[asterisk-commits] lmadsen: tag 1.6.1.0-rc4 r186326 - /tags/1.6.1.0-rc4/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Apr 3 11:12:35 CDT 2009


Author: lmadsen
Date: Fri Apr  3 11:12:31 2009
New Revision: 186326

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=186326
Log:
Importing files for 1.6.1.0-rc4 release.

Added:
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    tags/1.6.1.0-rc4/.version   (with props)
    tags/1.6.1.0-rc4/ChangeLog   (with props)

Added: tags/1.6.1.0-rc4/.lastclean
URL: http://svn.digium.com/svn-view/asterisk/tags/1.6.1.0-rc4/.lastclean?view=auto&rev=186326
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Added: tags/1.6.1.0-rc4/ChangeLog
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--- tags/1.6.1.0-rc4/ChangeLog (added)
+++ tags/1.6.1.0-rc4/ChangeLog Fri Apr  3 11:12:31 2009
@@ -1,0 +1,55890 @@
+2009-04-03  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.1.0-rc4 released.
+
+2009-04-03 15:54 +0000 [r186323]  Joshua Colp <jcolp at digium.com>
+
+	* include/asterisk/crypto.h, /: Merged revisions 186321 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri,
+	  03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5
+	  lines Fix a problem with the crypto variable definitions not
+	  actually being defined properly. (closes issue #14804) Reported
+	  by: jvandal ........ ................
+
+2009-04-03 14:33 +0000 [r186288]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr
+	  2009) | 20 lines Fix the ability to retrieve voicemail messages
+	  from IMAP. A recent change made interactive vm_states no longer
+	  get added to the list of vm_states and instead get stored in
+	  thread-local storage. In trunk and all the 1.6.X branches, the
+	  problem is that when we search for messages in a voicemail box,
+	  we would attempt to update the appropriate vm_state struct by
+	  directly searching in the list of vm_states instead of using the
+	  get_vm_state_by_imap_user function. This meant we could not find
+	  the interactive vm_state that we wanted. (closes issue #14685)
+	  Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson
+	  (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........
+
+2009-04-03 02:06 +0000 [r186232]  Russell Bryant <russell at digium.com>
+
+	* cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009)
+	  | 29 lines Merged revisions 186229 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009)
+	  | 21 lines Fix a memory leak in cdr_radius. I came across this
+	  while doing some testing of my ast_channel_ao2 branch. After
+	  running a test overnight that generated over 5 million calls,
+	  Asterisk had taken up about 1 GB of my system memory. So, I
+	  re-ran the test with MALLOC_DEBUG turned on. However, it showed
+	  no leaks in Asterisk during the test, even though Asterisk was
+	  still consuming it somehow. Instead, I turned to valgrind, which
+	  when run with --leak-check=full, told me exactly where the leak
+	  came from, which was from allocations inside the radiusclient-ng
+	  library. This explains why MALLOC_DEBUG did not report it. After
+	  a bit of analysis, I found that we were leaking a little bit of
+	  memory every time a CDR record was passed to cdr_radius. I don't
+	  actually have a radius server set up to receive CDR records.
+	  However, I always have my development systems compile and install
+	  all modules. In addition to making sure there are not build
+	  errors across modules, always loading modules helps find bugs
+	  like this, too, so it is strongly recommend for all developers.
+	  ........ ................
+
+2009-04-02 21:59 +0000 [r186177]  Mark Michelson <mmichelson at digium.com>
+
+	* configs/features.conf.sample, /: Merged revisions 186175 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500
+	  (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr
+	  2009) | 5 lines Fix instructions in one-step parking comment to
+	  make more sense. Changed a capital K to a lowercase k. ........
+	  ................
+
+2009-04-02 17:27 +0000 [r186108]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500
+	  (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02
+	  Apr 2009) | 3 lines ensure that the buffer passed to
+	  DAHDI_SET_BUFINFO is fully initialized ........ ................
