[asterisk-commits] bebuild: tag 1.8.23.0-rc1 r391247 - /tags/1.8.23.0-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Jun 10 09:39:13 CDT 2013


Author: bebuild
Date: Mon Jun 10 09:39:10 2013
New Revision: 391247

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=391247
Log:
Importing files for 1.8.23.0-rc1 release.

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+2013-06-10  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.23.0-rc1 Released.
+
+2013-06-10 14:15 +0000 [r391215]  Matthew Jordan <mjordan at digium.com>
+
+	* UPGRADE.txt, apps/app_queue.c, configs/queues.conf.sample: Add
+	  announce-to-first-user option for app_queue In r386792, the
+	  ability to play prompts to the first caller in a call queue was
+	  added. While this is arguably a bug fix for those who expect the
+	  first caller to continue receiving prompts while the agent is
+	  dialed, it has the side effect of preventing the first caller
+	  from hearing the agent immediately upon bridging. This may not be
+	  a problem for those who really want this option, but for those
+	  who didn't care whether or not the first caller in queue heard
+	  their position, it was an issue. This patch disables the ability
+	  for the first caller in the queue to hear prompts and adds a new
+	  option, announce-to-first-user, to queues.conf. Those who the
+	  behavior can enable it by setting this value to True. Note that
+	  if we ever implement the ability to have the prompts be stopped
+	  upon bridging, this option can be removed. (closes issue
+	  ASTERISK-21782) Reported by: Remi Quezada
+
+2013-06-10 09:30 +0000 [r391062-391143]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* channels/chan_iax2.c: chan_iax2: nativebridge refactor, missed
+	  unlock bridgecallno
+
+	* channels/chan_iax2.c: fix bad edit after conflict resolution
+
+	* channels/chan_iax2.c: IAX2: refactor nativebridge transfer remove
+	  triple checking of iaxs[fr->callno]->transferring reduce
+	  indentation. Reported by: alecdavis Tested by: alecdavis
+	  alecdavis (license 585) Review
+	  https://reviewboard.asterisk.org/r/2602/
+
+	* channels/chan_iax2.c: IAX2: fix race condition with nativebridge
+	  transfers. 1). When touching the bridgecallno, we need to lock
+	  it. 2). stop_stuff() which calls iax2_destroy_helper() Assumes
+	  the lock on the pvt is already held, when iax2_destroy_helper()
+	  is called. Thus we need to lock the bridgecallno pvt before we
+	  call stop_stuff(iaxs[fr->callno]->bridgecallno); 3). When
+	  evaluating the state of 'callno->transferring' of the current
+	  leg, we can't change it to READY unless the bridgecallno is
+	  locked. Why, if we are interrupted by the other call leg before
+	  'transferring = TRANSFER_RELEASED', the interrupt will find that
+	  it is READY and that the bridgecallno is also READY so Releases
+	  the legs. (closes issue ASTERISK-21409) Reported by: alecdavis
+	  Tested by: alecdavis alecdavis (license 585) Review
+	  https://reviewboard.asterisk.org/r/2594/
+
+2013-05-31 08:10 +0000 [r390181]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/chan_ooh323.c: reject call attempts when gatekeeper is
+	  configured but not registered (closes issue ASTERISK-21800)
+	  Reported by: Dmitry Melekhov Patches: ASTERISK-21800-1.patch
+	  Tested by: Dmitry Melekhov
+
+2013-05-29 20:10 +0000 [r390044]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/channel.c: Fix segfault when dealing with chan_agent
+	  channels. Check the returned bridged pointer for NULL to avoid a
+	  crash. It looks like chan_agent is returning a NULL pointer when
+	  it probably should be returning a pointer to the channel the
+	  Agent channel is pretending to be. (closes issue ASTERISK-21793)
+	  Reported by: Rodrigo P. Telles Patches:
+	  jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by
+	  rmudgett Tested by: Rodrigo P. Telles
+
+2013-05-28 17:35 +0000 [r389895]  Jonathan Rose <jrose at digium.com>
+
+	* main/slinfactory.c: Fix a memory copying bug in slinfactory which
+	  was causing mixmonitor issues. Reported by: Michael Walton Tested
+	  by: Jonathan Rose Patches: slinfactory.c.ASTERISK-21799.patch
+	  uploaded by Michael Walton (license 6502) (closes issue
+	  ASTERISK-21799)
+
+2013-05-24 11:42 +0000 [r389676]  Matthew Jordan <mjordan at digium.com>
+
+	* main/logger.c: Print all logger messages on shutdown When
+	  Asterisk shuts down and shuts down the loggin gsubsystem, any
+	  messages currently in flight will not get logged. This patch
+	  prevents the loop writing messages from breaking out prematurely,
+	  such that all of the messages are logged. (closes issue
+	  ASTERISK-21716) Reported by: Corey Farrell patches:
+	  logger-process-all-messages.patch uploaded by Corey Farrell
+	  (license 5909)
+
+2013-05-20 17:43 +0000 [r389244]  Jason Parker <jparker at digium.com>
+
+	* /: Add doxygen.log to svn:ignore property.
