On 05/03/2011 01:16 PM, Gary Graves wrote:
Can you answer both?
Can Asterisk be configured to _initiate_ such a change at some point,
mid-call?
I don't know of a way to do that. I suppose it might be possible if a
call were asynchronously transferred to a SIP peer that had different
codec requirements.
and
Will Asterisk properly react to such a re-INVITE and change codecs if
asked to do so by the dialog counterparty?
It should.
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