+
+2009-04-02 17:14 +0000 [r186062]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+	  186060 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009)
+	  | 16 lines Merged revisions 186059 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500
+	  (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
+	  ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02
+	  Apr 2009) | 2 lines Fix for AST-2009-003 ........
+	  ................ ................
+
+2009-04-02 13:53 +0000 [r185956]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500
+	  (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr
+	  2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and
+	  DAHDI_GET_PARAMS ioctls were recently corrected to show that they
+	  do, in fact, read data from userspace as part of their work. due
+	  to this fix, valgrind now reports a number of cases where
+	  chan_dahdi passed an uninitialized (or partially) buffer to these
+	  ioctls, which could lead to unexpected behavior. this patch
+	  corrects chan_dahdi to ensure that buffers passed to these ioctls
+	  are always fully initialized. ........ ................
+
+2009-04-01 19:06 +0000 [r185848]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009)
+	  | 16 lines Merged revisions 185845 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009)
+	  | 10 lines Fixes issue with dropped calles due to re-Invite glare
+	  and re-Invites never executing after a 491 Acknowledgement for
+	  491 responses were never being processed because it didn't match
+	  our pending invite's seqno. Since the ACK was never processed,
+	  the 491 frame would continue to be retransmitted until eventually
+	  the call was dropped due to max retries. Now during a pending
+	  invite, if we receive another invite, we send an 491 and hold on
+	  to that glare invite's seqno in the "glareinvite" variable for
+	  that sip_pvt struct. When ACK's are received, we first check to
+	  see if it is in response to our pending invite, if not we check
+	  to see if it is in response to a glare invite. In this case, it
+	  is in response to the glare invite and must be dealt with or the
+	  call is dropped. I've changed the wait time for resending the
+	  re-Invite after receving a 491 response to comply with RFC 3261.
+	  Before this patch the scheduled re-Invite would only change a
+	  flag indicating that the re-Invite should be sent out, now it
+	  actually sends it out as well. (closes issue #12013) Reported by:
+	  alx Review: http://reviewboard.digium.com/r/213/ ........
+	  ................
+
+2009-04-01 13:50 +0000 [r185774]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c, /: Merged revisions 185772 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009)
+	  | 14 lines Merged revisions 185771 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009)
+	  | 6 lines Fix a case where DTMF could bypass audiohooks. This
+	  change fixes a situation where an audiohook that wants DTMF would
+	  not actually get it. This is in the code path where we end DTMF
+	  digit length emulation while handling a NULL frame. ........
+	  ................
+
+2009-03-31 22:38 +0000 [r185666]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* utils, /: Merged revisions 185664 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 |
+	  kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line
+	  ignore copied (generated) file ........
+
+2009-03-31 22:05 +0000 [r185471-185602]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c, /: Merged revisions 185600 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar
+	  2009) | 12 lines Merged revisions 185599 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar
+	  2009) | 6 lines Fix crash that would occur if an empty member was
+	  specified in queues.conf. (closes issue #14796) Reported by: pida
+	  ........ ................
+
+	* apps/app_voicemail.c, /: Merged revisions 185469 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500
+	  (Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar
+	  2009) | 8 lines Fix Russian voicemail intro to say the word
+	  "messages" properly. (closes issue #14736) Reported by: chappell
+	  Patches: voicemail_no_messages.diff uploaded by chappell (license
+	  8) ........ ................
+
+2009-03-31 17:48 +0000 [r185427]  David Brooks <dbrooks at digium.com>
+
+	* /, channels/chan_gtalk.c: Merged revisions 185363 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500
+	  (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009)
+	  | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains
+	  extra whitespaces To drill into the xmpp to find the capabilities
+	  between channels, chan_gtalk calls iks_child() and iks_next().