+
+2013-05-15 15:54 +0000 [r388838]  kharwell <kharwell at localhost>:
+
+	* main/lock.c: Fix for segfault in __ast_rwlock_destroy with
+	  DEBUG_THREADS If DEBUG_THREADS is enabled __ast_rwlock_destroy
+	  causes a segfault while trying to access a possible NULL t->track
+	  object. A NULL check has been added before trying to access the
+	  memory. (closes issue ASTERISK-21724) Reported by: Corey Farrell
+	  Fixed by: Corey Farrell Patches: ast_rwlock_destroy-segv.patch
+	  uploaded by Corey Farrell (license 5909)
+
+2013-05-15 12:37 +0000 [r388768]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_srtp.c, configure, include/asterisk/autoconfig.h.in,
+	  configure.ac: Use srtp_shutdown when available This allows the
+	  SRTP library to be shut down properly when the functionality is
+	  offered by libsrtp. Review:
+	  https://reviewboard.asterisk.org/r/2538/ (closes issue
+	  ASTERISK-21719)
+
+2013-05-13 20:34 +0000 [r388596]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_srtp.c: Revert r388529 for now Adding the cleanup
+	  function needs some deeper thought since it apparently doesn't
+	  exist for all variants of libsrtp.
+
+2013-05-13 18:16 +0000 [r388532]  Jonathan Rose <jrose at digium.com>
+
+	* main/pbx.c: pbx: Fix lack of cleanup on macrolock and
+	  context_table (closes issue ASTERISK-21723) Reported by: Corey
+	  Farrell Patches: core-pbx-cleanup.patch uploaded by Correy
+	  Farrell (license 5909)
+
+2013-05-13 18:05 +0000 [r388529]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_srtp.c: Close libsrtp properly Ensure that libsrtp is
+	  shutdown properly when res_srtp is unloaded. (closes issue
+	  ASTERISK-21719) Reported by: Corey Farrell Patches:
+	  res_srtp-library-shutdown.patch uploaded by Corey Farrell
+
+2013-05-13 14:24 +0000 [r388477]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/manager.c: Fix SendText AMI action to never return non-zero.
+	  AMI actions must never return non-zero unless they intend to
+	  close the AMI connection. (Which is almost never.) (closes issue
+	  ASTERISK-21779) Reported by: Paul Goldbaum
+
+2013-05-10 22:09 +0000 [r388423-388425]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/misdn/isdn_msg_parser.c: Allow mISDN to send PROGRESS
+	  messsage. * Made isdn_msg_parser.c build a progress message with
+	  the mandatory progress indicator IE. (The mISDNuser NT state
+	  machine rejected sending the incomplete message.) Note: The
+	  associated mISDN and mISDNuser patches respectively are viewable
+	  here: http://svnview.digium.com/svn/thirdparty?view=rev&rev=200
+	  http://svnview.digium.com/svn/thirdparty?view=rev&rev=201 (closes
+	  issue AST-1153) Reported by: Guenther Kelleter Patches:
+	  progress-chan_misdn.diff (license #6372) patch uploaded by
+	  Guenther Kelleter progress-misdn.diff (license #6372) mISDN patch
+	  uploaded by Guenther Kelleter progress-misdnuser.diff (license
+	  #6372) mISDNuser patch uploaded by Guenther Kelleter
+
+	* utils: Add version.c to list of ignored files in the utils
+	  directory.
+
+2013-05-10 20:28 +0000 [r388376]  Mark Michelson <mmichelson at digium.com>
+
+	* pbx/pbx_dundi.c: Fix memory leak in pbx_dundi pbx_dundi added an
+	  io context without removing it. This caused a memory leak when
+	  the module was unloaded. (closes ASTERISK-21718) Reported by
+	  Corey Farrell Patches: pbx_dundi-ast_io_remove.patch uploaded by
+	  Corey Farrell (License #5909)
+
+2013-05-09 03:58 +0000 [r388111]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* res/res_rtp_asterisk.c: Fix The Payload Being Set On CN Packets
+	  And Do Not Set Marker Bit When we send out a CN packet (for
+	  instance, in the case of using rtpkeepalives), we are not setting
+	  the payload code properly. Also, we are setting the marker bit
+	  when we shouldn't be according to RFC 3389, section 4. AST_RTP_CN
+	  is not defined by AST_FORMAT codes. Therefore, we should be using
+	  ast_rtp_codecs_payload_code() rather than
+	  ast_rtp_codecs_payload_lookup(). 11 and trunk already use the
+	  appropriate function. * In 1.8, use ast_rtp_codecs_payload_code()
+	  * Remove the setting of the marker bit * Fix the debug message by
+	  incrementing the seqno after the debug message is set in order to
+	  display the correct seqno that was sent out (closes issue
+	  ASTERISK-21246) Reported by: Peter Katzmann Tested by: Peter
+	  Katzmann, Michael L. Young Patches:
+	  asterisk-21246-rtp-cng-payload-error_1.8_v2.diff uploaded by
+	  Michael L. Young (license 5026) Review:
+	  https://reviewboard.asterisk.org/r/2500/
+
+2013-05-08 07:17 +0000 [r387875]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* channels/chan_sip.c: chan_sip: NOTIFYs for BLF start queuing up
+	  and fail to be sent out after retries fail RFC6665 4.2.2: ...
