[Asterisk-Users] Proxied SIP
Chris Tooley
ctooley at gmail.com
Sun Feb 6 12:15:38 MST 2005
I have been playing with SER, but we need something to proxy the
initial part of an origination (outbound) call, and then negotiate
it's way out of the middle of the call. Like redirect appears to work
with calls that are termination (inbound).
Chris
On Sun, 06 Feb 2005 08:13:02 -0500, Todd Lieberman
<lists at tlsolutions.net> wrote:
> Chris Tooley wrote:
>
> >I want to install Asterisk for an organization that wants it to do
> >some call routing for them. They have a SIP provider that is going to
> >provide one termination and one origination account.
> >
> >We are going to have to route a rather large number of calls
> >(50-100,000 concurrent), but can't find any information on how to
> >proxy calls adaquately.
> >_______________________________________________
> >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
> >
> >
> look to SER but 100,000 calls requires a tremendious amount of
> bandwidth, make sure you have it!
>
More information about the asterisk-users
mailing list