[asterisk-commits] bebuild: tag 1.8.9.0-rc1 r349394 - /tags/1.8.9.0-rc1/ChangeLog

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Dec 29 13:19:22 CST 2011


Author: bebuild
Date: Thu Dec 29 13:19:18 2011
New Revision: 349394

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=349394
Log:
Add change log for 1.8.9.0

Added:
    tags/1.8.9.0-rc1/ChangeLog   (with props)

Added: tags/1.8.9.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.9.0-rc1/ChangeLog?view=auto&rev=349394
==============================================================================
--- tags/1.8.9.0-rc1/ChangeLog (added)
+++ tags/1.8.9.0-rc1/ChangeLog Thu Dec 29 13:19:18 2011
@@ -1,0 +1,36241 @@
+2011-12-29 15:13 +0000 [r349339]  Matthew Jordan <mjordan at digium.com>
+
+	* main/rtp_engine.c: Handle AST_CONTROL_UPDATE_RTP_PEER frames in
+	  local bridge loop Failing to handle AST_CONTROL_UPDATE_RTP_PEER
+	  frames in the local bridge loop causes the loop to exit
+	  prematurely. This causes a variety of negative side effects,
+	  depending on when the loop exits. This patch handles the frame by
+	  essentially swallowing the frame in the local loop, as the
+	  current channel drivers expect the RTP bridge to handle the
+	  frame, and, in the case of the local bridge loop, no additional
+	  action is necessary. (issue ASTERISK-19040) (issue
+	  ASTERISK-19128) (issue ASTERISK-17725) (issue ASTERISK-18340)
+	  (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
+	  by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1640/
+
+2011-12-28 21:30 +0000 [r349289]  Sean Bright <sean at malleable.com>
+
+	* main/audiohook.c: Use ast_audiohook_write_list_empty to determine
+	  if our lists are empty instead of duplicating that logic.
+
+2011-12-27 20:48 +0000 [r349194]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_musiconhold.c, res/res_timing_pthread.c,
+	  include/asterisk/module.h, res/res_timing_dahdi.c,
+	  res/res_timing_timerfd.c: Fix timing source dependency issues
+	  with MOH Prior to this patch, res_musiconhold existed at the same
+	  module priority level as the timing sources that it depends on.
+	  This would cause a problem when music on hold was reloaded, as
+	  the timing source could be changed after res_musiconhold was
+	  processed. This patch adds a new module priority level,
+	  AST_MODPRI_TIMING, that the various timing modules are now loaded
+	  at. This now occurs before loading other resource modules, such
+	  that the timing source is guaranteed to be set prior to resolving
+	  the timing source dependencies. (closes issue ASTERISK-17474)
+	  Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf,
+	  Wes Van Tlghem, elguero, Thomas Arimont Patches:
+	  asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff
+	  uploaded by elguero (License #5026)
+	  asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff
+	  uploaded by elguero (License #5026)
+	  asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by
+	  elguero (License #5026) Review:
+	  https://reviewboard.asterisk.org/r/1578/
+
+2011-12-27 17:09 +0000 [r349144]  Sean Bright <sean at malleable.com>
+
+	* main/audiohook.c: Once an audiohook is attached to a channel, we
+	  continue to transcode all of the frames, even after all of the
+	  hooks are detached. This patch short-cicuits us out before we
+	  transcode unnecessarily.
+
+2011-12-23 17:25 +0000 [r349044]  Sean Bright <sean at malleable.com>
+
+	* apps/app_chanspy.c: In ChanSpy, don't create audiohooks that will
+	  never be used. When ChanSpy is initialized it creates and
+	  attaches 3 audiohooks: 1) Read audio off of the channel that we
+	  are spying on 2) Write audio to the channel that we are spying on
+	  3) Write audio to the channel that is bridged to the channel that
+	  we are spying on. The first is always necessary, but the others
+	  are used only when specific options are passed to the ChanSpy
+	  application (B, d, w, and W to be specific). When those flags are
+	  not passed, neither of those audiohooks are ever sent frames, but
+	  we still try to process the hooks for each voice frame that we
+	  recieve on the channel. So in short - only create and attach
+	  audiohooks that we actually need.
