[Asterisk-Users] Voicemail Problem

Wojciech Tryc wojtek at VoipMan.ORG
Thu Feb 9 18:59:12 MST 2006


You don't have 'vm-goodbye' voice file. Check under /var/lib/asterisk/sounds....
Wojtek
  ----- Original Message ----- 
  From: Sam Lee 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Thursday, February 09, 2006 8:38 PM
  Subject: RE: [Asterisk-Users] Voicemail Problem


  Hey guys,

  Any hint at all ?



------------------------------------------------------------------------------
  From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sam Lee
  Sent: Thursday, February 09, 2006 3:30 PM
  To: asterisk-users at lists.digium.com
  Subject: [Asterisk-Users] Voicemail Problem


  I have just setup my OPENSER to work with the asterisk 1.2.2.
  I've set extension 400 in extension.conf to point to the VoicemailMain() application

  The entire program works fine, but there seems to be some problem whenever the call is hangup, either by pushing # to exit the VoicemailMain() apps or by hanging the phone. If the # button is push, should Asterisk send something back to tell OPENSER to hang up the party ?

  Here's the log of verbose level 3

  Asterisk*CLI>
      -- Playing 'vm-youhave' (language 'en')
      -- Playing 'vm-no' (language 'en')
      -- Playing 'vm-messages' (language 'en')
      -- Playing 'vm-opts' (language 'en')
      -- Playing 'vm-goodbye' (language 'en')
      -- Executing Playback("SIP/210.23.1.139-081ee3d8", "Goodbye") in new stack
  Feb  9 15:05:06 WARNING[23242]: file.c:509 ast_openstream_full: File Goodbye does not exist in any format
  Feb  9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile: Unable to open Goodbye (format alaw): No such file or dire
  ctory
  Feb  9 15:05:06 WARNING[23242]: app_playback.c:132 playback_exec: ast_streamfile failed on SIP/203.125.68.66-081ee3d8
  for Goodbye
      -- Executing Hangup("SIP/203.125.68.66-081ee3d8", "") in new stack
    == Spawn extension (default, 400, 3) exited non-zero on 'SIP/203.125.68.66-081ee3d8'
  Asterisk*CLI>

  Any idea what is this all about ?

  Regards,
  Sam


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