[asterisk-users] Problem with VoiceMailMain

2008-03-31 Thread mark morreny 
Dear all,

I noticed a very strange problem.  When I tried using VoiceMailMain to
record my unavailable message, the file does not get created even though I
can find the corresponding mssage from asterisk:

-- SIP/2001-b6307d78 Playing 'beep' (language 'en')
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/2000/unavail.tmp format: wav,
0x82828c8
-- User ended message by pressing #
-- SIP/2001-b6307d78 Playing 'auth-thankyou' (language 'en')
-- SIP/2001-b6307d78 Playing 'vm-review' (language 'en')
-- Saving message as is
-- SIP/2001-b6307d78 Playing 'vm-msgsaved' (language 'en')
-- SIP/2001-b6307d78 Playing 'vm-options' (language 'en')

Also, if I copies the 'unavil.wav' inside the /2000/ directory myself, it
gets deleted somehow.  How come this is happening?
What I want to do is to be able to copies some standard voicemail for
different extension, bu so far, it does not seem to work for me yet.

Can anyone give me some suggestion?

Thank you very much for your kind help.

Thanks,
Mark
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] problem with VoiceMailMain

2006-12-29 Thread Dima Pursanov 
Hi all:)
Can you answer,how to change parameters of VoiceMailMain application?(for 
example:i dont want to give permission to change password and etc)
___
--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

[Asterisk-Users] Problem with VoicemailMain

2006-06-13 Thread Ricardo Carvalho 

Hi,

I'm running SER with Asterisk, and I've configured VoicemailMain like this:

exten = 201,1,VoicemailMain(@default)
exten = 201,2,Hangup()

Although, after any user enter his voicemailmain mailbox, when the phone 
is hung up, the call still continues running in Asterisk, because I can 
see it in the debug output of the Asterisk CLI. The call only stops if 
before hung up, I press #.

What is causing this? Any ideas?

Regards,

Ricardo.
___
--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