+	  after a failed State NOTIFY transaction remove the subscription
+	  The problem is that the State Notify requests rely on the 200OK
+	  reponse for pacing control and to not confuse the notify
+	  susbsystem. The issue is, the pendinginvite isn't cleared if a
+	  response isn't received, thus further notify's are never sent.
+	  The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the
+	  subscription after failure. (closes issue ASTERISK-21677)
+	  Reported by: Dan Martens Tested by: Dan Martens, David Brillert,
+	  alecdavis alecdavis (license 585) Review
+	  https://reviewboard.asterisk.org/r/2475/
+
+2013-05-06 15:52 +0000 [r387688]  Russell Bryant <russell at russellbryant.com>
+
+	* apps/app_meetme.c: Make SLA reload more paranoid. Reload support
+	  was originally not included for SLA. It was added later, but in a
+	  fairly non-traditional way. It basically sets a flag indicating
+	  that a reload is pending, and then waits for a time where it
+	  thinks everything SLA related is idle and unused, and *then*
+	  executes the reload. It does this because the reload process is
+	  destructive. It starts by throwing everything away and starting
+	  over. There are a number of problems with this approach. One of
+	  them is that the check to see if anything in use was incomplete.
+	  This patch makes it more complete and thus less likely for a
+	  crash to occur during reload processing. However, this approach
+	  still has problems so some much more significant reworking of
+	  this code will need to come in as a next step. Patch credit and
+	  testing by CoreDial, LLC.
+
+2013-05-02 17:11 +0000 [r387421]  Matthew Jordan <mjordan at digium.com>
+
+	* utils/Makefile: Update utils Makefile to handle r387294 Alec's
+	  patch that added the Asterisk version to 'core show locks'
+	  angered the items in utils, as they exist somewhat outside of the
+	  Asterisk build system. Some day, this Makefile should get nuked
+	  from high orbit, but for now, include version.c in its list of
+	  stuff to pile in.
+
+2013-05-02 07:53 +0000 [r387294-387344]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip:
+	  Session-Expires: Set timer to correctly expire at (~2/3) of the
+	  interval when not the refresher RFC 4028 Section 10 if the side
+	  not performing refreshes does not receive a session refresh
+	  request before the session expiration, it SHOULD send a BYE to
+	  terminate the session, slightly before the session expiration.
+	  The minimum of 32 seconds and one third of the session interval
+	  is RECOMMENDED. Prior to this asterisk would refresh at 1/2 the
+	  Session-Expires interval, or if the remote device was the
+	  refresher, asterisk would timeout at interval end. Now, when not
+	  refresher, timeout as per RFC noted above. (closes issue
+	  ASTERISK-21742) Reported by: alecdavis Tested by: alecdavis
+	  alecdavis (license 585) Review
+	  https://reviewboard.asterisk.org/r/2488/
+
+	* channels/chan_sip.c: chan_sip: Honor Session-Expires in 200OK
+	  response when it's a RE-INVITE when asterisk is the refresher.
+	  RFC 4028 Section 7.2 "UACs MUST be prepared to receive a
+	  Session-Expires header field in a response, even if none were
+	  present in the request." What changed After ASTERISK-20787,
+	  inbound calls to asterisk with no Session-Expires in the INVITE
+	  are now are offered a Session-Expires (1800 asterisk default) in
+	  the response, with asterisk as the refresher. Symptom: After 900
+	  seconds (asterisk default refresher period 1800), asterisk
+	  RE-INVITEs the device, the device may respond with a much lower
+	  Session-Expires (180 in our case) value that it is now using.
