Voicemail not working

Asterisk Asterisk Support
1

Configs as below:

voicemail.conf:

[code]; Voicemail Configuration

[general]
format=wav49|wav
serveremail=asterisk@domain.com
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
emaildateformat=%A, %B %d, %Y at %r
pagerdateformat=%A, %B %d, %Y at %r
sendvoicemail=yes ; Allow the user to compose and send a voicemail while inside
operator=yes

[zonemessages]
eastern=America/New_York|‘vm-received’ Q ‘digits/at’ IMp
central=America/Chicago|‘vm-received’ Q ‘digits/at’ IMp
central24=America/Chicago|‘vm-received’ q ‘digits/at’ H N 'hours’
military=Zulu|‘vm-received’ q ‘digits/at’ H N ‘hours’ 'phonetic/z_p’
european=Europe/Copenhagen|‘vm-received’ a d b ‘digits/at’ HM
london=Europe/London|‘vm-received’ a d b ‘digits/at’ HM

[default]
101 => 1234,Phil,me@domain.com[/code]

extensions.conf:

[code][globals]
OFFICE=SIP/0004F24F5244
SEA=SIP/0004F2ACB23C

[LocalSets]
exten => 101,1,Dial(${OFFICE},10)

exten => 102,1,Dial(${SEA},10)
same => n,PlayBack(hello-world)
same => n,VoiceMail(101@default,s)
same => n,PlayBack(vm-goodbye)

include => services

[inbound]
exten => _.,1,Dial(${OFFICE},5)
same => n,VoiceMail(101@default,u)

[services]
exten => *98,1,VoiceMailMain()[/code]

When I dial either the inbound extension from outside or the 102 extension from inside, Asterisk hangs up when it reaches the VoiceMail() call. In the case of 102, it plays hello-world and then hangs up.

Looking at a pcap, Asterisk sends no RTP before hanging up but it does receive a short RTP stream.

Any clues as to why this is not working?

2

What is the actual console output?

3

Sorry, forgot to post that and then my post got stuck in moderation all morning. Below!

  == Using SIP RTP CoS mark 5
    -- Executing [s@inbound:1] Dial("SIP/synety-00000000", "SIP/0004F24F5244,5") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/0004F24F5244
    -- SIP/0004F24F5244-00000001 is ringing
    -- Nobody picked up in 5000 ms
    -- Executing [s@inbound:2] VoiceMail("SIP/synety-00000000", "101@default,u") in new stack
  == Spawn extension (inbound, s, 2) exited non-zero on 'SIP/synety-00000000'
    -- Executing [h@inbound:1] Dial("SIP/synety-00000000", "SIP/0004F24F5244,5") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/0004F24F5244
  == Spawn extension (inbound, h, 1) exited non-zero on 'SIP/synety-00000000'
4

Resolved by the folks in the IRC channel. I enabled debug log:

And saw this on console:

[Sep 29 17:27:52] WARNING[28728][C-00000001]: app_voicemail.c:1888 create_dirpath: ast_mkdir '/var/spool/asterisk/voicemail/default/101/tmp' failed: Permission denied [Sep 29 17:27:52] WARNING[28728][C-00000001]: app_voicemail.c:6406 leave_voicemail: Failed to make directory (/var/spool/asterisk/voicemail/default/101/temp)

Turned out to be a permissions error on /var/spool/asterisk/voicemail/default/