Here is the output of the CLI with verbose and debug set to 3 :
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Executing [...@local:1] Dial("SIP/*15-0849a370", "SIP/*11,60")
in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
[Nov 29 03:38:13] WARNING[24635]: app_dial.c:1528 dial_exec_full:
Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [...@local:2] VoiceMail("SIP/*15-0849a370", "*11")
in new stack
-- <SIP/*15-0849a370> Playing 'vm-intro.alaw' (language 'fr')
-- <SIP/*15-0849a370> Playing 'beep.alaw' (language 'fr')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/*11/
tmp/40taTt format: wav49, 0x849b338
-- x=1, open writing: /var/spool/asterisk/voicemail/default/*11/
tmp/40taTt format: gsm, 0x849c7c0
-- x=2, open writing: /var/spool/asterisk/voicemail/default/*11/
tmp/40taTt format: wav, 0x849cb08
-- User hung up
== Parsing '/var/spool/asterisk/voicemail/default/*11/INBOX/
msg0000.txt': == Found
== Spawn extension (local, *11, 2) exited non-zero on 'SIP/
*15-0849a370'
-- Executing [...@local:1] Hangup("SIP/*15-0849a370", "") in new
stack
== Spawn extension (local, h, 1) exited non-zero on 'SIP/
*15-0849a370'
Th Warren
Matthieu NICAISE
Responsable technique
GSM : 06 72 19 09 55
[email protected]
------------------------------------------------------------------------
Thinkro System
http://www.thinkrosystem.com/
Le 29 nov. 09 à 03:19, Warren Selby a écrit :
Do you have *11 registered in your voicemail.conf file? What does
the cli output look like when you try to leave a voicemail?
Thanks,
--Warren Selby
On Nov 28, 2009, at 7:22 PM, matthieu Nicaise <[email protected]
> wrote:
Hello everybody,
I'm using Asterisk ( 1.6.1.9 ) Voicemail.
For example, if i call extension *11 which is not registered, from
extension *12, i have no greetings at all, i only have the "please
leave a message after the beep".
I tried to record the busy, unavailable and temporary greetings for
extension *11 using VoiveMailMain and the file are well created on
the file system.
I cannot understand why those files are not played.
If i use VoiceMail(*11) in the extension.conf i have exactly the
same behaviour.
If i user VoiceMail(*11,b) the busy message is read.
Is that a normal behaviour ?
I can't understand why Asterisk is not using the Dial status
automaticaly.
Thank you for your help !
Matthieu NICAISE
Responsable technique
GSM : 06 72 19 09 55
[email protected]
------------------------------------------------------------------------
Thinkro System
http://www.thinkrosystem.com/
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