Queue DTMF frames to the channel if the call is terminated by Asterisk core or application

Forwarding RTP telephone event packets transparently to the network caused a regression that DTMF digits can't be read from ast_channel if the call is terminated by Asterisk core or applications, e.g. VoiceMailMain, call forwarding feature access codes, and etc. Now with this fix, DTMF frames will be queued to ast_channel simultaneously forwarding those RTP telephone event packets transparently.

Also revert 44071bb7. "Fix a regression that is introduced by RTP events generated by DSP"