[asterisk-commits] bebuild: tag 11.0.0-beta2 r373257 - /tags/11.0.0-beta2/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Sep 20 14:21:31 CDT 2012


Author: bebuild
Date: Thu Sep 20 14:21:27 2012
New Revision: 373257

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=373257
Log:
Importing files for 11.0.0-beta2 release.

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+2012-09-20  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 11.0.0-beta2 Released.
+
+2012-09-20 18:59 +0000 [r373235-373240]  Matthew Jordan <mjordan at digium.com>
+
+	* configs/extensions.conf.sample, CHANGES, apps/app_queue.c:
+	  app_queue: Support an 'agent available' hint Sets INUSE when no
+	  free agents, NOT_INUSE when an agent is free. modifes
+	  handle_statechange() scan members loop to scan for a free agent
+	  and updates the Queue:queuename_avial devstate. Previously exited
+	  early if the member was found in the queue. Now Exits later when
+	  both a member was found, and a free agent was found. alecdavis
+	  (license 585) Reported by: Alec Davis Tested by: alecdavis
+	  Review: https://reviewboard.asterisk.org/r/2121/ ~~~~ Support all
+	  ways a member can be available for 'agent available' hints Alec's
+	  patch in r373188 added the ability to subscribe to a hint for
+	  when Queue members are available. This patch modifies the check
+	  that determines when a Queue member is available by refactoring
+	  the availability checks in num_available_members into a shared
+	  function is_member_available. This should now handle the
+	  ringinuse option, as well as device state values other than
+	  AST_DEVICE_NOT_INUSE.
+
+	* res/res_rtp_asterisk.c, /: When processing RFC 2833 DTMF,
+	  accomodate increasing timestamps in End events While endpoints
+	  should not be changing the source timestamp between DTMF event
+	  packets, the fact is there exists those endpoints that do exactly
+	  that. To work around this, we absorb timestamps within the
+	  expected re-transmit period. Note that this period only affects
+	  End of Event packets, so it should not prevent the detection of
+	  new DTMF digits that happen to arrive right on top of each other.
+	  (closes issue ASTERISK-20424) Reported by: Vladimir Mikhelson
+	  Tested by: mjordan, Vladimir Mikhelson Review:
+	  https://reviewboard.asterisk.org/r/2124 ........ Merged revisions
+	  373236 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ........ Merged revisions 373237 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* configs/extensions.conf.sample, CHANGES, apps/app_queue.c: Add
+	  queue monitoring hints This patch adds support for hints on a
+	  queue. Hints can be added using the nomenclature 'Queue:name',
+	  where name is the name of the queue being monitored. This nifty
+	  feature was done by Alec Davis. Review:
+	  https://reviewboard.asterisk.org/r/1619 Reported by: Alec Davis
+	  Tested by: alecdavis patches: review1619.diff2 by alecdavis
+	  (license 585)
+
+2012-09-20 18:18 +0000 [r373229]  Joshua Colp <jcolp at digium.com>
+
+	* channels/sip/include/sip.h, res/res_rtp_asterisk.c,
+	  main/rtp_engine.c, channels/chan_sip.c, configure,
+	  include/asterisk/autoconfig.h.in, configure.ac,
+	  configs/sip.conf.sample, include/asterisk/rtp_engine.h: Add
+	  support for DTLS-SRTP to res_rtp_asterisk and chan_sip. As
+	  mentioned on the review for this, WebRTC has moved towards
+	  choosing DTLS-SRTP as the mechanism for key exchange for SRTP.
