Set("SIP/14-00000032", "CFBEXT=") in new stack
-- Executing [[email protected]:7] Set("SIP/14-00000032", "RT=5") in new stack
-- Executing [[email protected]:8] Macro("SIP/14-00000032", "record-enable,12,IN") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/14-00000032", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [[email protected]:4] ExecIf("SIP/14-00000032", "0?MacroExit()") in new stack
-- Executing [[email protected]:5] GotoIf("SIP/14-00000032", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [[email protected]:15] GotoIf("SIP/14-00000032", "1?IN") in new stack
-- Goto (macro-record-enable,s,20)
-- Executing [[email protected]:20] ExecIf("SIP/14-00000032", "1?MacroExit()") in new stack
-- Executing [[email protected]:9] Macro("SIP/14-00000032", "dial,5,tr,12") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/14-00000032", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [[email protected]:3] AGI("SIP/14-00000032", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi

dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘Xlite’ number is '14’
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 12 to extension map
– dialparties.agi: Extension 12 cf is disabled
– dialparties.agi: Extension 12 do not disturb is disabled
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
– dialparties.agi: dbset CALLTRACE/12 to 14
– dialparties.agi: Filtered ARG3: 12
– <SIP/14-00000032>AGI Script dialparties.agi completed, returning 0
– Executing [[email protected]:7] Dial(“SIP/14-00000032”, “SIP/12,5,tr”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called 12
– SIP/12-00000033 is ringing
– Remote UNIX connection
– Remote UNIX connection disconnected
– Nobody picked up in 5000 ms
– Executing [[email protected]:8] Set(“SIP/14-00000032”, “DIALSTATUS=NOANSWER”) in new stack
– Executing [[email protected]:9] GosubIf(“SIP/14-00000032”, “0?NOANSWER,1”) in new stack
– Executing [[email protected]:10] GotoIf(“SIP/14-00000032”, “0?exit,return”) in new stack
– Executing [[email protected]:11] Set(“SIP/14-00000032”, “SV_DIALSTATUS=NOANSWER”) in new stack
– Executing [[email protected]:12] GosubIf(“SIP/14-00000032”, “0?docfu,1”) in new stack
– Executing [[email protected]:13] GosubIf(“SIP/14-00000032”, “0?docfb,1”) in new stack
– Executing [[email protected]:14] Set(“SIP/14-00000032”, “DIALSTATUS=NOANSWER”) in new stack
– Executing [[email protected]:15] NoOp(“SIP/14-00000032”, “Voicemail is ‘12’”) in new stack
– Executing [[email protected]:16] GotoIf(“SIP/14-00000032”, “0?s-NOANSWER,1”) in new stack
– Executing [[email protected]:17] NoOp(“SIP/14-00000032”, “Sending to Voicemail box 12”) in new stack
– Executing [[email protected]:18] Macro(“SIP/14-00000032”, “vm,12,NOANSWER,”) in new stack
– Executing [[email protected]:1] Macro(“SIP/14-00000032”, “user-callerid,SKIPTTL”) in new stack
– Executing [[email protected]:1] Set(“SIP/14-00000032”, “AMPUSER=14”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/14-00000032”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/14-00000032”, “0?Set(REALCALLERIDNUM=14)”) in new stack
– Executing [[email protected]:4] Set(“SIP/14-00000032”, “AMPUSER=14”) in new stack
– Executing [[email protected]:5] Set(“SIP/14-00000032”, “AMPUSERCIDNAME=Xlite”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/14-00000032”, “0?report”) in new stack
– Executing [[email protected]:7] Set(“SIP/14-00000032”, “AMPUSERCID=14”) in new stack
– Executing [[email protected]:8] Set(“SIP/14-00000032”, “CALLERID(all)=“Xlite” <14>”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/14-00000032”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [[email protected]:18] Set(“SIP/14-00000032”, “CALLERID(number)=14”) in new stack
– Executing [[email protected]:19] Set(“SIP/14-00000032”, “CALLERID(name)=Xlite”) in new stack
– Executing [[email protected]:20] NoOp(“SIP/14-00000032”, “Using CallerID “Xlite” <14>”) in new stack
– Executing [[email protected]:2] Set(“SIP/14-00000032”, “VMGAIN=”"") in new stack
– Executing [[email protected]:3] GotoIf(“SIP/14-00000032”, “1?vmx,1”) in new stack
– Goto (macro-vm,vmx,1)
– Executing [[email protected]:1] Set(“SIP/14-00000032”, “MEXTEN=12”) in new stack
– Executing [[email protected]:2] Set(“SIP/14-00000032”, “MMODE=NOANSWER”) in new stack
– Executing [[email protected]:3] Set(“SIP/14-00000032”, “RETVM=”) in new stack
– Executing [[email protected]:4] Set(“SIP/14-00000032”, “MODE=unavail”) in new stack
– Executing [[email protected]:5] GotoIf(“SIP/14-00000032”, “1?chknomsg”) in new stack
– Goto (macro-vm,vmx,7)
– Executing [[email protected]:7] GotoIf(“SIP/14-00000032”, “0?s-NOANSWER,1”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/14-00000032”, “1?notdirect”) in new stack
– Goto (macro-vm,vmx,10)
– Executing [[email protected]:10] NoOp(“SIP/14-00000032”, "Checking if ext 12 is enabled: ") in new stack
– Executing [[email protected]:11] GotoIf(“SIP/14-00000032”, “1?s-NOANSWER,1”) in new stack
– Goto (macro-vm,s-NOANSWER,1)
– Executing [[email protected]:1] Macro(“SIP/14-00000032”, “get-vmcontext,12”) in new stack
– Executing [[email protected]:1] Set(“SIP/14-00000032”, “VMCONTEXT=default”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/14-00000032”, “0?200:300”) in new stack
– Goto (macro-get-vmcontext,s,300)
– Executing [[email protected]:300] NoOp(“SIP/14-00000032”, “”) in new stack
– Executing [[email protected]:2] VoiceMail(“SIP/14-00000032”, “[email protected],u”) in new stack
– <SIP/14-00000032> Playing ‘vm-theperson.ulaw’ (language ‘en’)
– Remote UNIX connection
– Remote UNIX connection disconnected
– <SIP/14-00000032> Playing ‘digits/1.ulaw’ (language ‘en’)
– <SIP/14-00000032> Playing ‘digits/2.ulaw’ (language ‘en’)
– <SIP/14-00000032> Playing ‘vm-isunavail.ulaw’ (language ‘en’)
– <SIP/14-00000032> Playing ‘vm-intro.ulaw’ (language ‘en’)
– Remote UNIX connection
– Remote UNIX connection disconnected
– <SIP/14-00000032> Playing ‘beep.ulaw’ (language ‘en’)
– Remote UNIX connection
– Remote UNIX connection disconnected
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/default/12/tmp/kTKAkT format: wav49, 0x9e3ff50
– x=1, open writing: /var/spool/asterisk/voicemail/default/12/tmp/kTKAkT format: wav, 0xa1702d0
– User ended message by pressing #
– <SIP/14-00000032> Playing ‘auth-thankyou.ulaw’ (language ‘en’)
– Remote UNIX connection
– Remote UNIX connection disconnected
– <SIP/14-00000032> Playing ‘vm-review.ulaw’ (language ‘en’)
– <SIP/14-00000032> Playing ‘vm-goodbye.ulaw’ (language ‘en’)
== Parsing ‘/var/spool/asterisk/voicemail/default/12/INBOX/msg0000.txt’: == Found
== Parsing ‘/var/spool/asterisk/voicemail/default/12/INBOX/msg0000.txt’: == Found
== Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on ‘SIP/14-00000032’ in macro ‘vm’
== Spawn extension (macro-exten-vm, s, 18) exited non-zero on ‘SIP/14-00000032’ in macro ‘exten-vm’
== Spawn extension (from-internal, 12, 2) exited non-zero on ‘SIP/14-00000032’
– Executing [[email protected]:1] Macro(“SIP/14-00000032”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/14-00000032”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/14-00000032”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [[email protected]:7] GotoIf(“SIP/14-00000032”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] Hangup(“SIP/14-00000032”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/14-00000032’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/14-00000032’