+	  iks_child() and iks_next() are functions in the iksemel xml
+	  parsing library that traverse xml nodes. The bug here is that
+	  both iks_child() and iks_next() will return the next iks_struct
+	  node *regardless* of type. chan_gtalk expects the next node to be
+	  of type IKS_TAG, which in most cases, it is, but in this case (a
+	  call being made from the Empathy IM client), there exists
+	  iks_struct nodes which are not IKS_TAG data (they are extraneous
+	  whitespaces), and chan_gtalk doesn't handle that case, so
+	  capabilities don't match, and a call cannot be made.
+	  iks_first_tag() and iks_next_tag(), on the other hand, will not
+	  return the very next iks_struct, but will check to see if the
+	  next iks_struct is of type IKS_TAG. If it isn't, it will be
+	  skipped, and the next struct of type IKS_TAG it finds will be
+	  returned. This assures that chan_gtalk will find the iks_struct
+	  it is looking for. This fix simply changes all calls to
+	  iks_child() and iks_next() to become calls to iks_first_tag() and
+	  iks_next_tag(), which resolves the capability matching. The
+	  following is a payload listing from Empathy, which, due to the
+	  extraneous whitespace, will not be parsed correctly by iksemel:
+	  <iq from='dbrooksjab at 235-22-24-10/Telepathy'
+	  to='astjab at 235-22-24-10/asterisk' type='set' id='542757715704'>
+	  <session xmlns='http://www.google.com/session'
+	  initiator='dbrooksjab at 235-22-24-10/Telepathy' type='initiate'
+	  id='1837267342'> <description
+	  xmlns='http://www.google.com/session/phone'> <payload-type
+	  clockrate='16000' name='speex' id='96'/> <payload-type
+	  clockrate='8000' name='PCMA' id='8'/> <payload-type
+	  clockrate='8000' name='PCMU' id='0'/> <payload-type
+	  clockrate='90000' name='MPA' id='97'/> <payload-type
+	  clockrate='16000' name='SIREN' id='98'/> <payload-type
+	  clockrate='8000' name='telephone-event' id='99'/> </description>
+	  </session> </iq> Review: http://reviewboard.digium.com/r/181/
+	  ........ ................
+
+2009-03-31 14:57 +0000 [r185263]  Russell Bryant <russell at digium.com>
+
+	* apps/app_queue.c, /: Merged revisions 185261 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 |
+	  russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines
+	  Don't free() an astobj2 object. (closes issue #14672) Reported
+	  by: makoto ........
+
+2009-03-31 14:10 +0000 [r185199]  Joshua Colp <jcolp at digium.com>
+
+	* /, main/audiohook.c: Merged revisions 185197 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) |
+	  15 lines Merged revisions 185196 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8
+	  lines Fix crash when moving audiohooks between channels. Handle
+	  the scenario where we are called to move audiohooks between
+	  channels and the source channel does not actually have any on it.
+	  (closes issue #14734) Reported by: corruptor ........
+	  ................
+
+2009-03-30 20:50 +0000 [r185126-185127]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/misdn_config.c, /, configs/misdn.conf.sample: Merged
+	  revisions 185123 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009)
+	  | 9 lines Merged revisions 185121 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009)
+	  | 1 line Update the channel allocation method documentation.
+	  ........ ................
+
+	* channels/misdn/isdn_lib.c, /: Merged revisions 185122 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500
+	  (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009)
+	  | 19 lines Make chan_misdn BRI TE side normally defer channel
+	  selection to the NT side. Channel allocation collisions are not
+	  handled by chan_misdn very well. This patch simply avoids the
+	  problem for BRI only. For PRI, allocation collisions are still
+	  possible but less likely since there are simply more channels
+	  available and each end could use a different allocation strategy.
+	  misdn.conf options available: te_choose_channel - Use to force
+	  the TE side to allocate channels. method - Specify the channel
+	  allocation strategy. (closes issue #13488) Reported by:
+	  Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich
+	  Tested by: crich, siepkes, festr ........ ................