+	  This commit adds support for this but makes it available for
+	  normal SIP clients as well. Testing has been done to ensure that
+	  this introduces no regressions with existing behavior and also
+	  that it functions as expected. Review:
+	  https://reviewboard.asterisk.org/r/2113/
+
+2012-09-20 17:15 +0000 [r373220]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/features.h, main/channel.c,
+	  apps/app_directed_pickup.c, funcs/func_channel.c,
+	  main/features.c, include/asterisk/channel.h: Named call pickup
+	  groups. Fixes, missing functionality, and improvements. *
+	  ASTERISK-20383 Missing named call pickup group features:
+	  CHANNEL(callgroup) - Need CHANNEL(namedcallgroup)
+	  CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup) Pickup() -
+	  Needs to also select from named pickup groups. * ASTERISK-20384
+	  Using the pickupexten, the pickup channel selection could fail
+	  even though there was a call it could have picked up. In a call
+	  pickup race when there are multiple calls to pickup and two
+	  extensions try to pickup a call, it is conceivable that the loser
+	  will not pick up any call even though it could have picked up the
+	  next oldest matching call. Regression because of the named call
+	  pickup group feature. * See ASTERISK-20386 for the implementation
+	  improvements. These are the changes in channel.c and channel.h. *
+	  Fixed some locking issues in CHANNEL(). (closes issue
+	  ASTERISK-20383) Reported by: rmudgett (closes issue
+	  ASTERISK-20384) Reported by: rmudgett (closes issue
+	  ASTERISK-20386) Reported by: rmudgett Tested by: rmudgett Review:
+	  https://reviewboard.asterisk.org/r/2112/
+
+2012-09-20 13:00 +0000 [r373211]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/chan_sip.c: Correct handling of unknown SDP stream types
+	  When the patch to handle arbitrary SDP stream arrangements went
+	  into Asterisk, it also included an ability to transparently
+	  decline unknown stream types. The scanf calls used were not
+	  checked properly causing this part of the functionality to be
+	  broken. (closes issue ASTERISK-20203)
+
+2012-09-18 20:14 +0000 [r373133]  Sean Bright <sean at malleable.com>
+
+	* main/manager.c, /: Don't crash when passing a NULL message to
+	  __astman_get_header. Before this commit, __astman_get_header
+	  would blindly dereference the passed in 'struct message *' to
+	  traverse the header list. There are cases, however, such as
+	  '*CLI> sip qualify peer foo' where the message pointer is NULL,
+	  so we need to check for that. ........ Merged revisions 373131
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  Merged revisions 373132 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-18 15:47 +0000 [r373119]  dlee <dlee at localhost>:
+
+	* Makefile, include/asterisk/utils.h, configure,
+	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
+	  -fnested-functions compile flag, if needed. In order to use
+	  nested functions on some versions of GCC (e.g. GCC on OS X), the
+	  -fnested-functions flag must be passed to the compiler. This
+	  patch adds detection logic to ./configure to add the flag if
+	  necessary. It also adds a comment to utils.h as to why the nested
+	  function needs a prototype. (closes issue ASTERISK-20399)
+	  Reported by: David M. Lee Review:
+	  https://reviewboard.asterisk.org/r/2102/
+
+2012-09-15 00:27 +0000 [r373107]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_ss7.c, /: Made companding law for SS7 calls only
+	  determined by SS7 signaling type. For SS7, the companding law for
+	  a call was chosen inconsistently depending upon ss7type (ITU vs
+	  ANSI) and the DAHDI companding default (T1 vs E1). For incoming
+	  calls, the companding law was determined by ss7type. For outgoing
+	  calls, the companding law was determined by the DAHDI default.
+	  With the wrong combination you would get A-law/u-law conflicts.
+	  An A-law/u-law conflict sounds like bad static on the line. SS7
+	  ITU signaling with E1 line: ok SS7 ITU signaling with T1 line:
+	  noise SS7 ANSI signaling with E1 line: noise SS7 ANSI signaling
+	  with T1 line: ok * Fix the companding law used to be determined
+	  by the SS7 signaling type only. ........ Merged revisions 373090
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  Merged revisions 373101 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-14 19:50 +0000 [r373079]  Matthew Jordan <mjordan at digium.com>
+
+	* main/tcptls.c, /, channels/chan_sip.c, main/libasteriskssl.c:
+	  Resolve memory leaks in TLS initialization and TLS client
+	  connections This patch resolves two sources of memory leaks when
+	  using TLS in Asterisk: 1) It removes improper initialization (and
+	  multiple re-initializations) of portions of the SSL library.
+	  Asterisk calls SSL_library_init and SSL_load_error_strings during
+	  SSL initialization; collectively this obviates the need for
+	  calling any of the following during initialization or client
+	  connection handling: * ERR_load_crypto_strings (handled by
+	  SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for
+	  SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for
+	  SSL_library_init) 2) Failure to completely clean up all memory
+	  allocated by Asterisk and by the SSL library for TLS clients.
