[Asterisk-Users] Voicemail Problem

[Sam Lee]

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060209/4f8b82b7/attachment.htm