[Asterisk-Users] SER + ASTERISK voicemail

[harry gaillac]

Hello,

Thanks for help it's ok with static file
voicemail.conf
However something is wrong with ARA .

app_voicemail search entries in voicemail.conf ?!
I set apps/Makefile for USE_ODBC_STORAGE.


Regards
Harry
//////////////////////////////////////////////////////
Connected to Asterisk CVS-HEAD currently running on
serveur1 (pid = 2584)
Verbosity is at least 3
    -- Executing VoiceMail("SIP/asterisk-8db8", "b84")
in new stack
Aug 29 16:11:40 WARNING[7947]: app_voicemail.c:2602
leave_voicemail: No entry in voicemail config  file
for '84'
Aug 29 16:11:50 WARNING[7947]: pbx.c:2336
__ast_pbx_run: Timeout, but no rule 't' in context
'loc al'
serveur1*CLI> odbc show
Name: asterisk
DSN: asterisk
Connected: yes
serveur1*CLI>
///////////////////////////////////////////////////////
--- Steve Blair <blairs at isc.upenn.edu> a écrit :

> 
> You'll want some rules in your sip.conf to handle
> the connection from 
> SER. A
> starting point might be:
> 
>    [<ser ip addr>:<ser port ?= 5060>]
>    type=peer
>    context=<my sip context name>
>    tos=lowdelay                    ; tos delay
>    allow=ulaw                     ; dtmfmode=inband
> only works with ulaw 
> or alaw!
>    dtmfmode=inband                ; Choices are
> inband, rfc2833, or info
> 
> You'll then want some rules in extensions.conf to
> accept the call and 
> redirect it
> to mailboxes defined in your voicemail.conf or in
> MySQL. Something like:
> 
>    [general]
>    context=<my sip context name>
>    switch => Realtime/<my sip context
> name>@extensions
>    static=yes
> 
>   [<my sip context name>]
> 
>   exten => _uXXXXX,1,VoiceMail(${EXTEN}@<my sip
> context name>)
>   exten => _XXXXX,1,VoiceMail(${EXTEN}@<my sip
> context name>)
>   exten => _bXXXXX,1,VoiceMail(${EXTEN}@<my sip
> context name>))
>   exten => #,2,Hangup                     ; Hang
> them up.
> 
> Steve
> 
> harry gaillac wrote:
> 
> >Hello,
> >
> >I try set Ua---SER----Asterisk (voicemail/ARA)
> >                |
> >               Ua
> >ser stable
> >asterisk cvs head 
> >
> >I read
>
>http://mail.iptel.org/pipermail/serusers/2005-February/015997.html
> >to forward unavailable or busy sip agents to
> asterisk
> >voicemail in failure route.
> >
> >How may I configure extensions.conf and ser.cfg ?
> >I have been trying without success!
> >
> >Regards
> >Harry
> >
> >
> >	
> >
> >	
> >		
>
>___________________________________________________________________________
> 
> >Appel audio GRATUIT partout dans le monde avec le
> nouveau Yahoo! Messenger 
> >Téléchargez cette version sur
> http://fr.messenger.yahoo.com
> >_______________________________________________
> >--Bandwidth and Colocation sponsored by
> Easynews.com --
> >
> >Asterisk-Users mailing list
> >Asterisk-Users at lists.digium.com
>
>http://lists.digium.com/mailman/listinfo/asterisk-users
> >To UNSUBSCRIBE or update options visit:
> >  
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> >  
> >
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com
> --
> 
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 



	

	
		
___________________________________________________________________________ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.yahoo.com