+	  This included not freeing the SSL_CTX object in the SIP channel
+	  driver, as well as not clearing the error stack when the TLS
+	  client exited. Note that these memory leaks were found by Thomas
+	  Arimont, and this patch was essentially written by him with some
+	  minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont
+	  Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas
+	  Arimont (license 5525) Review:
+	  https://reviewboard.asterisk.org/r/2105 ........ Merged revisions
+	  373061 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ........ Merged revisions 373062 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-13 20:04 +0000 [r373029-373047]  dlee <dlee at localhost>:
+
+	* main/Makefile: Fixed make clean when configured
+	  --disable-asteriskssl
+
+	* main/channel.c, /, include/asterisk/channel.h: Fix timeouts for
+	  ast_waitfordigit[_full]. ast_waitfordigit_full would simply pass
+	  its timeout to ast_waitfor_nandfds, expecting it to decrement the
+	  timeout by however many milliseconds were waited. This is a
+	  problem if it consistently waits less than 1ms. The timeout will
+	  never be decremented, and we wait... FOREVER! This patch makes
+	  ast_waitfordigit_full manage the timeout itself. It maintains the
+	  previously undocumented behavior that negative timeouts wait
+	  forever. (closes issue ASTERISK-20375) Reported by: Mark
+	  Michelson Tested by: Mark Michelson Review:
+	  https://reviewboard.asterisk.org/r/2109/ ........ Merged
+	  revisions 373024 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 373025 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-12 20:53 +0000 [r372995]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_motif.c: Skip any non-content information when
+	  looking for and handling content. This fixes a bug with Jitsi and
+	  conference calling. Jitsi implements XEP-0298 which places some
+	  conference-info information in the session-initiate request which
+	  chan_motif did not expect to occur.
+
+2012-09-12 18:23 +0000 [r372984]  Jonathan Rose <jrose at digium.com>
+
+	* res/res_xmpp.c: res_xmpp: Fix a segfault caused by bodyless
+	  messages (closes issue ASTERISK-20361) Reported by: Noah
+	  Engelberth Review: https://reviewboard.asterisk.org/r/2108/
+
+2012-09-12 15:19 +0000 [r372937]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Add channel name to a warning to make
+	  debugging easier. The "autodestruct with owner in place" message
+	  is typically indicative of a channel reference leak. Printing out
+	  the name of the channel in the message may be helpful when trying
+	  to debug the issue. ........ Merged revisions 372932 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372933 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-12 14:18 +0000 [r372930]  dlee <dlee at localhost>:
+
+	* main/Makefile: Fixed r372696 when configured
+	  --disable-asteriskssl; properly install libasteriskssl.dylib on
+	  OS X. I didn't realize that libasteriskssl.c was still compiled,
+	  even when you disable asteriskssl; it simple gets statically
+	  linked into asterisk.
+
+2012-09-11 22:32 +0000 [r372917]  Jonathan Rose <jrose at digium.com>
+
+	* channels/chan_local.c, /: chan_local: Switch from using a random
+	  4 digit hex identifier to unique id Changes chan_local channels
+	  to use an 8 digit hex identifier generated atomically and
+	  sequentially in order to eliminate the chance of having multiple
+	  channels with the same name during high call volume situations.
+	  (issue ASTERISK-20318) Reported by: Dan Cropp Review:
+	  https://reviewboard.asterisk.org/r/2104/ ........ Merged
+	  revisions 372902 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372916 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-11 21:15 +0000 [r372886-372888]  Mark Michelson <mmichelson at digium.com>
+
+	* main/asterisk.c, /, include/asterisk/_private.h, main/message.c:
+	  Fix inability to shutdown gracefully due to an unending channel
+	  reference. message.c makes use of a special message queue channel
+	  that exists in thread storage. This channel never goes away due
+	  to the fact that the taskprocessor used by message.c does not get
+	  shut down, meaning that it never ends the thread that stores the
+	  channel. This patch fixes the problem by shutting down the
+	  taskprocessor when Asterisk is shut down. In addition, the thread
+	  storage has a destructor that will release the channel reference
+	  when the taskprocessor is destroyed. (closes issue AST-937)
+	  Reported by Jason Parker Patches: AST-937.patch uploaded by Mark
+	  Michelson (License #5049) Tested by Jason Parker ........ Merged
+	  revisions 372885 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* /, main/features.c: Fix bad channel application data reference.
