Properly load say.conf upon reload of module app_playback.

If say.conf did not exists prior to originally loading module app_playback it
would not load on subsequent reloads of the module once it had been created.
This occurred because upon reload of the app_playback module it would only
load a new configuration if an old one had previously existed.  This fix simply
removed the association between checking if an old configuration existed and
the loading of the new one.

(closes issue ASTERISK-20800)
Reported by: pgoergler
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Merged revisions 381216 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@381217 f38db490-d61c-443f-a65b-d21fe96a405b
Fix some more REF_DEBUG-related build errors

When sip_ref_peer and sip_unref_peer were exported to be usable in
channels/sip/security_events.c, modifications to those functions when
building under REF_DEBUG were not taken into account. This change
moves the necessary defines into sip.h to make them accessible to
other parts of chan_sip that need them.


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@381282 f38db490-d61c-443f-a65b-d21fe96a405b
Do not allow native RTP bridging if packetization of media streams differs.

The RTP engine will no longer allow for local and remote native RTP bridges
if packetization of streams differs. Allowing native bridging in this scenario
has been known to cause FAX failures.

(closes ASTERISK-20650)
Reported by: Maciej Krajewski
Patches:
	ASTERISK-20659.patch uploaded by Mark Michelson (License #5049)

Review: https://reviewboard.asterisk.org/r/2319
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Merged revisions 381281 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@381306 f38db490-d61c-443f-a65b-d21fe96a405b