https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep 2009) | 3 lines
  
  When selecting DONT_OPTIMIZE in menuselect, explicitly pass -O0 to the compiler
  so we override any default optimization levels for a particular install.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220721 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Check for remotesecret option was unintentionally always true, which therefore
caused the secret option to never be used. Thanks to dvossel for pointing out
the exact fix.

(closes issue #15943)
Reported by: tpsast



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220718 65c4cc65-6c06-0410-ace0-fbb531ad65f3

*  Removed unneeded and uninitialized service_lock.
*  Fixed potential locking imbalance in pri_dchannel():PRI_EVENT_RESTART.
*  Fixed verbose message typo in pri_dchannel():PRI_EVENT_RESTART.


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Pointed out to me on IRC by snuff-home. Thanks


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Including version.h here causes this file to get recompiled after
every commit or update, which is not needed.



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ast_channel_get_full() tries to use OBJ_POINTER to optimize name-based
channel lookups, but this will not work properly when the channel's full
name was not supplied; for name-prefix searches, there is no value in
doing a hash-based lookup, and in fact doing so could result in many
channels being skipped.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220494 65c4cc65-6c06-0410-ace0-fbb531ad65f3

JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).

(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo

Review: https://reviewboard.asterisk.org/r/88/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Change the default behavior of Set, AGI, and pbx_realtime to 1.6 behavior by default (starting in 1.6.3).


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220417 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15939)
Reported by: dvossel

Review: https://reviewboard.asterisk.org/r/375/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220365 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Most of the functionality here is gained simply by setting the feature flag
on the bridge config. However, the dial limit functionality has been moved from
app_dial to the features code and has been made public so both app_dial and
the bridge app can use it.

(closes issue #13165)
Reported by: tim_ringenbach
Patches:
      app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540),
      modified by me


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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines
  
  Implicitly sending a progress signal breaks some applications.
  Call Progress() in your dialplan if you explicitly want progress to be sent.
  (Reverts change 216430, closes issue #15957)
  Reported by: Pavel Troller on the Asterisk-Dev mailing list
  http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
........


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep 2009) | 1 line
  
  Resolve parallel build warnings.
  
  Reported by Klaus Darilion on the asterisk-dev mailing list.
........


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu, 24 Sep 2009) | 2 lines
  
  Remove the remaining bashisms in the Makefile/mkpkgconfig
........


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009) | 7 lines
  
  mkpkgconfig does not need bash so make it use /bin/sh
  This fixes building on all systems that don't have bash
  at /bin/bash
  
  Reported by _ys on #asterisk-dev
  Tested by _ys on #asterisk-dev
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220028 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15739)
 Reported by: DLNoah, jeffg
 Patches: 
       20090914__issue15739.diff.txt uploaded by tilghman (license 14)
       20090922__issue15739.diff.txt uploaded by tilghman (license 14)
 Tested by: DLNoah, jeffg


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219987 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14818)
 Reported by: alea-soluciones
 Patches: 
       chan_mgcp-setvars-svn-trunk-r219899.patch uploaded by alea (license 514)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219952 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15946)
 Reported by: jtodd
 Patches: 
       leif-is-a-wuss.txt uploaded by jtodd (license 870)
 Tested by: jparker, tilghman, jtodd, russellb, mmichelson, seanbright, kpfleming, and the rest of the usual suspects


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219951 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This commit adds the doxygen changes that I've made to describe the Mantis
work flow documentation for the open source issue tracker. This should make
it easier to determine the flow of issues through the issue tracker, and what
those statuses mean.

(closes issue #15902)
Reported by: lmadsen
Patches:
      mantisworkflow.h uploaded by lmadsen (license 10)

Review: https://reviewboard.asterisk.org/r/367/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219895 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) | 10 lines
  
  When IMAP variables were changed during a reload, Voicemail did not use the new values.
  This change introduces a configuration version variable, which ensures that
  connections with the old values are not reused but are allowed to expire
  normally.
  (closes issue #15934)
   Reported by: viniciusfontes
   Patches: 
         20090922__issue15934.diff.txt uploaded by tilghman (license 14)
   Tested by: viniciusfontes
........


