----- Original Message -----
Sent: Thursday, February 09, 2006 8:38
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
Hey guys,
Any hint at all ?
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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