[asterisk-dev] SIP codec selection. Can I select a codec and reinvite?

[Gregory Boehnlein]

> There is this super secret document in the Asterisk source code called
> "channelvariables.txt" (or README.variables if using 1.2.x) that
> contains information on an even more super secret channel variable
> called SIP_CODEC.  You might want to look into it.

That is one way to go about it. I actually see a Typo in that file..
(Missing "*" before the description of the variable! ;) I actually think
that might have been added after BJ hacked up that function for me.

To be clear, if you set the variable in the dialplan before you send the
call to the peer, will it override the settings in sip.conf as far as
allowed protocols?

Also, can you read that variable on an inbound call-leg? Or is it a set only
variable?