+
+2011-12-23 15:24 +0000 [r348992]  Kinsey Moore <kmoore at digium.com>
+
+	* apps/app_dial.c: Fix missing doc tags found while fixing
+	  ASTERISK-18689 Add missing <variable></variable> tags in app_dial
+	  documentation.
+
+2011-12-23 02:09 +0000 [r348940]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/pbx.h, main/pbx.c, channels/chan_sip.c: Fix
+	  extension state callback references in chan_sip. Chan_sip gives a
+	  dialog reference to the extension state callback and assumes that
+	  when ast_extension_state_del() returns, the callback cannot
+	  happen anymore. Chan_sip then reduces the dialog reference count
+	  associated with the callback. Recent changes (ASTERISK-17760)
+	  have resulted in the potential for the callback to happen after
+	  ast_extension_state_del() has returned. For chan_sip, this could
+	  be very bad because the dialog pointer could have already been
+	  destroyed. * Added ast_extension_state_add_destroy() so chan_sip
+	  can account for the sip_pvt reference given to the extension
+	  state callback when the extension state callback is deleted. *
+	  Fix pbx.c awkward statecbs handling in
+	  ast_extension_state_add_destroy() and handle_statechange() now
+	  that the struct ast_state_cb has a destructor to call. * Ensure
+	  that ast_extension_state_add_destroy() will never return -1 or 0
+	  for a successful registration. * Fixed pbx.c statecbs_cmp() to
+	  compare the correct information. The passed in value to compare
+	  is a change_cb function pointer not an object pointer. * Make
+	  pbx.c ast_merge_contexts_and_delete() not perform callbacks with
+	  AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for
+	  deadlocking when those locks are held during the callback. *
+	  Removed unused lock declaration for the pbx.c store_hints list.
+	  (closes issue ASTERISK-18844) Reported by: rmudgett Review:
+	  https://reviewboard.asterisk.org/r/1635/
+
+2011-12-22 22:31 +0000 [r348888]  Matthew Jordan <mjordan at digium.com>
+
+	* cel/cel_pgsql.c: Fix for memory leaks / cleanup in cel_pgsql
+	  There were a number of issues in cel_pgsql's pgsql_log method: *
+	  If either sql or sql2 could not be allocated, the method would
+	  return while the pgsql_lock was still locked * If the execution
+	  of the log statement succeeded, the sql and sql2 structs were
+	  never free'd * Reconnection successes were logged as ERRORs. In
+	  general, the severity of several logging statements was reduced
+	  (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested
+	  by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/
+
+2011-12-22 18:38 +0000 [r348833]  Terry Wilson <twilson at digium.com>
+
+	* include/asterisk/frame.h: Allow packetization vaules > 127
+	  According to the RTP packetization documentation, and the maximum
+	  values listed in AST_FORMAT_LIST, we should support values > that
+	  the signed char array that ast_codec_pref makes available to
+	  store the value. All places in the code treat the framing field
+	  as though it were an int array instaead of a char array anyway,
+	  so this just fixes the type of the array. (closes issue
+	  ASTERISK-18876) Review: https://reviewboard.asterisk.org/r/1639/
+
+2011-12-20 23:08 +0000 [r348735]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_iax2.c: Fix chan_iax2 to not report an RDNIS number
+	  if it is blank. Some ISDN switches complain or block the call if
+	  the RDNIS number is empty. * Made chan_iax2 not save a RDNIS
+	  number into the ast_channel if the string is blank. This is what
+	  other channel drivers do. (closes issue ASTERISK-17152) Reported
+	  by: rmudgett
+
+2011-12-19 21:31 +0000 [r348647]  Richard Mudgett <rmudgett at digium.com>
+
+	* configure, configure.ac: Fix crashes on other platforms caused by
+	  interference from Darwin weak symbol support. Support weak
+	  symbols on a platform specific basis. The Mac OS X (Darwin)
+	  support must be isolated from the other platforms because it has
+	  caused other platforms to crash. Several other platforms
+	  including Linux have GCC versions that define the weak attribute.