+	  Asterisk ignores this response, as it's deemed both an INBOUND
+	  CALL, and a RE-INVITE. After 180 seconds the device times out and
+	  sends BYE (hangs up), asterisk is still working with the
+	  refresher period of 1800 as it ignored the 'Session Expires: 180'
+	  in the previous 200OK response. Fix: handle_response_invite()
+	  when 200OK, remove check for outbound and reinvite. (closes issue
+	  ASTERISK-21664) Reported by: alecdavis Tested by: alecdavis
+	  alecdavis (license 585) Review
+	  https://reviewboard.asterisk.org/r/2463/
+
+	* channels/chan_dahdi.c: chan_dahdi: fix lower bound check with -ve
+	  integer conversion from a float Lower bound of a 16bit signed int
+	  is -32768 not -32767 (closes issue ASTERISK-21744) Reported by:
+	  alecdavis Tested by: alecdavis alecdavis (license 585)
+
+	* main/utils.c: Add Asterisk Version to core show locks Assist with
+	  reporting 'core show locks' when submitting bug reports. Example
+	  below: =========================== == SVN-branch-1.8-... ==
+	  Currently Held Locks =========================== (closes issue
+	  ASTERISK-21743) Reported by: alecdavis Tested by: alecdavis
+	  alecdavis (license 585)
+
+2013-05-01 21:15 +0000 [r387036-387213]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_rtp_asterisk.c: Clear the DTMF sending digit tracking on
+	  off nominal paths In certain situations, when the RTP engine goes
+	  to send a DTMF end digit it may be in a situation where the
+	  remote address is no longer available, or the digit that was
+	  supposed to be sent is invalid. In such cases, we need to clear
+	  the RTP counters appropriately. Otherwise, when the RTP source is
+	  set again, we'll continue to think that we're in the middle of
+	  sending a DTMF digit, which can confuse the remote party
+	  (signficantly). (closes issue ASTERISK-21522) Reported by: Corey
+	  Farrell patches: rtp_dtmf_process_end.patch uploaded by Corey
+	  Farrell (License 5909)
+
+	* channels/chan_sip.c: Prevent crash in 'sip show peers' when the
+	  number of peers on a system is large When you have lots of SIP
+	  peers (according to the issue reporter, around 3500), the 'sip
+	  show peers' CLI command or AMI action can crash due to a poorly
+	  placed string duplication that occurs on the stack. This patch
+	  refactors the command to not allocate the string on the stack,
+	  and handles the formatting of a single peer in a separate
+	  function call. (closes issue ASTERISK-21466) Reported by:
+	  Guillaume Knispel patches:
+	  fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch
+	  uploaded by gknispel (License 6492)
+
+	* main/features.c: Fix CDR not being created during an externally
+	  initiated blind transfer Way back when in the dark days of
+	  Asterisk 1.8.9, blind transferring a call in a context that
+	  included the 'h' extension would inadvertently execute the hangup
+	  code logic on the transferred channel. This was a "bad thing".
+	  The fix was to properly check for the softhangup flags on the
+	  channel and only execute the 'h' extension logic (and, in later
+	  versions, hangup handler logic) if the channel was well and truly
+	  dead (Jim). Unfortunately, CDRs are fickle. Setting the
+	  softhangup flag when we detected that the channel was leaving the
+	  bridge (but not to die) caused some crucial snippet of CDR code,
+	  lying in ambush in the middle of the bridging code, to not get
+	  executed. This had the effect of blowing away one of the CDRs
+	  that is typically created during a blind transfer. While we live
+	  and die by the adage "don't touch CDRs in release branches", this
+	  was our bad. The attached patch restores the CDR behavior, and
+	  still manages to not run the 'h' extension during a blind
+	  transfer (at least not when it's supposed to). Thanks to Steve
+	  Davies for diagnosing this and providing a fix. Review:
+	  https://reviewboard.asterisk.org/r/2476 (closes issue
+	  ASTERISK-21394) Reported by: Ishfaq Malik Tested by: Ishfaq
+	  Malik, mjordan patches: fix_missing_blindXfer_cdr2 uploaded by
+	  one47 (License 5012)
+
+2013-04-30 13:45 +0000 [r386929]  Sean Bright <sean at malleable.com>
+
+	* include/asterisk/utils.h: Use the proper lower bound when doing
+	  saturation arithmetic. 16 bit signed integers have a range of
+	  [-32768, 32768). The existing code was using the interval
+	  (-32768, 32768) instead. This patch fixes that. Review:
+	  https://reviewboard.asterisk.org/r/2479/
+
+2013-04-29 23:34 +0000 [r386877]  Rusty Newton <rnewton at digium.com>
+
+	* sounds/Makefile: Modifying sounds/Makefile to pull down 1.4.24
+	  core sounds 1.4.24 core sounds includes a full set of Italian
+	  prompts for core sounds and a fix for the missing voicemail
+	  prompts in the Russian language. (closes issue ASTERISK-19431)
+	  (closes issue ASTERISK-19721)
+
+2013-04-29 08:36 +0000 [r386792]  Olle Johansson <oej at edvina.net>
+
+	* CHANGES, apps/app_queue.c: Play periodic prompst for first call
+	  in a call queue Review: https://reviewboard.asterisk.org/r/2263/
+
+2013-04-26 21:26 +0000 [r386641-386672]  Matthew Jordan <mjordan at digium.com>
+
+	* main/config.c: Clean up memory leak in config file on off nominal
+	  paths when glob is allowed If a system allows for its usage,
+	  Asterisk will use glob to help parse Asterisk .conf files. The
+	  config file loading routine was leaking the memory allocated by
+	  the glob() routine when the config file was in an unmodified or
+	  invalid state. This patch properly calls globfree in those off
+	  nominal paths. (closes issue ASTERISK-21412) Reported by: Corey
+	  Farrell patches: config_glob_leak.patch uploaded by Corey Farrell
+	  (license 5909)
+
+	* main/features.c: Clean up resources in features on exit This
+	  patch cleans up two things features: * It properly unregisters
+	  the CLI commands that features registered * It cancels and
+	  performs a pthread_join on the created parking thread. This not
+	  only properly joins a non-detached thread, but also prevents
+	  disposing of the parking lots prior to the parking thread
+	  completely exiting. (closes issue ASTERISK-21407) Reported by:
+	  Corey Farrell patches: features_shutdown-r2.patch uploaded by
+	  Corey Farrell (License 5909)
+
+2013-04-25 02:43 +0000 [r386483]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* channels/chan_sip.c: Change Case On Forcerport For Consistency *
+	  Change "ForcerPort" to "Forcerport" to match everywhere else it
+	  is displayed
+
+2013-04-22 16:10 +0000 [r386256]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/channel.c: Fix crash when AMI redirect action redirects two
+	  channels out of a bridge. The two party bridging loops were
+	  changing the bridge peer pointers without the channel locks held.
