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

[Dan]

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.

Thanks,
Dan
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