+	  However, this attribute is only setup for use in the code by
+	  Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang
+	  Review: https://reviewboard.asterisk.org/r/1617/
+
+2011-12-18 18:27 +0000 [r348516]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* configs/sip.conf.sample, /: Correct two flaws in sip.conf.sample
+	  related to AST-2011-013. * The sample file listed *two* values
+	  for the 'nat' option as being the default. Only 'force_rport' is
+	  the default. * The warning about having differing 'nat' settings
+	  confusingly referred to both peers and users. ........ Merged
+	  revisions 348515 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+
+2011-12-16 23:51 +0000 [r348310-348464]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/channel.c, main/features.c: Clean-up on isle five for
+	  __ast_request_and_dial() and ast_call_forward(). * Add locking
+	  when a channel inherits variables and datastores in
+	  __ast_request_and_dial() and ast_call_forward(). Note: The
+	  involved channels are not active so there was minimal potential
+	  for problems. * Remove calls to ast_set_callerid() in
+	  __ast_request_and_dial() and ast_call_forward() because the set
+	  information is for the wrong direction. * Don't use C++ keywords
+	  for variable names in ast_call_forward(). * Run the redirecting
+	  interception macro if defined when forwarding a call in
+	  ast_call_forward(). Note: Currently will never execute because
+	  the only callers that supply a calling channel supply a hungup or
+	  zombie channel. * Make feature_request_and_dial() put the
+	  transferee into autoservice when it calls ast_call_forward() in
+	  case a redirection interception macro is run. Note: Currently
+	  will never happen because the caller channel (Party B) is always
+	  hungup at this time. * Make feature_request_and_dial() ignore the
+	  AST_CONTROL_PROCEEDING frame to silence a log message.
+
+	* main/channel.c: Fix cut and past error in ast_call_forward().
+	  (issue ASTERISK-18836)
+
+	* include/asterisk/cdr.h, apps/app_followme.c, apps/app_queue.c,
+	  res/res_monitor.c, main/channel.c, main/pbx.c,
+	  apps/app_authenticate.c, funcs/func_cdr.c, main/features.c: Fix
+	  crash during CDR update. The ast_cdr_setcid() and
+	  ast_cdr_update() were shown in ASTERISK-18836 to be called by
+	  different threads for the same channel. The channel driver thread
+	  and the PBX thread running dialplan. * Add lock protection around
+	  CDR API calls that access an ast_channel pointer. (closes issue
+	  ASTERISK-18836) Reported by: gpluser Review:
+	  https://reviewboard.asterisk.org/r/1628/
+
+	* apps/app_parkandannounce.c: Fix ParkAndAnnounce to pass the
+	  CallerID to the announcing channel. ParkAndAnnounce tried to pass
+	  the CallerID to the announcing channel but the ID was wiped out
+	  by the channel masquerade done when parking the call. * Save the
+	  CallerID before parking the channel to pass it to the announcing
+	  channel. * Fixed a minor memory leak in ParkAndAnnounce. *
+	  Updated some ParkAndAnnounce log messages.
+
+2011-12-14 22:01 +0000 [r348212]  Matthew Nicholson <mnicholson at digium.com>
+
+	* res/res_fax.c: Don't clear LOCALSTATIONID before sending or
+	  receiving. The user may set that variable. ASTERISK-18921
+
+2011-12-14 20:34 +0000 [r348154-348157]  Jonathan Rose <jrose at digium.com>
+
+	* configs/features.conf.sample: Fix accidental use of tabs instead
+	  of spaces from previous features.conf.sample change
+
+	* configs/features.conf.sample: Document PARKINGSLOT variable in
+	  features.conf.sample (issue ASTERISK-16239)
+
+2011-12-13 23:00 +0000 [r348101]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_followme.c, bridges/bridge_builtin_features.c: Fix
+	  FollowMe CallerID on outgoing calls. The addition of the
+	  Connected Line support changed how CallerID is passed to outgoing
+	  calls. The FollowMe application was not updated to pass CallerID
+	  to the outgoing calls. * Fix FollowMe CallerID on outgoing calls.
