[asterisk-users] VoiceMail greetings

[matthieu Nicaise]

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 [*11 at 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 [*11 at 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 [h at 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
technique at thinkrosystem.com
------------------------------------------------------------------------
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 <technique at thinkrosystem.com 
> > 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
>> technique at thinkrosystem.com
>> ------------------------------------------------------------------------
>> Thinkro System
>> http://www.thinkrosystem.com/
>>
>>
>>
>>
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