According to Kevin Fleming, this is not supported.

        -----Original Message----- 
        From: unplug [mailto:[EMAIL PROTECTED] 
        Sent: Tue 6/20/2006 10:03 PM 
        To: Asterisk Users Mailing List - Non-Commercial Discussion 
        Cc: 
        Subject: Re: [Asterisk-Users] asterisk load balance
        
        

        I am confusing where the asterisk should store the register
        information in realtime mode.
        As in my configuration,
        UA1 ---- asterisk1 +
        UA2 ---- asterisk2 + database
        UA3 ----- asterisk3 +
        3 UAs connected to 3 asterisk with a common database to store user
        information and dial plan.  However, asterisk1 seems doesn't know
        there are UA2 and UA3 already registered in the system.
        I wonder the register information should be store in DB.  When there
        is a invite request, asterisk will query the database and find out the
        calling party contact information.  Am I right?  But in the case
        above, asterisk only know the UA which register to it.  Anyone can
        tell me the real mechanism of realtime for the UA registration?  How
        and where asterisk to get the user registration when there is an
        invite comming?
        
        On 6/18/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:
        > On Sat, 17 Jun 2006, Douglas Garstang wrote:
        >
        > > Good grief I hate Outlook webmail. I can't reply inline.
        > Switch to thunderbird ;)
        >
        > >
        > > Anyway, I disagree that all state info except hinting can be 
replicated. What about call transfers? If a call is sitting on pbx1, and the 
user transfers a call, if it goes to pbx2, Asterisk will complain that it 
cannot transfer the call as it doesn't know anything about it
        >
        > Well, I'm not sure what the problem with call transfers is.  We have 
two
        > registration servers, in which the phones can and do register with 
either
        > server.  If one phone makes a call on one server, they can complete 
the
        > call with anyone else on their server, plus anyone on the other 
servers.
        > The server just treats the transfer and bridge like any other phone 
call.
        > If the phone is on another server, it hands off the conversation to 
that
        > server after the transfer.
        >
        > And I think I'll address your NFS problems.  Are you doing that for
        > redundancy's sake or just for MWI?  If it's just for MWI, then you 
might
        > be better off setting up some scripts that drop some msgXXXX.txt 
files in
        > the user's "voicemail" box on the registration servers.  No need to
        > replicate registration to the voicemail server, that's just extra 
unneeded
        > traffic.  Plus, with something like that, you don't have to worry 
about
        > the voicemail nfs share dying and bringing down the asterisk network. 
 If
        > it's for redundancy, set up another voicemail server or two, and use 
DRBD
        > or some sort of sync tool between them, with the MWI script and you'll
        > have fixed the redundancy problem.
        >
        >
        > --
        > Aaron Daniel
        > Computer Systems Technician
        > Sam Houston State University
        > [EMAIL PROTECTED]
        > (936) 294-4198
        > _______________________________________________
        > --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
        >
        _______________________________________________
        --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
        

_______________________________________________
--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

Reply via email to