+	  * Restructured findmeexec() to fix several memory leaks and
+	  eliminate some duplicated code. * Made check the return value of
+	  create_followme_number(). Putting a NULL into the numbers list is
+	  bad if create_followme_number() fails. * Fixed a couple uses of
+	  ast_strdupa() inside loops. * The changes to
+	  bridge_builtin_features.c fix a similar CallerID issue with the
+	  bridging API attended and blind transfers. (Not used at this
+	  time.) (closes issue ASTERISK-17557) Reported by: hamlet505a
+	  Tested by: rmudgett Review:
+	  https://reviewboard.asterisk.org/r/1612/
+
+2011-12-13 15:16 +0000 [r348048]  Stefan Schmidt <sst at sil.at>
+
+	* channels/chan_sip.c: Fix possible misshandling of an incoming SIP
+	  response as a peer poke response. Also make sure peer has even
+	  qualify enabled when handle a peer poke response. (closes issue
+	  ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and
+	  UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed
+	  by: David Vossel
+
+2011-12-12 19:22 +0000 [r347995]  Terry Wilson <twilson at digium.com>
+
+	* res/res_srtp.c: Add a separate buffer for SRTCP packets The
+	  function ast_srtp_protect used a common buffer for both SRTP and
+	  SRTCP packets. Since this function can be called from multiple
+	  threads for the same SRTP session (scheduler for SRTCP and
+	  channel for SRTP) it was possible for the packets to become
+	  corrupted as the buffer was used by both threads simultaneously.
+	  This patch adds a separate buffer for SRTCP packets to avoid the
+	  problem. (closes issue ASTERISK-18889, Reported/patch by Daniel
+	  Collins)
+
+2011-12-09 01:19 +0000 [r347811]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/pbx.c: Fix some parsing issues in
+	  add_exten_to_pattern_tree(). * Simplify compare_char() and avoid
+	  potential sign extension issue. * Fix infinite loop in
+	  add_exten_to_pattern_tree() handling of character set escape
+	  handling. * Added buffer overflow checks in
+	  add_exten_to_pattern_tree() character set collection. * Made
+	  ignore empty character sets. * Added escape character handling to
+	  end-of-range character in character sets. This has a slight
+	  change in behavior if the end-of-range character is an escape
+	  character. You must now escape it. * Fix potential sign extension
+	  issue when expanding character set ranges. * Made remove
+	  duplicated characters from character sets. The duplicate
+	  characters lower extension matching priority and prevent
+	  duplicate extension detection. * Fix escape character handling
+	  when the escape character is trying to escape the end-of-string.
+	  We could have continued processing characters after the end of
+	  the exten string. We could have added the previous character to
+	  the pattern matching tree incorrectly. (closes issue
+	  ASTERISK-18909) Reported by: Luke-Jr
+
+2011-12-08 21:28 +0000 [r347718]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* channels/chan_sip.c: Fix regression when using tcpenable=no and
+	  tlsenable=yes. The tlsenable settings are tucked away in
+	  main/tcptls.c, so I missed them when resolving ASTERISK-18837.
+	  This should resolve the test suite breakage of the sip tls tests.
+	  Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt
+	  Jordan
+
+2011-12-08 17:50 +0000 [r347595]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/features.c: Mark channel running the h exten with the
+	  soft-hangup flag. When a bridge is broken, ast_bridge_call()
+	  might execute the h exten on the calling channel. However, that
+	  channel may not have been the channel that broke the bridge by
+	  hanging up. The channel executing the h exten must be in a hung
+	  up state so things like AGI run in the correct mode. * Make sure
+	  ast_bridge_call() marks the channel it is executing the h exten
+	  on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as
+	  to match the pbx.c main dialplan execution loop when it executes
+	  the h exten.) (closes issue ASTERISK-18811) Reported by: David
+	  Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621)
+	  patch uploaded by rmudgett Tested by: David Hajek, rmudgett
+
+2011-12-08 16:19 +0000 [r347531]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c: Don't crash on INFO automon request with
+	  no channel AST-2011-014. When automon was enabled in
