[asterisk-users] No audio after transferring to voicemail

Jeremy Phillips JeremyP at cohesivelogic.com
Fri Oct 31 10:30:20 CDT 2008


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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