Hi,
It seems there is random behavior that causes screening mode to be
activated when a user calls and the line answered and then forwarded using
a dial command such as:
EXEC Dial SIP/13365551212@8x8|60SIP/13365541212@8x8
|60SIP/13365531212@8x8
I have a couple of questions about asterisk 1.6:
Can you change codecs mid-call upon re-invite?
Can you handle the SDP offer-answer in the 200-ACK instead of the usual
INVITE-200?
Thanks in advance for any insight.
Gary
--
...@evaristesys.comwrote:
On 05/03/2011 12:43 PM, Gary Graves wrote:
Can you change codecs mid-call upon re-invite?
Do you mean: can Asterisk be configured to _initiate_ such a change at
some point, mid-call? Or do you mean: Will Asterisk properly react to such
a re-INVITE and change codecs if asked to do so