+	  features.conf, it was possible to crash Asterisk by sending an
+	  INFO request if no channel had been created yet. (closes issue
+	  ASTERISK-18805) ........ Merged revisions 347530 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+
+2011-12-07 21:36 +0000 [r347438]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/manager.c: Update AMI Getvar and Setvar documentation about
+	  supplying a channel name. (closes issue ASTERISK-18958) Reported
+	  by: Red Patches: jira_asterisk_18958_v1.8.patch (license #5621)
+	  patch uploaded by rmudgett
+
+2011-12-07 20:23 +0000 [r347369]  Jonathan Rose <jrose at digium.com>
+
+	* apps/app_meetme.c: Fix: Meetme recording variables from realtime
+	  DB use null entries over channel variables Meetme would attempt
+	  to substitute the realtime values of RECORDING_FILE and
+	  RECORDING_FORMAT from the meetme db entry instead of using the
+	  channel variable set for those variables in spite of those
+	  database entries being NULL or even lacking a column to represent
+	  them. (closes issue ASTERISK-18873) Reported by: Byron Clark
+	  Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license
+	  6157)
+
+2011-12-06 23:47 +0000 [r347292]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_sip.c: Make SIP INFO messages for dtmf-relay
+	  signals case insensitive. (closes issue ASTERISK-18924) Reported
+	  by: Kevin Taylor
+
+2011-12-06 21:44 +0000 [r347239]  Jonathan Rose <jrose at digium.com>
+
+	* main/pbx.c: Documents CHANNEL(musicclass) taking priority over
+	  m([x]) in waitExten If waitExten specifies a music class to use
+	  with its music on hold option, it will use CHANNEL(musicclass)
+	  instead if that channel variable has been set on the initiating
+	  channel. This documents that behavior in the waitExten app so
+	  that this can be known without checking the documentation of the
+	  code in function local_ast_moh_start. (closes issue
+	  ASTERISK-18804)
+
+2011-12-06 19:39 +0000 [r347111-347166]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* channels/chan_sip.c: Don't allow transport=tcp when tcpenable=no.
+	  When tcpenable=no, sending to transport=tcp hosts was still
+	  allowed. Resolving the source address wasn't possible and yielded
+	  the string "(null)" in SIP messages. Fixed that and a couple of
+	  not-so-correct log messages. (closes issue ASTERISK-18837)
+	  Reported by: Andreas Topp Review:
+	  https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan
+
+	* apps/app_voicemail.c: Add regression tests for issue
+	  ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572
+	  Reviewed by: Matt Jordan
+
+	* apps/app_voicemail.c: Move setting of voicemail zonetag and
+	  locale up a bit. The voicemail [general] zonetag and locale
+	  variables weren't loaded until after the mailboxes were
+	  initialized. This caused the settings to be unset for those
+	  mailboxes until a reload was performed. (closes issue
+	  ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570
+	  Reviewed by: Matt Jordan
+
+2011-12-06 17:05 +0000 [r347058]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/sip/include/sip.h, channels/chan_sip.c: Fixed crash from
+	  orphaned MWI subscriptions in chan_sip This patch resolves the
+	  issue where MWI subscriptions are orphaned by subsequent SIP
+	  SUBSCRIBE messages. When a peer is removed, either by pruning
+	  realtime SIP peers or by unloading / loading chan_sip, the MWI
+	  subscriptions that were orphaned would still be on the event
+	  engine list of valid subscriptions but have a pointer to a peer
+	  that no longer was valid. When an MWI event would occur, this
+	  would cause a seg fault. (closes issue ASTERISK-18663) Reported
+	  by: Ross Beer Tested by: Ross Beer, Matt Jordan Patches:
+	  blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
+	  Review: https://reviewboard.asterisk.org/r/1610/
+
+2011-12-05 17:39 +0000 [r347006]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, channels/sig_analog.c,
+	  channels/sig_analog.h: Restore call progress code for analog
+	  ports. Extracting sig_analog from chan_dahdi lost call progress
+	  detection functionality. * Fix analog ports from considering a
+	  call answered immediately after dialing has completed if the
+	  callprogress option is enabled. (closes issue ASTERISK-18841)
