[Asterisk-Users] Passing audio stream through Asterisk or not?

[Steven Critchfield]

On Sat, 2003-05-31 at 08:06, Dan wrote:
> Hi all,
>  
> One short question.
> When one extension (let's say ATA-186, SIP image, G.723 codec
> selected) try to call an external SIP address like:
> SIP/user at domain.com, where another identical ATA-186 is available with
> G.723 codec selectrd,
> after the signaling phase, the call is established through Asterisk or
> directly between the two ATAs?
> There is no G.723 codec available on Asterisk
> I need to know this because of the firewall.

if you turn off the reinvite in the asterisk configs for those ata186s
then it will pass through asterisk even if asterisk doesn't understand
the codec.

-- 
Steven Critchfield <critch at basesys.com>