[asterisk: 3/5] Merge branch 'f15'

[Jeffrey C. Ollie]

commit 71c325b9ab92a2454ce947010837a5aa6ded01eb
Merge: 3d55dc6 c981d06
Author: Jeffrey C. Ollie <jeff at ocjtech.us>
Date:   Fri Jul 1 10:14:00 2011 -0500

    Merge branch 'f15'
    
    Conflicts:
    	asterisk.spec

 asterisk.spec |   43 +++++++++++++++++++++++++++++++++++++------
 1 files changed, 37 insertions(+), 6 deletions(-)
---
diff --cc asterisk.spec
index 1218e4f,412ac4a..e3f3a6c
--- a/asterisk.spec
+++ b/asterisk.spec
@@@ -1223,59 -1251,9 +1251,62 @@@ f
  %{_libdir}/asterisk/modules/app_voicemail_plain.so
  
  %changelog
 +* Thu Jun 30 2011 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.8.5-0.1.rc1
 +- The Asterisk Development Team has announced the first release candidate of
 +- Asterisk 1.8.5. This release candidate is available for immediate download at
 +- http://downloads.asterisk.org/pub/telephony/asterisk/
 +-
 +- The release of Asterisk 1.8.5-rc1 resolves several issues reported by the
 +- community and would have not been possible without your participation.
 +- Thank you!
 +-
 +- The following is a sample of the issues resolved in this release candidate:
 +-
 +- * Fix Deadlock with attended transfer of SIP call
 +-  (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,
 +-   cmaj)
 +-
 +- * Fixes thread blocking issue in the sip TCP/TLS implementation.
 +-  (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois,
 +-   rossbeer, kowalma, Freddi_Fonet)
 +-
 +- * Be more tolerant of what URI we accept for call completion PUBLISH requests.
 +-  (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)
 +-
 +- * Fix a nasty chanspy bug which was causing a channel leak every time a spied on
 +-  channel made a call.
 +-  (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
 +-
 +- * This patch fixes a bug with MeetMe behavior where the 'P' option for always
 +-  prompting for a pin is ignored for the first caller.
 +-  (Closes issue #18070. Reported by mav3rick. Patched by bbryant)
 +-
 +- * Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If
 +-  the call that the dialplan started an AGI script for is hungup while the AGI
 +-  script is in the middle of a command then the AGI script is not notified of
 +-  the hangup.
 +-  (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
 +-
 +- * Resolve issue where leaving a voicemail, the MWI message is never sent. The
 +-  same thing happens when checking a voicemail and marking it as read.
 +-  (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard
 +-   Mudgett)
 +-
 +- * Resolve issue where wait for leader with Music On Hold allows crosstalk
 +-  between participants. Parenthesis in the wrong position. Regression from issue
 +-  #14365 when expanding conference flags to use 64 bits.
 +-  (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)
 +-
 +- * Fix timerfd locking issue.
 +-  (Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz)
 +-
 +- For a full list of changes in this release candidate, please see the ChangeLog:
 +-
 +- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5-rc1
 +
+ * Fri Jul  1 2011 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.8.4.4-2
+ - Fix systemd dependencies in EL6 and F15
+ 
  * Thu Jun 30 2011 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.8.4.4-2
  - Fedora Directory Server -> 389 Directory Server