[asterisk-users] Asterisk 1.6 Questions

[Alex Balashov]

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.

-- 
Alex Balashov - Principal
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