Reverting r411189 so that it can be put up for public review

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  r411189 | jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines

  chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)

  Prior to this patch, the P-Asserted-Identity header would include anonymous
  caller id information which seems to go against the point of the
  P-Asserted-Identity header. Now the real caller ID information will be
  included in this header. Also, no privacy header would be included.
  This patch adds 'Privacy: id' to outgoing SIP messages that include the
  P-Asserted-Identity header.

  (closes issue AST-1301)
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Merged revisions 412328 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 412329 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 412330 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@412331 f38db490-d61c-443f-a65b-d21fe96a405b
chan_sip.c: Moved some sip_pvt unrefs after their last use.

* Moved sip_pvt unref in ast_hangup() and handle_request_do() to the end
of the function.  The unref needs to happen after the last use of the
pointer.
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Merged revisions 412348 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 412383 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@412384 f38db490-d61c-443f-a65b-d21fe96a405b
chan_sip.c: Fix channel staging assertion failure.

The failing assertion ensures that the final snapshot gets generated so
CDR records can get finalized.  The only place where a channel staging
snapshot flag could be left set is in chan_sip.c:handle_request_bye().
The function could return before clearing the flag because the channel
could dissappear while the function had to have the channel unlocked.

* Fixed handle_request_bye() channel snapshot staging coverage area to not
have a return in the middle of it and be unable to clear the staging flag.

* Pushed the channel snapshot staging coverage area into
ast_rtp_instance_set_stats_vars() to ensure that the staging is not
interrutped.

* Made callers of ast_rtp_instance_set_stats_vars() not call it with any
channels or channel driver private locks held to eliminate the deadlock
potential.  The callers must hold references to the passed in channel and
rtp objects.

* Eliminated sip_hangup() trying to get the bridge peer.  It is futile at
this point because the channel could never be in a bridge.

Review: https://reviewboard.asterisk.org/r/3431/
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Merged revisions 412385 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@412386 f38db490-d61c-443f-a65b-d21fe96a405b