+
+2009-03-30 16:47 +0000 [r185088]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c, /: Merged revisions 185072 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar
+	  2009) | 45 lines Merged revisions 185031 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar
+	  2009) | 39 lines Fix queue weight behavior so that calls in
+	  low-weight queues are not inappropriately blocked. (This is
+	  copied and pasted from the review request I made for this patch)
+	  Asterisk has some odd behavior when queue weights are used. The
+	  current logic used when potentially calling a queue member is: If
+	  the member we are going to call is part of another queue and
+	  _that other queue has any callers in it_ and has a higher weight
+	  than the queue we are calling from, then don't try to contact
+	  that member. The issue here is what I have marked with
+	  underscores. If the higher-weighted queue has any callers in it
+	  at all, then the queue member will be unreachable from the
+	  lower-weighted queue. This has the potential to be really really
+	  bad if using a queue strategy, such as leastrecent or
+	  fewestcalls, with the potential to call the same member
+	  repeatedly. The fix proposed by garychen on issue 13220 is very
+	  simple and, as far as I can see, works well for this situation.
+	  With this set of changes, the logic used becomes: If the member
+	  we are going to call is part of another queue, the other queue
+	  has a higher weight than the queue we are calling from, and the
+	  higher weight queue has at least as many callers as available
+	  members, then do not try to contact the queue member. If the
+	  higher weighted queue has fewer callers than available members,
+	  then there is no reason to deny the call to this member since the
+	  other queue can afford to spare a member. Since the fix involved
+	  writing a generic function for determining the number of
+	  available members in the queue, I also modified the is_our_turn
+	  function to make use of the new num_available_members function to
+	  determine if it is our turn to try calling a member. There is one
+	  small behavior change. Before writing this patch, if you had
+	  autofill disabled, then if you were the head caller in a queue,
+	  you would automatically be told that it was your turn to try
+	  calling a member. This did not take into account whether there
+	  were actually any queue members available to take the call. Now
+	  we actually make sure there is at least one member available to
+	  take the call if autofill is disabled. (closes issue #13220)
+	  Reported by: garychen Review:
+	  http://reviewboard.digium.com/r/202/ ........ ................
+
+2009-03-30 14:41 +0000 [r184950]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 184948 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) |
+	  21 lines Merged revisions 184947 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) |
+	  14 lines Improve our handling of T38 in the initial INVITE from a
+	  device. We now answer with matching media streams to what is
+	  requested. If an INVITE is received with both a T38 and RTP media
+	  stream this means we answer with both. For any outgoing calls
+	  created as a result of this inbound one no T38 is requested in
+	  the initial INVITE. Instead if we start receiving udptl packets
+	  we trigger a reinvite on the outbound side. (closes issue #12437)
+	  Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu
+	  Review: http://reviewboard.digium.com/r/208/ ........
+	  ................
+
+2009-03-30 13:57 +0000 [r184912]  Russell Bryant <russell at digium.com>
+
+	* channels/h323/Makefile.in, /: Merged revisions 184910 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30
+	  Mar 2009) | 4 lines Fix build error when chan_h323 is not being
+	  built. (reported by cai1982 in #asterisk-dev) ........
+
+2009-03-29 05:52 +0000 [r184840-184845]  Russell Bryant <russell at digium.com>
+
+	* apps/app_followme.c, /: Merged revisions 184843 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009)
+	  | 13 lines Merged revisions 184842 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009)
+	  | 5 lines Ensure targs variable is fully initialized. (closes
+	  issue #14758) Reported by: tim_ringenbach ........
+	  ................
+
+	* channels/Makefile, /: Merged revisions 184838 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 |
+	  russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines
+	  Simplify chan_h323 build to not require a second run of "make".
+	  (closes issue #14715) Reported by: jthurman Patches:
+	  h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license
+	  614) Tested by: tzafrir, russell ........