+	  Reported by: Richard Miller Patches: chan_dahdi.diff (license
+	  #5685) patch uploaded by Richard Miller (Modified by me)
+	  sig_analog.c.diff (license #5685) patch uploaded by Richard
+	  Miller (Modified by me) sig_analog.h.diff (license #5685) patch
+	  uploaded by Richard Miller
+
+2011-12-05 14:56 +0000 [r346954]  Jonathan Rose <jrose at digium.com>
+
+	* main/pbx.c: Resolve duplicate label used in multiple priorities
+	  for the same extension. Prior to this patch, if labels with the
+	  same name were used for different priorities in the same
+	  extension, the new label would be accepted, but it would be
+	  unusable since attempts to reach that label would just go to the
+	  first one. Now pbx.c detects this, generates a warning in logs,
+	  and culls the label before adding it to the dialplan. (closes
+	  issue ASTERISK-18807) Reported by: Kenneth Shumard Patches:
+	  pbx.c.patch uploaded by Kenneth Shumard (License 5077)
+
+2011-12-05 14:45 +0000 [r346951]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_jabber.exports.in: Fix chan_jingle/gtalk load regression
+	  introduced in r346087 Add missing symbol exports for
+	  ast_aji_client_destroy and ast_aji_buddy_destroy for usage
+	  outside res_jabber. Testing of these changes focused on
+	  res_jabber itself, so this problem was missed. Reported-by:
+	  Michael Spiceland
+
+2011-12-04 09:57 +0000 [r346899]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and
+	  domain ACL bypass. The code that allowed admins to create users
+	  with domain-only uri's had stopped to work in 1.8 because of the
+	  reqresp parser rewrites. This is fixed now: if you have a
+	  [mydomain.com] sip user, you can register with useraddr
+	  sip:mydomain.com. Note that in that case -- if you're using
+	  domain ACLs (a configured domain list) -- mydomain.com must be in
+	  the allow list as well. Reviewboard r1606 shows a list of
+	  registration combinations and which SIP response codes are
+	  returned. Review: https://reviewboard.asterisk.org/r/1533/
+	  Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes
+	  issue ASTERISK-18741)
+
+2011-12-02 16:19 +0000 [r346762]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/chan_ooh323.c, channels/chan_h323.c: process null frame
+	  pointer returned by ast_rtp_instance_read correctly (closes issue
+	  ASTERISK-16697) Reported by: under Patches: segfault.diff
+	  (License #5871) patch uploaded by under
+
+2011-12-01 21:11 +0000 [r346700]  Richard Mudgett <rmudgett at digium.com>
+
+	* configs/res_stun_monitor.conf.sample, include/asterisk/stun.h,
+	  main/stun.c, res/res_stun_monitor.c: Re-resolve the STUN address
+	  if a STUN poll fails for res_stun_monitor. The STUN socket must
+	  remain open between polls or the external address seen by the
+	  STUN server is likely to change. However, if the STUN request
+	  poll fails then the STUN server address needs to be re-resolved
+	  and the STUN socket needs to be closed and reopened. * Re-resolve
+	  the STUN server address and create a new socket if the STUN
+	  request poll fails. * Fix ast_stun_request() return value
+	  consistency. * Fix ast_stun_request() to check the received
+	  packet for expected message type and transaction ID. * Fix
+	  ast_stun_request() to read packets until timeout or an associated
+	  response packet is found. The stun_purge_socket() hack is no
+	  longer required. * Reduce ast_stun_request() error messages to
+	  debug output. * No longer pass in the destination address to
+	  ast_stun_request() if the socket is already bound or connected to
+	  the destination. (closes issue ASTERISK-18327) Reported by:
+	  Wolfram Joost Tested by: rmudgett Review:
+	  https://reviewboard.asterisk.org/r/1595/
+
+2011-12-01 20:36 +0000 [r346564-346697]  Jonathan Rose <jrose at digium.com>
+
+	* channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180
+	  ringing. 183 Ringing isn't even a thing. 183 is actually a
+	  session progress message. (closes issue ASTERISK-18925) Reported
+	  by: Sebastian Denz Tested by: jrose Patches:
+	  asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian
+	  Denz (License #6139)
+
+	* include/asterisk/tcptls.h, main/tcptls.c, channels/chan_sip.c:
+	  r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) |
+	  18 lines Cleaning up chan_sip/tcptls file descriptor closing.
