[Asterisk-Users] Proxied SIP

[Radovan.Mihalik]

I have solved this with IAX loop on asterisk.
Internal SIP phone - SIP asterisk inbound - IAX call to the same 
Asterisk box - Outbound SIP call.

Calls are bridged and outgoing calls are appear to be from asterisk
IP. This solves also incoming calls.
 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Chris
Tooley
Sent: Sunday, February 06, 2005 8:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Proxied SIP

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!
>
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