+	  Thus when ast_channel_massquerade() tested and used the pointer
+	  there is a small window of opportunity for the pointers to become
+	  NULL even though the masquerade code has the channels locked.
+	  (closes issue ASTERISK-21356) Reported by: William luke Patches:
+	  jira_asterisk_21356_v11.patch (license #5621) patch uploaded by
+	  rmudgett Tested by: William luke
+
+2013-04-19 15:59 +0000 [r386109]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_timing_pthread.c: Prevent res_timing_pthread from
+	  blocking callers There were several reports of deadlock when
+	  using res_timing_pthread. Backtraces indicated that one thread
+	  was blocked waiting for the write to the pipe to complete and
+	  this thread held the container lock for the timers. Therefore any
+	  thread that wanted to create a new timer or read an existing
+	  timer would block waiting for either the timer lock or the
+	  container lock and deadlock ensued. This patch changes the way
+	  the pipe is used to eliminate this source of deadlocks: 1) The
+	  pipe is placed in non-blocking mode so that it would never block
+	  even if the following changes someone fail... 2) Instead of
+	  writing bytes into the pipe for each "tick" that's fired the pipe
+	  now has two states--signaled and unsignaled. If signaled, the
+	  pipe is hot and any pollers of the read side filedescriptor will
+	  be woken up. If unsigned the pipe is idle. This eliminates even
+	  the chance of filling up the pipe and reduces the potential
+	  overhead of calling unnecessary writes. 3) Since we're tracking
+	  the signaled / unsignaled state, we can eliminate the exta poll
+	  system call for every firing because we know that there is data
+	  to be read. (closes issue ASTERISK-21389) Reported by: Matt
+	  Jordan Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis patches:
+	  0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch
+	  uploaded by sruffell (License 5417) (closes issue ASTERISK-19754)
+	  Reported by: Nikola Ciprich (closes issue ASTERISK-20577)
+	  Reported by: Kien Kennedy (closes issue ASTERISK-17436) Reported
+	  by: Henry Fernandes (closes issue ASTERISK-17467) Reported by:
+	  isrl (closes issue ASTERISK-17458) Reported by: isrl Review:
+	  https://reviewboard.asterisk.org/r/2441/
+
+2013-04-19 05:18 +0000 [r386049]  David M. Lee <dlee at digium.com>
+
+	* main/cli.c: cli.c: Properly initialize debug_modules and
+	  verbose_modules. This avoids some lock errors on the core set
+	  {debug,verbose} commands.