+
+2009-03-27 19:17 +0000 [r184765]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_iax2.c, main/timing.c, main/channel.c, /,
+	  include/asterisk/timing.h, include/asterisk/channel.h: Merged
+	  revisions 184762 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184762 |
+	  kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12
+	  lines Improve timing interface to remember which provider
+	  provided a timer The ability to load/unload timing interfaces is
+	  nice, but it means that when a timer is allocated, it may come
+	  from provider A, but later provider B becomes the 'preferred'
+	  provider. If this happens, all timer API calls on the timer that
+	  was provided by provider A will actually be handed to provider B,
+	  which will say WTF and return an error. This patch changes the
+	  timer API to include a pointer to the provider of the timer
+	  handle so that future operations on the timer will be forwarded
+	  to the proper provider. (closes issue #14697) Reported by: moy
+	  Review: http://reviewboard.digium.com/r/211/ ........
+
+2009-03-27 18:09 +0000 [r184728]  Russell Bryant <russell at digium.com>
+
+	* main/manager.c, /, apps/app_minivm.c: Merged revisions 184726 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27
+	  Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure
+	  we use the best RNG available. ........
+
+2009-03-27 15:54 +0000 [r184675]  Joshua Colp <jcolp at digium.com>
+
+	* /, res/res_agi.c: Merged revisions 184673 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 |
+	  file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix
+	  speech structure leak in the AGI speech recognition integration.
+	  The AGI dialplan applications did not destroy the speech
+	  structure automatically if it was not destroyed by the running
+	  AGI script. They will now do this. (issue LUMENVOX-15) ........
+
+2009-03-27 14:04 +0000 [r184631]  Russell Bryant <russell at digium.com>
+
+	* main/asterisk.c, include/asterisk/utils.h, main/pbx.c, /,
+	  res/ais/evt.c, main/event.c, pbx/pbx_dundi.c: Merged revisions
+	  184630 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 |
+	  russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines
+	  Change g_eid to ast_eid_default. ........
+
+2009-03-27 13:22 +0000 [r184587]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 184566 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) |
+	  16 lines Merged revisions 184565 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9
+	  lines Fix an issue where nat=yes would not always take effect for
+	  the RTP session on outgoing calls. If calls were placed using an
+	  IP address or hostname the global nat setting was copied over but
+	  was not set on the RTP session itself. This caused the RTP stack
+	  to not perform symmetric RTP actions. (closes issue #14546)
+	  Reported by: acunningham ........ ................
+
+2009-03-27 02:25 +0000 [r184513-184547]  Russell Bryant <russell at digium.com>
+
+	* /, include/asterisk/lock.h: Merged revisions 184531 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r184531 | russell | 2009-03-26 21:20:23 -0500 (Thu, 26 Mar 2009)
+	  | 20 lines Fix some issues with rwlock corruption that caused
+	  deadlock like symptoms. When dvossel and I were doing some load
+	  testing last week, we noticed that we could make Asterisk trunk
+	  lock up instantly when we started generating a bunch of calls.
+	  The backtraces of locked threads were bizarre, and many were
+	  stuck on an _unlock_ of an rwlock. The changes are: 1) Fix a
+	  number of places where a backtrace would be loaded into an
+	  invalid index of the backtrace array. It's an off by one error,
+	  which ends up writing over the rwlock itself. 2) Ensure that in
+	  the array of held locks, we NULL out an index once it is not
+	  being used so that it's not confusing when analyzing its
+	  contents. 3) Remove a bunch of logging referring to an rwlock
+	  operating being done with "deep reentrancy". It is normal for
+	  _many_ threads to hold a read lock on an rwlock. ........
+
+	* /, main/file.c: Merged revisions 184515 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 |
+	  russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines
+	  Don't act surprised if we get a -1 indication. ........
+
+	* include/asterisk/heap.h, /, main/heap.c: Merged revisions 184512
+	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r184512 | russell | 2009-03-26 20:35:56 -0500 (Thu, 26
+	  Mar 2009) | 2 lines Pass more useful information through to lock
+	  tracking when DEBUG_THREADS is on. ........
