[Asterisk-Users] question re voicemail

Glenn Dalgliesh asterisk at techhat.com
Sun Jan 11 18:38:52 MST 2004


I think this is the syntax you are looking for

[sip]
exten => 5104112978,1,Dial(SIP/5104112978,20,tr)
exten => 5104112978,2,Voicemail,u5104112978

exten => 5104112978,102,Voicemail,b5104112978


  ----- Original Message ----- 
  From: Jess Magnaye 
  To: asterisk-users at lists.digium.com 
  Sent: Monday, January 05, 2004 4:28 PM
  Subject: [Asterisk-Users] question re voicemail


  Hi,
  I just setup my * with digium. I started testing voicemail first between atas, and i am not sure why it is not prompting me any when the call is not answered or if busy.  i only get continuous ringback and the following message: 

  asterisk*CLI> 
      -- Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new stack
      -- Called 5104112978
      -- SIP/5104112978-3f88 is ringing
      -- Nobody picked up in 20000 ms

  I wonder if my u<extension> and b<extension> config is correct, mispelled, or something else is missing.  Note that ata to ata via * works, as well as getting to VoicemailMain via extension 1234.    Please help.  My config are found below.  I appreciate your help.


  sip.conf
  -----------

  [6882332]
  type=friend
  username=6882332
  secret=test
  host=dynamic
  defaultip=172.30.200.27
  dtmfmode=rfc2833
  mailbox=6882332
  callerid = "test1" <6882332>
  context=sip

  [5104112978]
  type=friend
  username=5104112978
  secret=test
  host=dynamic
  ;canreinvite=no
  defaultip=172.30.200.26
  dtmfmode=rfc2833
  mailbox=5104112978
  callerid = "test2" <5104112978>
  context=sip

  extensions.conf
  ------------------------
  voicemail.conf
  ---------------------

  [default]
  6882332 => 6882332,test1,jess at arretni.com
  5104112978 => 5104112978,test2, jess at arretni.com
  9011 => 9011,Asterisk,jess at arretni.com
  1111 => 1111,Nada,jess at arretni.com
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