+	  When channels get bridged due to an AMI bridge action or a DTMF
+	  attended transfer, the two channels that get bridged have their
+	  application data pointing to the other channel's name. This means
+	  that if one channel is hung up but the other moves on, it means
+	  that the channel that moves on will have its application data
+	  pointing at freed memory. (issue ASTERISK-20335) Reported by:
+	  aragon ........ Merged revisions 372840 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372841 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-11 17:16 +0000 [r372864]  dlee <dlee at localhost>:
+
+	* Makefile, /: Corrects the astsbindir setting when installing the
+	  sample asterisk.conf. (closes issue ASTERISK-20406) ........
+	  Merged revisions 372863 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-10 20:59 +0000 [r372795-372806]  Kinsey Moore <kmoore at digium.com>
+
+	* /, channels/chan_iax2.c: Ensure iax2 debug output is displayed
+	  when expected When IAX2 debug was changed from iax_showframe to
+	  iax_outputframe, some instances were missed (or added afterward).
+	  This was causing debug output to not be displayed when expected.
+	  (closes issue ASTERISK-20338) Reported-by: John Covert Patch-by:
+	  John Covert ........ Merged revisions 372804 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372805 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* channels/chan_jingle.c, include/asterisk/doxygen/architecture.h,
+	  main/devicestate.c, channels/chan_gtalk.c, res/res_jabber.c:
+	  Deprecate chan_gtalk, chan_jingle, and res_jabber chan_gtalk,
+	  chan_jingle, and res_jabber are now deprecated in favor of using
+	  chan_motif and res_xmpp. They are a feature-equivalent
+	  replacement and are written to be more easily maintainable.
+	  (closes issue ASTERISK-20298) Review:
+	  https://reviewboard.asterisk.org/r/2082/ Reported-by: Leif Madsen
+
+2012-09-10 19:19 +0000 [r372777]  dlee <dlee at localhost>:
+
+	* res/res_rtp_asterisk.c: res_rtp_asterisk: Eliminate "type-punned
+	  pointer" build warning. Removes "res_rtp_asterisk.c:706: warning:
+	  dereferencing type-punned pointer will break strict-aliasing
+	  rules" warning from the build on 32-bit platforms. The problem is
+	  that 'size' was referenced aliased to both (pj_size_t *) and
+	  (pj_ssize_t *). Now just make a copy of size that is the right
+	  type so there isn't any pointer aliasing happening. It also adds
+	  comments and asserts regarding what looks like an inappropriate
+	  use of pj_sock_sendto, but is actually totally fine. (closes
+	  issue ASTERISK-20368) Reported by: Shaun Ruffel Tested by:
+	  Michael L. Young Patches:
+	  0001-res_rtp_asterisk-Eliminate-type-punned-pointer-build.patch
+	  uploaded by Shaun Ruffel (license 5417) slightly modified by
+	  David M. Lee.
+
+2012-09-10 18:50 +0000 [r372768]  Jonathan Rose <jrose at digium.com>
+
+	* /, apps/app_meetme.c: app_meetme: Document that 'p' option will
+	  continue in dialplan. (closes issue AST-991) Reported by John
+	  Bigelow ........ Merged revisions 372765 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372767 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-10 18:37 +0000 [r372766]  Kinsey Moore <kmoore at digium.com>
+
+	* /: Recorded merge of revisions 372764 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10 ........ Warn on
+	  CLI when UDPTL init fails This adds a CLI warning when a SDP
+	  offer is rejected due to UDPTL initialization failure.
+	  Previously, there was no indication of the reason for offer
+	  rejection in this case. (closes issue ASTERISK-20357)
+	  Reported-by: Francesco Usseglio Gaudi ........ Merged revisions
+	  372763 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-10 17:33 +0000 [r372754]  Jonathan Rose <jrose at digium.com>
+
+	* main/channel.c, /: Masquerade: Retain parkinglot settings made by
+	  CHANNEL function. Prior to this patch, the user would have a
+	  parkinglot set on a channel that was parked and when the channel
+	  was retrieved, any attempt by that channel to park would simply
+	  use the default. This patch makes parkinglot values set in this
+	  way be retained through the masquerade. (closes issue AST-990)
+	  Reported by: Nick Huskinson Patches:
+	  masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose
+	  (license 6182) ........ Merged revisions 372736 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372737 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-09 01:25 +0000 [r372711]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/sip/sdp_crypto.c, /: Only re-create an SRTP session when
+	  needed In r356604, SRTP handling was fixed to accomodate multiple
+	  crypto keys in an SDP offer and the ability to re-create an SRTP
+	  session when the crypto keys changed. In certain circumstances -
+	  most notably when a phone is put on hold after having been
+	  bridged for a significant amount of time - the act of re-creating
+	  the SRTP session causes problems for certain models of phones.
