[Asterisk-Users] reinvite, DISA, and switching codec's.

[Luki]

James,

> Am I right in saying that because Asterisk has Answer()'d the call and
> done DISA(...), I can't do a re-invite to bridge the call between the
> PAP2 and the VoIP provider?

Yes, you can reinvite after Dial()'ing your provider, but you probably
won't be able to switch codecs once the call is connected. I may be
wrong so just try it :). The ATA must be able to talk directly to your
provider in such a case (i.e. not NAT).

--Luki