+
+2013-04-16 23:11 +0000 [r385916]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* main/devicestate.c, res/res_jabber.c: Distributed Device State
+	  broken at sites using res_xmpp or res_jabber where Secuity
+	  Advisory AST-2012-015 is inplace res_jabber/res_xmpp were not
+	  adding AST_EVENT_IE_CACHABLE to the event as each message came
+	  in, then devstate_change_collector_cb() was unable to find
+	  AST_EVENT_IE_CACHABLE in the event, so defaulted incorrectly to
+	  AST_DEVSTATE_NOT_CACHABLE. (issue ASTERISK-20175) (closes issue
+	  ASTERISK-21429) (closes issue ASTERISK-21069) (closes issue
+	  ASTERISK-21164) Reported by: alecdavis Tested by: alecdavis
+	  alecdavis (license 585) Review
+	  https://reviewboard.asterisk.org/r/2452/
+
+2013-04-15 17:07 +0000 [r385745]  Jason Parker <jparker at digium.com>
+
+	* Makefile: Don't unnecessarily rebuild things on every run of
+	  'make'. Review: https://reviewboard.asterisk.org/r/2449/
+
+2013-04-15 14:38 +0000 [r385683]  David M. Lee <dlee at digium.com>
+
+	* BSDmakefile, contrib/realtime/mysql/voicemail_data.sql,
+	  build_tools/sha1sum-sh, res/res_mutestream.c,
+	  configs/res_curl.conf.sample, tests/test_func_file.c,
+	  include/asterisk/select.h, res/res_rtp_multicast.c,
+	  include/asterisk/bridging_technology.h, tests/test_locale.c,
+	  include/asterisk/bridging_features.h, doc/Makefile,
+	  tests/test_poll.c, res/res_timing_kqueue.c,
+	  contrib/realtime/mysql/musiconhold.sql,
+	  contrib/realtime/mysql/queue_log.sql, channels/sig_ss7.c,
+	  channels/sig_ss7.h, channels/chan_multicast_rtp.c,
+	  tests/test_expr.c, apps/app_saycounted.c,
+	  contrib/realtime/mysql/voicemail_messages.sql: Fix the
+	  svn:keywords property on several files. Normally I think keyword
+	  expansion is silly, but the one time it would have been good, it
+	  didn't work because the property had quotes in it. This patch
+	  fixes obviously busted svn:keywords properties.
+
+2013-04-14 02:58 +0000 [r385633-385636]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_rtp_multicast.c: Calculate the timestamp for outbound RTP
+	  if we don't have timing information This patch calculates the
+	  timestamp for outbound RTP when we don't have timing information.
+	  This uses the same approach in res_rtp_asterisk. Thanks to both
+	  Pietro and Tzafrir for providing patches. (closes issue
+	  ASTERISK-19883) Reported by: Giacomo Trovato Tested by: Pietro
+	  Bertera, Tzafrir Cohen patches: rtp-timestamp-1.8.patch uploaded
+	  by tzafrir (License 5035) rtp-timestamp.patch uploaded by
+	  pbertera (License 5943)
+
+	* channels/chan_alsa.c: Don't attempt to create a voice frame on a
+	  read error Prior to this patch, a read error in snd_pcm_readi
+	  would still be treated as a nominal result when constructing a
+	  voice frame from the expected data. Since the value returned is
+	  negative, as opposed to the number of samples read, this could
+	  result in a crash. With this patch, we now return a null frame
+	  when a read error is detected. Note that the patch on
+	  ASTERISK-21329 was modified slightly for this commit, in that we
+	  bail immediately on detecting the read error, rather than
+	  bypassing the construction of the voice frame. (closes issue
+	  ASTERISK-21329) Reported by: Keiichiro Kawasaki patches:
+	  chan_alsa.diff uploaded by kawasaki (License 6489)
+
+2013-04-12 22:34 +0000 [r385551-385593]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* apps/app_queue.c: Fix Manager Segfault When app_queue Is Unloaded
+	  When app_queue is unloaded, some manager commands are not being
+	  unregistered which result in a segfault. This patch corrects
+	  this. (closes issue ASTERISK-21397) Reported by: Peter Katzmann,
+	  Corey Farrell Tested by: Corey Farrell Patches:
+	  asterisk-21397-missing-unreg-manager-cmd_1.8.diff Michael L.
+	  Young (license 5026)
+	  asterisk-21397-missing-unreg-manager-cmd_11.diff Michael L. Young
+	  (license 5026) Review: https://reviewboard.asterisk.org/r/2444/
+
+	* apps/app_voicemail.c: Fix app_voicemail Segfault And A Few Memory
+	  Leaks The original report was that app_voicemail would crash.