+	  The patch committed in r356604 always re-created the SRTP session
+	  regardless of whether or not the cryptographic keys changed.
+	  Since this is technically not necessary, this patch modifies the
+	  behavior to only re-create the SRTP session if Asterisk detects
+	  that the remote key has changed. This allows models of phones
+	  that do not handle the SRTP session changing to continue to work,
+	  while also providing the behavior needed for those phones that do
+	  re-negotiate cryptographic keys. (issue ASTERISK-20194) Reported
+	  by: Nicolo Mazzon Tested by: Nicolo Mazzon Review:
+	  https://reviewboard.asterisk.org/r/2099 ........ Merged revisions
+	  372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ........ Merged revisions 372710 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-08 05:51 +0000 [r372696]  dlee <dlee at localhost>:
+
+	* /, main/Makefile: Recorded merge of revisions 372695 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10 ........ Add
+	  OPENSSL_INCLUDE to the CFLAGS for ssl.c and tcptls.c. Without
+	  this flag, those files will compile with the system installed
+	  OpenSSL headers (if they exist). This is a real bummer if a
+	  different path was specified using --with-ssl= (closes issue
+	  ASTERISK-20392) ........ Merged revisions 372682 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-07 23:07 +0000 [r372622-372657]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, main/astmm.c: Fix MALLOC_DEBUG version of ast_strndup().
+	  (closes issue ASTERISK-20349) Reported by: Brent Eagles ........
+	  Merged revisions 372655 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372656 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* /, funcs/func_math.c: Remove annoying unconditional debug message
+	  from INC/DEC functions. (closes issue AST-1001) Reported by:
+	  Guenther Kelleter ........ Merged revisions 372628 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372629 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* /, apps/app_queue.c: Fix exception path typo in app_queue.c
+	  try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy
+	  Pepper Patches: fix-local-channel-locking.patch (license #6350)
+	  patch uploaded by Jeremy Pepper ........ Merged revisions 372624
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  Merged revisions 372625 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* apps/app_voicemail.c, /: Fix VoicemailUserEntry event headers
+	  ServerEmail and MailCommand reported values. The AMI action
+	  VoicemailUsersList VoicemailUserEntry event headers ServerEmail
+	  and MailCommand did not report the global values if they were not
+	  overridden. The VoicemailUserEntry event header ServerEmail was
+	  not populated with the global value if the voicemail user did not
+	  override it. The VoicemailUserEntry event header MailCommand was
+	  never populated with a value. * Removed unused struct ast_vm_user
+	  member mailcmd[]. (closes issue AST-973) Reported by: John
+	  Bigelow Tested by: rmudgett ........ Merged revisions 372620 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372621 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-07 21:04 +0000 [r372609-372611]  dlee <dlee at localhost>:
+
+	* res/pjproject/pjlib-util/lib, res/pjproject/pjmedia/bin,
+	  res/pjproject/third_party/bin, res/pjproject/third_party/gsm/lib,
+	  res/pjproject/lib, res/pjproject/pjlib/lib,
+	  res/pjproject/third_party/gsm/bin, res/pjproject/pjnath/lib,
+	  res/pjproject/pjsip/lib, res/pjproject/pjsip-apps/lib,
+	  res/pjproject/pjsip/bin, res/pjproject/pjsip-apps/bin,
+	  res/pjproject/pjmedia/lib, res/pjproject/third_party/lib,
+	  codecs/ilbc: svn:ignore cleanup. * pjproject bin and lib
+	  directories should pretty much ignore everything * Ignore *.o in
+	  codecs/ilbc
+
+	* res/Makefile: Fix parallel make for res_asterisk_rtp. Fixes a
+	  build regression introduced in r369517 "Add support for
+	  ICE/STUN/TURN in res_rtp_asterisk and chan_sip." [1]. [1]
+	  http://svnview.digium.com/svn/asterisk?view=revision&revision=369517
+	  When compiling asterisk in parallel like: $ make -j 10 It's
+	  possible to get errors like the following:
+	  .pjlib-util-test-x86_64-unknown-linux-gnu.depend:120: *** missing
+	  separator. Stop. make[4]: *** [depend] Error 2 make[3]: *** [dep]
+	  Error 1 make[2]: ***
+	  [/home/sruffell/asterisk-working/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a]
+	  Error 2 make[3]: warning: jobserver unavailable: using -j1. Add
+	  `+' to parent make rule. This is because the build system is
+	  trying to build each of the libraries in pjproject in parallel.