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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21 Sep 2009) | 3 lines
  
  Reverting merge 219520. This change was not necessary.
........


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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) | 8 lines
  
  Really stop the stream, when ast_closestream() is called.
  (closes issue #15129)
   Reported by: bmh
   Patches: 
         20090918__issue15129.diff.txt uploaded by tilghman (license 14)
   Review:
         https://reviewboard.asterisk.org/r/372/
........


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........
  r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009) | 6 lines
  
  Make sure the iax_pvt exists before dereferencing it.
  
  This fixes the latest crash posted on issue 15609.
  
  (issue #15609)
........


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  r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) | 9 lines
  
  iax2 frame double free
  
  The iax frame's retrans sched id was written over right
  before iax2_frame_free was called.  In iax2_frame_free that
  retrans id is used to delete the sched item.  By writing over
  the retrans field before the sched item could be deleted, it was
  possible for a retransmit to occur on a freed frame.
........


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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) | 14 lines
  
  via-header branches not updated correctly on INVITE
  
  INVITE requests must always contain a new unique branch id. When
  a new branch id is created for an INVITE, the dialog's invite_branch
  variable must be updated so CANCEL requests use the correct branch id.
  
  (closes issue #15262)
  Reported by: maniax
  Patches:
        asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608)
        invite_new_branch_trunk.diff uploaded by dvossel (license 671)
  Tested by: maniax, dvossel
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219451 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15696)
 Reported by: fhackenberger
 Patches: 
       maxsecs.patch uploaded by fhackenberger (license 592)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219412 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15340)
Reported by: lmsteffan
Patches:
      deadlock.patch uploaded by lmsteffan (license 779)
Tested by: lmsteffan



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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep 2009) | 6 lines
  
  Send a 100 Trying response when we detect a spiral.
  
  This was problematic during spiral tests at SIPit...
  along with some other things as well.
........


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) | 21 lines
  
  INVITE w/Replaces deadlock fix
  
  This patch cleans up the locking logic in chan_sip.c's
  handle_invite_replaces() function as well as making use
  of ast_do_masquerade() rather than forcing the masquerade
  on an ast_read().  The code had several redundant unlocks
  that would result in 'freed more times than we've locked!'
  errors. I cleaned these up as well as moving all the unlock
  logic to the end of the function.  This patch should also
  resolve the issue people were having with the replacecall
  channel never being unlocked with one legged calls.
  
  (closes issue #15151)
  Reported by: irroot
  Patches:
        invite_w_replaces_1.4.diff uploaded by dvossel (license 671)
  Tested by: irroot, dvossel
  
  Review: https://reviewboard.asterisk.org/r/371/
........


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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines
  
  Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down.
  
  This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list.  This fix makes the channel unavabile at the time when the CDR backend is invoked.  This has been documented in include/asterisk/cdr.h.
  
  (closes issue #15316)
  Reported by: vmarrone
  Tested by: mnicholson
  
  Review: https://reviewboard.asterisk.org/r/362/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219139 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In addition, there's a bit of cleanup to the arguments and documentation, in which
I discovered that the last feature added to this application duplicated an option
(oops!) and changed that option so that it now works.
(closes issue #14909)
 Reported by: junky
 Patches: 
       __20090901-spy_hangup_trunk.diff uploaded by lmadsen (license 10)
 Tested by: amilcar, junky, flujan, lmadsen


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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines
  
  Properly deal with quotes in the arguments of '#exec' includes.
  (closes issue #15583)
   Reported by: pkempgen
   Patches: 
         20090726__issue15583.diff.txt uploaded by tilghman (license 14)
         20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169)
   Tested by: pkempgen
........


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(closes issue #15017)
 Reported by: tzafrir
 Patches: 
       20090916__issue15017.diff.txt uploaded by tilghman (license 14)
 Tested by: tzafrir


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219007 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15359)
Reported by: snuffy
Patches:
      bug_res_jabber_unused_defines.diff uploaded by snuffy (license 35)


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