app_agent_pool: Return to dialplan if the agent fails to ack the call.

Improvements to the agent pool functionality.

* AgentRequest no longer hangs up the caller if the agent fails to connect
with the caller.  It now continues in the dialplan.

* AgentRequest returns AGENT_STATUS set to NOT_CONNECTED if the agent
failed to connect with the call.  Most likely because the agent did not
acknowledge the call in time or got disconnected.

* The agent alerting play file configured by the agent.conf custom_beep
option can now be disabled by setting the option to an empty string.  The
agent is effectively alerted to a call presence when MOH stops.

* Fixed bridge reference leak when the agent connects with a caller.

ASTERISK-23499 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3551/


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/12@414747 f38db490-d61c-443f-a65b-d21fe96a405b
res_pjsip_session: Fix leaked video RTP ports.

Simply enabling PJSIP to negotiage a video codec (e.g., h264) would leak
video RTP ports if the codec were not negotiated by an incoming call.

* Made add_sdp_streams() associate the handler with the media stream if
the handler handled the media stream.  Otherwise, when the
ast_sip_session_media object was destroyed it didn't know how to clean up
the RTP resources.

* Fixed sdp_requires_deferral() associating the handler with the media
stream when deciding if the SDP processing needs to be deferred for T.38.
Like the leaked video RTP ports, the T.38 handler needs to clean up
allocated resources from deciding if SDP processing needs to be deffered.

* Cleaned up some dead code in handle_incoming_sdp() and
sdp_requires_deferral().

ASTERISK-23721 #close
Reported by: cervajs

Review: https://reviewboard.asterisk.org/r/3571/


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/12@414749 f38db490-d61c-443f-a65b-d21fe96a405b