+	  Now the build will build pjproject in a single job and link the
+	  results into res_asterisk_rtp. Parallel builds, on one test
+	  system, saves ~1.5 minutes from a default Asterisk build: Single
+	  job: $ git clean -fdx >/dev/null && time ( ./configure >/dev/null
+	  2>&1 && make >/dev/null 2>&1 ) real 2m34.529s user 1m41.810s sys
+	  0m15.970s Parallel make: $ git clean -fdx >/dev/null && time (
+	  ./configure >/dev/null 2>&1 && make -j10 >/dev/null 2>&1 ) real
+	  1m2.353s user 2m39.120s sys 0m18.850s (closes issue
+	  ASTERISK-20362) Reported by: Shaun Ruffel Patches:
+	  0001-res_asterisk_rtp-Fix-build-error-when-using-parallel.patch
+	  uploaded by Shaun Ruffel (License #5417)
+
+2012-09-07 02:26 +0000 [r372531-372583]  Matthew Jordan <mjordan at digium.com>
+
+	* /, apps/app_minivm.c: Free ast_str objects when temp file fails
+	  to be created in MiniVM The previous commit (r372554) was from a
+	  patch that was written before r366880, which ensured that ast_str
+	  objects allocated in the sendmail routine were free'd in off
+	  nominal paths. This commit frees the string objects in the off
+	  nominal path introduced in r372554. (issue ASTERISK-17133)
+	  Reported by: Tzafrir Cohen ........ Merged revisions 372581 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372582 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* /, apps/app_minivm.c: Fix file descriptor leak and pointer scope
+	  issue in MiniVM when sending mail When MiniVM sends an e-mail and
+	  it has the volgain option set, it will spawn sox in a separate
+	  process to handle the manipulation of the sound file. In doing
+	  so, it creates a temporary file. There are two problems here: 1)
+	  The file descriptor returned from mkstemp is leaked 2) The
+	  finalfilename character pointer points to a buffer that loses
+	  scope once volgain processing is finished. Note that in r316265,
+	  Russell fixed some gcc warnings by using the return value of the
+	  mkstemp call. A warning was placed in minivm that the file
+	  descriptor was going to be leaked. This patch reverts that
+	  change, as it handles the leak and 'uses' the file descriptor
+	  returned from mkstemp. (closes issue ASTERISK-17133) Reported by:
+	  Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir
+	  Cohen (license #5035) ........ Merged revisions 372554 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372555 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* apps/app_queue.c: Update QueueMemberStatus event documentation to
+	  include member status values The Status: header in a
+	  QueueMemberStatus event (and other QueueMember* events) is the
+	  numeric value of the device state corresponding to that Queue
+	  Member. As those values are not exactly obvious, listing them in
+	  the documentation is useful. Matt Riddell reported this
+	  indirectly through the wiki page. (closes issue ASTERISK-20243)
+	  Reported by: Matt Riddell
+
+2012-09-06 22:12 +0000 [r372523]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, channels/sig_pri.c: Fix loss of MOH on an ISDN channel when
+	  parking a call for the second time. Using the AMI redirect action
+	  to take an ISDN call out of a parking lot causes the MOH state to
+	  get confused. The redirect action does not take the call off of
+	  hold. When the call is subsequently parked again, the call no
+	  longer hears MOH. * Make chan_dahdi/sig_pri restart MOH on
+	  repeated AST_CONTROL_HOLD frames if it is already in a state
+	  where it is supposed to be sending MOH. The MOH may have been
+	  stopped by other means. (Such as killing the generator.) This
+	  simple fix is done rather than making the AMI redirect action
+	  post an AST_CONTROL_UNHOLD unconditionally when it redirects a
+	  channel and thus potentially breaking something with an
+	  unexpected AST_CONTROL_UNHOLD. (closes issue ABE-2873) Patches:
+	  jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by
+	  rmudgett ........ Merged revisions 372521 from
+	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+	  ........ Merged revisions 372522 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-06 21:42 +0000 [r372519]  Kinsey Moore <kmoore at digium.com>
+
+	* /, apps/app_queue.c: Ensure listed queues are not offered for
+	  completion When using tab-completion for the list of queues on
