Fix a bug with applying the end result of the codec negotiation to the Asterisk channel.


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@389567 f38db490-d61c-443f-a65b-d21fe96a405b
Fix a bug where the DTMF mode was not set on newly created RTP instances in the res_sip_sdp_rtp module.


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@389568 f38db490-d61c-443f-a65b-d21fe96a405b
Fix inverted test preventing DTMF disconnect from working.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@389569 f38db490-d61c-443f-a65b-d21fe96a405b