[Asterisk-Users] SER + ASTERISK voicemail

[Steve Blair]

You'll want some rules in your sip.conf to handle the connection from 
SER. A
starting point might be:

   [<ser ip addr>:<ser port ?= 5060>]
   type=peer
   context=<my sip context name>
   tos=lowdelay                    ; tos delay
   allow=ulaw                     ; dtmfmode=inband only works with ulaw 
or alaw!
   dtmfmode=inband                ; Choices are inband, rfc2833, or info

You'll then want some rules in extensions.conf to accept the call and 
redirect it
to mailboxes defined in your voicemail.conf or in MySQL. Something like:

   [general]
   context=<my sip context name>
   switch => Realtime/<my sip context name>@extensions
   static=yes

  [<my sip context name>]

  exten => _uXXXXX,1,VoiceMail(${EXTEN}@<my sip context name>)
  exten => _XXXXX,1,VoiceMail(${EXTEN}@<my sip context name>)
  exten => _bXXXXX,1,VoiceMail(${EXTEN}@<my sip context name>))
  exten => #,2,Hangup                     ; Hang them up.

Steve

harry gaillac wrote:

>Hello,
>
>I try set Ua---SER----Asterisk (voicemail/ARA)
>                |
>               Ua
>ser stable
>asterisk cvs head 
>
>I read
>http://mail.iptel.org/pipermail/serusers/2005-February/015997.html
>to forward unavailable or busy sip agents to asterisk
>voicemail in failure route.
>
>How may I configure extensions.conf and ser.cfg ?
>I have been trying without success!
>
>Regards
>Harry
>
>
>	
>
>	
>		
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