+
+2009-03-26 22:19 +0000 [r184451]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* sounds/Makefile, /: Merged revisions 184448 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar
+	  2009) | 9 lines Merged revisions 184447 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar
+	  2009) | 3 lines use new, improved 8kHz prompts ........
+	  ................
+
+2009-03-26 21:18 +0000 [r184394]  David Vossel <dvossel at digium.com>
+
+	* /, apps/app_test.c: Merged revisions 184389 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r184389 | dvossel | 2009-03-26 16:09:37 -0500 (Thu, 26 Mar 2009)
+	  | 14 lines Merged revisions 184388 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009)
+	  | 8 lines pri loop TestClient/TestServer fails: server SEND DTMF
+	  8 app_test was failing when sending the last DTMF digit, 8,
+	  because of the 100ms pause issued after DTMF is sent. During this
+	  pause the other side would hang up causing the test to look like
+	  it failed. Now the other side waits a second before hanging up.
+	  (closes issue #12442) Reported by: tzafrir ........
+	  ................
+
+2009-03-25 22:13 +0000 [r184325-184345]  Russell Bryant <russell at digium.com>
+
+	* /, main/event.c: Merged revisions 184344 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184344 |
+	  russell | 2009-03-25 17:11:35 -0500 (Wed, 25 Mar 2009) | 2 lines
+	  Remove unneeded AST_LIST_ENTRY() and comment on the purpose of
+	  ast_event_ref. ........
+
+	* channels/chan_iax2.c, channels/chan_dahdi.c,
+	  include/asterisk/event.h, channels/chan_skinny.c, res/ais/evt.c,
+	  main/event.c, include/asterisk/strings.h, main/asterisk.c,
+	  channels/chan_mgcp.c, apps/app_voicemail.c,
+	  channels/chan_unistim.c, include/asterisk/devicestate.h, /,
+	  channels/chan_sip.c, main/devicestate.c,
+	  include/asterisk/_private.h: Merged revisions 184339 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009)
+	  | 35 lines Improve performance of the ast_event cache
+	  functionality. This code comes from
+	  svn/asterisk/team/russell/event_performance/. Here is a summary
+	  of the changes that have been made, in order of both invasiveness
+	  and performance impact, from smallest to largest. 1) Asterisk
+	  1.6.1 introduces some additional logic to be able to handle
+	  distributed device state. This functionality comes at a cost. One
+	  relatively minor change in this patch is that the extra
+	  processing required for distributed device state is now
+	  completely bypassed if it's not needed. 2) One of the things that
+	  I noticed when profiling this code was that a _lot_ of time was
+	  spent doing string comparisons. I changed the way strings are
+	  represented in an event to include a hash value at the front. So,
+	  before doing a string comparison, we do an integer comparison on
+	  the hash. 3) Finally, the code that handles the event cache has
+	  been re-written. I tried to do this in a such a way that it had
+	  minimal impact on the API. I did have to change one API call,
+	  though - ast_event_queue_and_cache(). However, the way it works
+	  now is nicer, IMO. Each type of event that can be cached (MWI,
+	  device state) has its own hash table and rules for hashing and
+	  comparing objects. This by far made the biggest impact on
+	  performance. For additional details regarding this code and how
+	  it was tested, please see the review request. (closes issue
+	  #14738) Reported by: russell Review:
+	  http://reviewboard.digium.com/r/205/ ........
+
+	* /: add reviewboard:url property.
+
+2009-03-25 19:26 +0000 [r184282]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 184280 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 |
+	  file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix
+	  issue with a T38 reinvite being sent even if not configured to do
+	  so. If we receive a T38 request negotiate control frame we should
+	  only attempt to do so if the option is enabled on the dialog.
+	  ........