+	  This was caused by ast_config_load() returning
+	  CONFIG_STATUS_FILEINVALID but no checks being performed for that
+	  return status. After adding the initial patch to fix this issue,
+	  Jaco Kroon (jkroon) added some fixes to memory leaks he had
+	  discovered. During review, Walter Doekes (wdoekes) suggested
+	  adding a helper function in order to determine if we had a valid
+	  configuration or not. This patch does the following: * Creates a
+	  helper function to check if the configuration is valid * Adds
+	  calls to the new helper function where appropiate * Fixes memory
+	  leaks where the code returned without running
+	  ast_config_destroy() on the configuration that was loaded (closes
+	  issue ASTERISK-21302) Reported by: Jaco Kroon Tested by: Jaco
+	  Kroon, Michael L. Young Patches:
+	  asterisk-11.3.0-app_voicemail-ast_config-fixes.patch Jaco Kroon
+	  (license 5671) asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff
+	  Michael L. Young (license 5026) Review:
+	  https://reviewboard.asterisk.org/r/2443/
+
+2013-04-12 08:46 +0000 [r385402-385429]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* channels/chan_iax2.c: IAX2 defer_full_frames fail to get sent
+	  Ensure iax2_process_thread is signalled when a deferred frame is
+	  queued to it. (issue ASTERISK-18827) Reported by: alecdavis
+	  Tested by: alecdavis alecdavis (license 585) Review
+	  https://reviewboard.asterisk.org/r/2426/
+
+	* channels/chan_iax2.c: IAX2, prevent network thread starting
+	  before all helper threads are ready On startup, it's possible for
+	  a frame to arrive before the processing threads were ready. In
+	  iax2_process_thread() the first pass through falls into
+	  ast_cond_wait, should a frame arrive before we are at
+	  ast_cond_wait, the signal will be ignored. The result
+	  iax2_process_thread stays at ast_cond_wait forever, with deferred
+	  frames being queued. Fix: When creating initial idle
+	  iax2_process_threads, wait for init_cond to be signalled after
+	  each thread is started. (issue ASTERISK-18827) Reported by:
+	  alecdavis Tested by: alecdavis alecdavis (license 585) Review
+	  https://reviewboard.asterisk.org/r/2427/
+
+2013-04-10 14:22 +0000 [r385170-385190]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_config_ldap.c: Use LDAP memory management functions
+	  instead of Asterisk's When MALLOC_DEBUG is enabled with
+	  res_config_ldap, issues (munmap_chunk: invalid pointer errors)
+	  can occur as the memory is being allocated with Asterisk's
+	  wrappers around malloc/calloc/free/strdup, as opposed to the LDAP
+	  library's wrappers. This patch uses the LDAP library's wrappers
+	  where appropriate, so that compiling with MALLOC_DEBUG doesn't
+	  cause more problems than it solves. Note that the patch listed
+	  below was modified slightly for this commit to account for some
+	  additional memory allocation/deallocations. (closes issue
+	  ASTERISK-17386) Reported by: John Covert Tested by: Andrew Latham
+	  patches: issue18789-1.8-r316873.patch uploaded by seanbright
+	  (License 5060)
+
+	* channels/chan_sip.c: Fix crash in chan_sip when a core initiated
+	  op occurs at the same time as a BYE When a BYE request is
+	  processed in chan_sip, the current SIP dialog is detached from
+	  its associated Asterisk channel structure. The tech_pvt pointer
+	  in the channel object is set to NULL, and the dialog persists for
+	  an RFC mandated period of time to handle re-transmits. While this
+	  process occurs, the channel is locked (which is good).
+	  Unfortunately, operations that are initiated externally have no
+	  way of knowing that the channel they've just obtained (which is
+	  still valid) and that they are attempting to lock is about to
+	  have its tech_pvt pointer removed. By the time they obtain the
+	  channel lock and call the channel technology callback, the
+	  tech_pvt is NULL. This patch adds a few checks to some channel
+	  callbacks that make sure the tech_pvt isn't NULL before using it.
+	  Prime offenders were the DTMF digit callbacks, which would crash
+	  if AMI initiated a DTMF on the channel at the same time as a BYE
+	  was received from the UA. This patch also adds checks on
+	  sip_transfer (as AMI can also cause a callback into this
+	  function), as well as sip_indicate (as lots of things can queue
+	  an indication onto a channel). Review:
+	  https://reviewboard.asterisk.org/r/2434/ (closes issue
+	  ASTERISK-20225) Reported by: Jeff Hoppe
+
+2013-04-08 23:34 +0000 [r385047]  Rusty Newton <rnewton at digium.com>
+
+	* configs/extconfig.conf.sample: Modified the list of keys for the
+	  driver backends for sake of sample clarity Added a line showing
+	  the mapping of "mysql" to res_config_mysql available in add-ons.
+	  We used "mysql" as an example driver key in the sample, but
+	  didn't show what module it mapped too. Also added a subtitle
+	  above the list of keys for driver backends.
