You could use a sip proxy front end like Kamailio.
Sent from my iPhone
On Nov 18, 2010, at 7:39 AM, Antônio Theóphilo anto...@freeddom.com wrote:
Hi All
Does anyone know about any tool that does to Asterisk what mod_jk does for
JBoss/Tomcat: a load-balance/failover server that is
Thank you for the answer Darren.
In fact I have an application that requests a call to a real person through an
AMI interface and get some client information. Using a SIP Proxy is an option
but I prefer that the interface between the app and the Asterisk could be the
AMI (or HTTP).
Regards
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
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,
On Fri, 16 Jun 2006, Douglas Garstang wrote:
Unless you can guarantee that the system that is currently processing a call
will be the system that handles a transfer request from a phone, are the same,
then transfers will not work.
Incorrect. Transfers work fine between multiple asterisk
That sounds fine except where registrations are involved although I'd suggest you look into SRV as well as RR for the DNS to more finely balance the load for clients which support it. Doug's mail says it all where registrations are involved - not all state information is stored in the database so
as it doesn't know anything about it
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Fri 6/16/2006 11:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: RE: [Asterisk-Users] asterisk load
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,
Unless you can guarantee that the system that is currently processing a call
will be the system that handles a transfer request from a phone, are the same,
then transfers will not work.
Round robin DNS won't work at all. Every time you send out a SIP message, your
going to be sending it to a