+
+2009-03-25 15:12 +0000 [r184223]  Eliel C. Sardanons <eliels at gmail.com>
+
+	* main/asterisk.c, /: Merged revisions 184220 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) |
+	  19 lines Merged revisions 184188 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) |
+	  13 lines Avoid destroying the CLI line when moving the cursor
+	  backward and trying to autocomplete. When moving the cursor
+	  backward and pressing TAB to autocomplete, a NULL is put in the
+	  line and we are loosing what we have already wrote after the
+	  actual cursor position. (closes issue #14373) Reported by: eliel
+	  Patches: asterisk.c.patch uploaded by eliel (license 64) Tested
+	  by: lmadsen ........ ................
+
+2009-03-25 01:55 +0000 [r184149]  Russell Bryant <russell at digium.com>
+
+	* main/timing.c, utils/Makefile, /, include/asterisk/compat.h:
+	  Merged revisions 184147 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 |
+	  russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines
+	  Fix build issues on Mac OSX. (closes issue #14714) Reported by:
+	  ygor ........
+
+2009-03-24 22:42 +0000 [r184081]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_senddtmf.c, /: Merged revisions 184079 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar
+	  2009) | 15 lines Merged revisions 184078 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar
+	  2009) | 9 lines Change NULL pointer check to be ast_strlen_zero.
+	  The 'digit' variable is guaranteed to be non-NULL, so the if
+	  statement could never evaluate true. Changing to ast_strlen_zero
+	  makes the logic correct. This was found while reviewing
+	  ast_channel_ao2 code review. ........ ................
+
+2009-03-24 21:47 +0000 [r184039]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 184037 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009)
+	  | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low
+	  and =medium The default codec configuration for chan_iax2 is
+	  bandwidth=low. I noticed slin16 being negotiated as the codec in
+	  some test calls, but that no longer happens after this change.
+	  ........
+
+2009-03-24 15:28 +0000 [r183867-183916]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, configs/voicemail.conf.sample: Merged revisions 183914 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500
+	  (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009)
+	  | 3 lines Additionally note that the operator option needs an 'o'
+	  extension. (Related to issue #14731) ........ ................
+
+	* /, main/http.c: Merged revisions 183865 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 |
+	  tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines
+	  Allow browsers to cache images and other static content. (This is
+	  a regression over 1.4) ........
+
+2009-03-23 18:59 +0000 [r183768]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_monitor.c, /: Merged revisions 183766 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar
+	  2009) | 13 lines Merged revisions 183700 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar
+	  2009) | 7 lines Fix a memory leak in res_monitor.c The only way
+	  that this leak would occur is if Monitor were started using the
+	  Manager interface and no File: header were given. Discovered
+	  while reviewing the ast_channel_ao2 review request. ........
+	  ................
+
+2009-03-23 18:12 +0000 [r183703]  Leif Madsen <lmadsen at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 183701 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009)
+	  | 7 lines Fixes a documentation error introduced during the CLI
+	  cleanup at AstriDevCon 2008. (closes issue #14655) Reported by:
+	  ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728)
+	  Tested by: lmadsen ........
+
+2009-03-20 17:08 +0000 [r183563]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 183560 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r183560 | russell | 2009-03-20 12:00:58 -0500
+	  (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009)
+	  | 2 lines Fix a crash in IAX2 registration handling found during
+	  load testing with dvossel. ........ ................
+
+2009-03-19 20:33 +0000 [r183438]  David Vossel <dvossel at digium.com>
+
+	* include/asterisk/features.h, apps/app_dial.c, /, main/features.c:
+	  Merged revisions 183436 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009)
+	  | 13 lines Merged revisions 183386 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009)
+	  | 6 lines Cleaning up a few things in detect disconnect patch
+	  Initialized ast_call_feature in detect_disconnect to avoid
+	  accessing uninitialized memory. Cleaned up /param tags in
+	  features.h. No longer send dynamic features in
+	  ast_feature_detect. issue #11583 ........ ................
+
+2009-03-19 19:19 +0000 [r183333]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 183321 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500
+	  (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via
+	  svnmerge from

[... 55221 lines stripped ...]



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