asterisk users - Oct 2008 - No audio after transferring to voicemail

Hello All,

I'm having an issue where asterisk doesn't hear any audio after
transferring to voicemail. Here is the dial plan and console output.

DIAL PLAN
[voicepulse-in]
exten => _14259491337,1,NoOp(Incoming call from VoicePulse)
exten => _14259491337,2,Ringing
exten => _14259491337,3,Wait(1)
exten => _14259491337,4,Dial(SIP/1337,20)
exten => _14259491337,5,VoiceMail(1337)
exten => _14259491337,6,Wait(1)
exten => _14259491337,7,HangUp

CONSOLE OUTPUT
-- Executing [14259491337 at voicepulse-in:4]
Dial("SIP/<vpuser>-081d2800", "SIP/1337|20") in new
stack
    -- Called 1337
    -- SIP/1337-081cfce0 is ringing
    -- Nobody picked up in 20000 ms
    -- Executing [14259491337 at voicepulse-in:5]
VoiceMail("SIP/<vpuser>-081d2800", "1337") in new
stack
    -- <SIP/<vpuser>-081d2800> Playing 'vm-intro' (language
'en')
    -- <SIP/<vpuser>-081d2800> Playing 'beep' (language
'en')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/1337/tmp/KzD4A1
format: wav, 0x8184358
[Oct 31 08:21:02] WARNING[22354]: app.c:602 __ast_play_and_record: No audio
available on SIP/<vpuser>-081d2800??
    -- User hung up
[Oct 31 08:21:02] NOTICE[22354]: pbx.c:1631
pbx_substitute_variables_helper_full: Error in extension logic (missing
'}')
  == Spawn extension (voicepulse-in, 14259491337, 5) exited non-zero on
'SIP/<vpuser>-081d2800'

Any help would be greatly appreciated!

Thanks,

Jeremy Phillips
M: 540.322.7980 | T: 425.949.1337 | B: http://jeremyphillips.org

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