[Asterisk-Users] Clustering "NEW THREAD", Almost Working

[Wai Wu]

That is a show stopper. However, if your clients are in groups behind
their respected router, you might be able to give them a little linux
app such that this app can PERSONIFY the phones to send a packet to the
respected server.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Benjamin
Lawetz
Sent: Tuesday, March 14, 2006 1:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Clustering "NEW THREAD", Almost Working

Had this working also at some point, but had one killer problem... NAT
issues! Most of our clients are natted, and depending on the router,
they only allow traffic to return from the server that the traffic was
sent to.
So the invites coming from other servers were being dropped.

But besides that worked like a charm.

Ben 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of JR
Richardson
Sent: March 14, 2006 12:57 PM
To: asterisk-users at lists.digium.com
Subject: RE: [Asterisk-Users] Clustering "NEW THREAD", Almost Working

Yes, SIP realtime is working with multiple * servers all accessing the
same MySQL database, add a sip phone in the database and the phone can
register with any server without the need to configure any server, just
add the phone in the database, petty cool. 

JR

------------------------------

Message: 21
Date: Tue, 14 Mar 2006 10:18:06 -0500
From: Wai Wu <wwu at Calltrol.com>
Subject: RE: [Asterisk-Users] Clustering "NEW THREAD", Almost Working
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <B0430B20D208514CB2AFF57E81645C3194CC at k3-1.Calltrol.com>
Content-Type: text/plain; charset="us-ascii"

Now, I know what you guys been talking about. It is like DSN for sip
phones, not really clustering. I original thought that you guys want to
setup some thing that can fail over to a different sip server if the
server running the IVR dies.
 
 
 From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Douglas
Garstang
Sent: Tuesday, March 14, 2006 12:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Clustering "NEW THREAD", Almost Working


Holy crap. You got SIP realtime working? I've tried it twice before and
it failed the same way twice. Do you have multiple Asterisk boxes
accessing the same sip info (ie phones) in the same table on the same
database? Digium has said numerous times this known not to work,
although I cant' work out why as it's just reading from a common table.


JR Richardson
Engineering for the Masses

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