That is very off. If you answer the extension do you have two way audio on the call?

Also what distribution did you install from? FreePBX distro installs 2.9 so you did not use that.

If you are using Asterisk Now I strongly suggest you update to at least FreePBX 2.8 using module admin and whatever Asterisk you can via yum within the same train you are on (example stay within 1.6.x).

Best guess, you have the trunk defined with a codec, and there is no voice files for Voicemail application in Asterisk in that format, that is why Voicemail exits as soon as it would play the voice file…

Thanks for the suggestions. Assuming this is how one checks the codes; in both the trunk outgoing and incoming settings I have the following line:
allow=g729&ulaw&alaw&ilbc

Under the add-on “Asterisk SIP Settings”, those 4 codecs are checked, the rest are unchecked. Non-Standard g726 and T38 Pass-Through are both “No”

Here are my settings to help

Outgoing
username=USERNAME
type=peer
qualify=yes
secret=PASSWORD
nat=auto
insecure=very
host=PROVIDER.com
fromuser=USERNAME
fromdomain=PROVIDER.com
dtmfmode=rfc2833
disallow=all
allow=g729&ilbc&ulaw&alaw

Incoming
username=USERNAME
type=user
secret=PASSWORD
nat=auto
insecure=very
host=PROVIDER.com
fromdomain=PROVIDER.com
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=g729&ulaw&alaw&ilbc

Do you have any g.729 licenses installed? They have to be bought from Digium.

Look in /var/lib/asterisk/sounds/en that you have voice files that ends in .g729

Awesome, the problem is now solved. I deleted g729 from the trunk incoming and outgoing allow settings and I unchecked g729 under Asterisk SIP Settings.

In other news, this also fixed a problem with the IVR. (I don’t have a license for that codec, and there are no files in the …/sounds/en directory ending in .g729.)

Thanks for the help.