+	  "queue reset stats" or "queue reload
+	  {all|members|parameters|rules}", the tab-completion listing for
+	  further queues erroneously listed queues that had already been
+	  added to the list. The tab-completion listing now only displays
+	  queues that are not already in the list. (closes issue AST-963)
+	  Reported-by: John Bigelow ........ Merged revisions 372517 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372518 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-06 18:55 +0000 [r372500]  dsessions <dsessions at localhost>:
+
+	* channels/chan_sip.c, configs/res_ldap.conf.sample: LDAP Realtime
+	  Peers Cannot Register Prior to 1.8, it was not necessary for an
+	  explicit "type" to be set for an asterisk LDAP realtime peer. Now
+	  the routine find_peer actually checks the type field during
+	  registration and fails to find the peer if it is not set. The
+	  attached patches make the realtime type equal whatever type is
+	  being searched for if the type is 0 upon return from routine
+	  build_peer. (closes issue ASTERISK-17222) Reported by: John
+	  Covert Patch by: David Vossel Tested by: Darren Sessions Review:
+	  https://reviewboard.asterisk.org/r/2095/
+
+2012-09-06 15:56 +0000 [r372473]  Jonathan Rose <jrose at digium.com>
+
+	* /, UPGRADE-1.8.txt: chan_sip: Note change in behavior to how
+	  directmediapermit/deny ACL works r366547 introduced a change to
+	  the directmedia ACL for chan_sip which modified the behavior
+	  significantly. Prior to the patch, this option would bridge peers
+	  with directmedia if a peer's IP address matched its own
+	  directmedia ACL. After that patch, the peer would check the
+	  bridged peer's ACL instead. This change has been present since
+	  1.8.14.0. That patched failed to document the change in
+	  Upgrade.txt, so this patch adds mention of that change to
+	  UPGRADE.txt (UPGRADE-1.8.txt in newer branches) (issue AST-876)
+	  ........ Merged revisions 372471 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372472 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-06 14:30 +0000 [r372446]  Kinsey Moore <kmoore at digium.com>
+
+	* /, apps/app_queue.c: Ensure "rules" is tab-completable for "queue
+	  show" Previously, tabbing at the end of "queue show" produced a
+	  list of available queues about which information could be shown,
+	  but did not include an alternative command, "rules", to access
+	  information about queue rules. The "rules" item should now be
+	  shown in the list of tab-completable items. (closes issue
+	  AST-958) Reported-by: John Bigelow ........ Merged revisions
+	  372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ........ Merged revisions 372445 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-06 02:50 +0000 [r372392-372419]  Matthew Jordan <mjordan at digium.com>
+
+	* /, pbx/pbx_dundi.c: Fix DUNDi message routing bug when
+	  neighboring peer is unreachable Consider a scenario where DUNDi
+	  peer PBX1 has two peers that are its neighbors, PBX2 and PBX3,
+	  and where PBX2 and PBX3 are also neighbors. If the connection is
+	  temporarily broken between PBX1 and PBX3, PBX1 should not include
+	  PBX3 in the list of peers it sends to PBX2 in a DPDISCOVER
+	  message, as it cannot send messages to PBX3. If it does, PBX2
+	  will assume that PBX3 already received the message and fail to
+	  forward the message on to PBX3 itself. This patch fixes this by
+	  only including peers in a DPDISCOVER message that are reachable
+	  by the sending node. This includes all peers with an empty
+	  address (00:00:00:00:00:00) and that are have been reached by a
+	  qualify message. This patch also prevents attempting to qualify a
+	  dynamic peer with an empty address until that peer registers.
+	  (closes issue ASTERISK-19309) Reported by: Peter Racz patches:
+	  dundi_routing.patch uploaded by Peter Racz (license 6290) The
+	  patch uploaded by Peter was modified slightly for this commit.
+	  ........ Merged revisions 372417 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372418 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* /, apps/app_followme.c: Allow configured numbers for FollowMe to
+	  be greater than 90 characters When parsing a 'number' defined in
+	  followme.conf, FollowMe previously parsed the number in the
+	  configuration file into a buffer with a length of 90 characters.
