----- 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
_______________________________________________
--Bandwidth and
Colocation provided by Easynews.com --
Asterisk-Users mailing
list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users