+
+2013-04-08 19:55 +0000 [r385008]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* UPGRADE.txt, channels/chan_sip.c: Fix For Not Overriding The
+	  Default Settings In chan_sip The initial report was that the
+	  "nat" setting in the [general] section was not having any effect
+	  in overriding the default setting. Upon confirming that this was
+	  happening and looking into what was causing this, it was
+	  discovered that other default settings would not be overriden as
+	  well. This patch works similar to what occurs in build_peer(). We
+	  create a temporary ast_flags structure and using a mask, we
+	  override the default settings with whatever is set in the
+	  [general] section. In the bug report, the reporter who helped to
+	  test this patch noted that the directmedia settings were being
+	  overriden properly as well as the nat settings. (closes issue
+	  ASTERISK-21225) Reported by: Alexandre Vezina Tested by:
+	  Alexandre Vezina, Michael L. Young Patches:
+	  asterisk-21225-handle-options-default-prob_1.8_v4.diff.diff
+	  Michael L. Young (license 5026) Review:
+	  https://reviewboard.asterisk.org/r/2386/
+
+2013-04-04 19:31 +0000 [r384779]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* contrib/realtime/postgresql/realtime.sql,
+	  contrib/realtime/mysql/sippeers.sql, channels/chan_sip.c:
+	  Backport Appropiate NAT Setting Cleanup In ASTERISK-20904, the
+	  focus was around the changes to NAT that took place in Asterisk
+	  11. Since the report stated that 1.8 was fine, we didn't take a
+	  look at 1.8 at the time. While working on ASTERISK-21225, I could
+	  see that 1.8 would benefit from having some of those changes
+	  applied to it. This patch does the following: * The important
+	  part of this patch is that it sets the peer's flags earlier in
+	  build_peer so that the code properly uses the peer's flags based
+	  on the peer's configuration. * constify req parameter in
+	  check_via() * update realtime schemas under the contrib directory
+	  to handle properly the NAT settings available in 1.8 as well as
+	  to handle the changes made in 11 to make upgrading easier when
+	  installing newer versions of Asterisk (closes issue
+	  ASTERISK-21243) Reported by: Michael L. Young Patches:
+	  asterisk-20904-changes_for_1.8.diff Michael L. Young (license
+	  5026) Review: https://reviewboard.asterisk.org/r/2422/
+
+2013-04-03 20:13 +0000 [r384685]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
+	  configs/chan_dahdi.conf.sample: chan_dahdi: Add
+	  inband_on_proceeding compatibility option. The new
+	  inband_on_proceeding option causes Asterisk to assume inband
+	  audio may be present when a PROCEEDING message is received. Q.931
+	  Section 5.1.2 says the network cannot assume that the CPE side
+	  has attached to the B channel at this time without explicitly
+	  sending the progress indicator ie informing the CPE side to
+	  attach to the B channel for audio. However, some non-compliant
+	  ISDN switches send a PROCEEDING without the progress indicator ie
+	  indicating inband audio is available and assume that the CPE
+	  device has connected the media path for listening to ringback and
+	  other messages. ASTERISK-17834 which causes this issue was
+	  dealing with a non-compliant network switch. (closes issue
+	  ASTERISK-21151) Reported by: Gianluca Merlo Tested by: rmudgett
+
+2013-04-03 17:05 +0000 [r384640]  Matthew Jordan <mjordan at digium.com>
+
+	* funcs/func_channel.c: Update documentation for CHANNEL function
+	  Document that you can read/write the 'accountcode' and 'amaflags'
+	  on a channel.
+
+2013-04-02 17:33 +0000 [r384544]  David M. Lee <dlee at digium.com>
+
+	* Makefile: Fixed spurious rebuilds of func_version.
+	  func_version.so was being rebuilt every time, because build.h was
+	  changing every build, because of the cleantest dependency that
+	  was added in r384410 to fix parallel make bugs. Now build.h will
+	  only be created if it does not exist, which was the original
+	  behavior of the Makefile.
+
+2013-04-01 13:18 +0000 [r384410]  David M. Lee <dlee at digium.com>
+
+	* Makefile: Fix parallel make problems. Occasionally, make -j would
+	  fail due to missing includes, or other unusual errors. This was
+	  due to the 'cleantest' target, which was designed to force a make
+	  clean when some change in the code would cause the typical
+	  depedency checking to fail. Several targets in the main Makefile
+	  did not depend upon cleantest, hence would run in parallel to it.
+	  By adding the dependency, make -j runs happily now. Review:
+	  https://reviewboard.asterisk.org/r/2418/
+
+2013-03-29 16:23 +0000 [r384325]  Jonathan Rose <jrose at digium.com>
+
+	* apps/app_voicemail.c: app_voicemail: Add blank argument to
+	  externnotify if no context argument At least one call to
+	  run_externnotify provides a NULL context parameter and because
+	  the snprintf statement doesn't account for a NULL context
+	  parameter, it simply writes '(null)' to the arguments string
+	  instead. This patch makes it write two quotes back to back for
+	  that argument instead in the event of a NULL context. (closes
+	  issue ASTERISK-18207) Reported by: Barry L. Kline Patches:
+	  modified from patch-20130306 uploaded by Karsten Wemheuer
+	  (License 5930)
+
+2013-05-17  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.22.0 Released.
+
+2013-05-13  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.22.0-rc2 Released.
+
+	* Distributed Device State broken at sites using res_xmpp or res_jabber
+	  where Secuity Advisory AST-2012-015 is inplace
+
+	  res_jabber/res_xmpp were not adding AST_EVENT_IE_CACHABLE to the
+	  event as each message came in, then devstate_change_collector_cb()
+	  was unable to find AST_EVENT_IE_CACHABLE in the event, so defaulted
+	  incorrectly to AST_DEVSTATE_NOT_CACHABLE.
+
+	* Fix CDR not being created during an externally initiated blind
+	  transfer
+
+	  Way back when in the dark days of Asterisk 1.8.9, blind transferring
+	  a call in a context that included the 'h' extension would
+	  inadvertently execute the hangup code logic on the transferred
+	  channel. This was a "bad thing". The fix was to properly check for
+	  the softhangup flags on the channel and only execute the 'h'

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