asterisk users - May 2011 - Asterisk 1.6 Questions

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
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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 by the 
dialog counterparty?
> Can you handle the SDP offer-answer in the 200-ACK instead of the
> usual INVITE-200?
Doesn't seem to.  Looking at chan_sip.c in 1.6.2.13, there is no call 
to add_sdp() that is not made either in the context of 1) an initial 
INVITE request or 2) a re-INVITE or 3) the construction of a response. 
  Nothing in the case of the production of an end-to-end ACK.

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