+	  This can artificially limit some parallel dial scenarios. This
+	  patch allows for numbers of any length to be defined in the
+	  configuration file. Note that Clod Patry originally wrote a patch
+	  to fix this problem and received a Ship It! on the JIRA issue.
+	  The patch originally expanded the buffer to 256 characters.
+	  Instead, the patch being committed duplicates the string in the
+	  config file on the stack before parsing it for consumption by the
+	  application. (closes issue ASTERISK-16879) Reported by: Clod
+	  Patry Tested by: mjordan patches: followme_no_limit.diff uploaded
+	  by Clod Patry (license #5138) Slightly modified for this commit.
+	  ........ Merged revisions 372390 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372391 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 19:43 +0000 [r372373]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/dsp.c, /: Fix compile error. ........ Merged revisions
+	  372372 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 19:24 +0000 [r372365]  Kinsey Moore <kmoore at digium.com>
+
+	* main/manager.c, /: Correct documentation for ModuleLoad AMI
+	  action The documentation incorrectly listed 'rtp' as a reloadable
+	  subsystem and left out many other reloadable subsystems. It is
+	  now also documented that subsystems may only be reloaded, not
+	  loaded or unloaded. (closes issue AST-977) Reported-by: John
+	  Bigelow ........ Merged revisions 372354 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372358 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 18:46 +0000 [r372342]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* main/dsp.c, /: dsp.c: in ast_mf_detect_init incorrectly sets
+	  goertzel samples to 160, should be MF_GSIZE Related
+	  https://reviewboard.asterisk.org/r/2097/ ........ Merged
+	  revisions 372339 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372341 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 18:36 +0000 [r372340]  Kinsey Moore <kmoore at digium.com>
+
+	* main/pbx.c, /: Ensure counts generated in
+	  manager_show_dialplan_helper are correct When
+	  manager_show_dialplan_helper was written, the counter increment
+	  for the total number of contexts was placed with the extensions
+	  increment instead of in the enclosing loop. This function should
+	  now generate correct context counts. (closes issue AST-970)
+	  Reported-by: John Bigelow ........ Merged revisions 372337 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 372338 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 17:35 +0000 [r372327-372328]  Richard Mudgett <rmudgett at digium.com>
+
+	* res/res_rtp_asterisk.c: Fix coding guidelines issue with a recent
+	  commit.
+
+	* res/res_rtp_asterisk.c: Fix RTP/RTCP read error message
+	  confusion. The RTP/RTCP read error message can report "fail:
+	  success" when the read failure is because of an ICE failure. *
+	  Changed __rtp_recvfrom() to generate a PJ ICE message when ICE
+	  fails. * Changed RTP/RTCP read error message to indicate an
+	  unspecified error when errno is zero. (closes issue
+	  ASTERISK-20288) Reported by: Joern Krebs Patches:
+	  jira_asterisk_20288_err_msg.patch (license #5621) patch uploaded
+	  by rmudgett (modified)
+
+2012-09-05 16:04 +0000 [r372311]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_rtp_asterisk.c, main/rtp_engine.c,
+	  include/asterisk/rtp_engine.h: Re-fix sending unnegotiated
+	  payloads during a P2P RTP bridge. The previous fix still would
+	  look in the static_RTP_PT table, which is inappropriate since we
+	  specifically want to find a codec that has been negotiated.
+	  (closes issue ASTERISK-20296) reported by NITESH BANSAL Patches:
+	  codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418)
+
+2012-09-05 13:47 +0000 [r372289]  Matthew Jordan <mjordan at digium.com>
+
+	* apps/app_voicemail.c, /: Fix memory leaks in app_voicemail when
+	  using IMAP storage or realtime config This patch fixes two memory
+	  leaks: 1. When find_user is called with NULL as its first
+	  parameter, the voicemail user returned is allocated on the heap.
+	  The inboxcount2 function uses find_user in such a fashion when
+	  counting new messages, and fails to free the resulting voicemail
+	  user object. 2. When populate_defaults is called on a voicemail
+	  user, it wipes whatever flags have been set on the object by
+	  copying over the global flags object. If the VM_ALLOCED flag was
+	  ste on the voicemail user prior to doing so, that flag is
+	  removed. This leaks the voicemail user when free_user is later
+	  called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek
+	  patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277)
+	  Patch slightly modified for this commit. Review:
+	  https://reviewboard.asterisk.org/r/2096 ........ Merged revisions
+	  372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8

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