+	  This patch attempts to eliminate various possible instances of
+	  undefined behavior caused by invoking close/fclose in situations
+	  where fclose may have already been issued on a
+	  tcptls_session_instance and/or closing file descriptors that
+	  don't have a valid index for fd (-1). Thanks for more than a
+	  little help from wdoekes. (closes issue ASTERISK-18700) Reported
+	  by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane
+	  Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas
+	  Review: https://reviewboard.asterisk.org/r/1576/
+
+2011-11-30 19:36 +0000 [r346472]  Leif Madsen <lmadsen at digium.com>
+
+	* configs/queues.conf.sample: Update queues.conf.sample
+	  documentation. Update the documentation surrounding the use of
+	  MONITOR_EXEC to make it more clear that it can be used for both
+	  Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413)
+	  Reported by: David Woolley Patches:
+	  issue18817_mixmonitor_queues_doc.diff by Michael L. Young
+	  (License #5026)
+
+2011-11-28 14:30 +0000 [r346292]  Stefan Schmidt <sst at sil.at>
+
+	* res/res_rtp_asterisk.c: Fix regression that 'rtp/rtcp set debup
+	  ip' only works when also a port was specified. (closes issue
+	  ASTERISK-18693) Reported by: Davide Dal Fra Review:
+	  https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter
+	  Doekes
+
+2011-11-23 22:52 +0000 [r346239]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_iax2.c, include/asterisk/acl.h,
+	  channels/chan_skinny.c, channels/chan_h323.c, main/acl.c: Fix
+	  calls to ast_get_ip() not initializing the address family.
+
+2011-11-23 20:15 +0000 [r346144-346147]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text()
+	  function. In r116240, get_msg_text() got an extra parameter to
+	  fix the unwanted addition of trailing newlines to SIP MESSAGE
+	  bodies. This caused all linefeeds to be trimmed, which isn't
+	  right either. This is a stop-gap; the right fix is to return the
+	  original SIP request body. Review:
+	  https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan
+
+	* include/asterisk/strings.h: Fix ast_str_truncate signedness
+	  warning and documentation. Review:
+	  https://reviewboard.asterisk.org/r/1594
+
+2011-11-23 17:12 +0000 [r346086]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/chan_gtalk.c, res/res_jabber.c, channels/chan_jingle.c,
+	  include/asterisk/jabber.h: Fix res_jabber resource leaks This
+	  should fix almost all resource leaks in res_jabber that involve
+	  ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where
+	  ast_aji_get_client would sometimes bump an object's refcount and
+	  sometimes not. Review: https://reviewboard.asterisk.org/r/1553
+
+2011-11-23 16:09 +0000 [r346030]  Terry Wilson <twilson at digium.com>
+
+	* res/res_musiconhold.c: Resume playing existing hold music for
+	  cached realtime MOH As a result of the fix for ASTERISK-18039,
+	  realtime caching MOH no longer properly resumes playing back a
+	  file between different holds in the same call. This is because
+	  scanning for new files causes the existing file array to be
+	  emptied and we were just comparing that the saved pointer to the
+	  filename matched the pointer to the filename in a particular
+	  position in the array. An easy fix is to save the filename
+	  instead of a pointer to it and then do a strcmp instead of
+	  comparing the addresses. (closes issue ASTERISK-18912) Review:
+	  https://reviewboard.asterisk.org/r/1596/
+
+2011-11-22 22:55 +0000 [r345976]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/dnsmgr.h, main/dnsmgr.c: Fix dnsmgr entries to
+	  ask for the same address family each time. The dnsmgr refresh
+	  would always get the first address found regardless of the
+	  original address family requested. So if you asked for only IPv4
+	  addresses originally, you might get an IPv6 address on refresh. *
+	  Saved the original address family requested by
+	  ast_dnsmgr_lookup() to be used when the address is refreshed.
+
+2011-11-22 20:29 +0000 [r345923]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* include/asterisk/logger.h: Clarify why the AST_LOG_* macros exist
+	  next to the LOG_* macros. (issue ASTERISK-17973)
+
+2011-11-21 21:03 +0000 [r345828-345829]  Terry Wilson <twilson at digium.com>
+
+	* CHANGES: Change nat=yes to nat=force_rport in CHANGES Fix a small
+	  documentation merge issue ASTERISK-18862
+
+	* configs/sip.conf.sample, CHANGES, /, channels/chan_sip.c: Default
+	  to nat=yes; warn when nat in general and peer differ It is
+	  possible to enumerate SIP usernames when the general and
+	  user/peer nat settings differ in whether to respond to the port a
+	  request is sent from or the port listed for responses in the Via
+	  header. In 1.4 and 1.6.2, this would mean if one setting was
+	  nat=yes or nat=route and the other was either nat=no or
+	  nat=never. In 1.8 and 10, this would mean when one was
+	  nat=force_rport and the other was nat=no. In order to address
+	  this problem, it was decided to switch the default behavior to
+	  nat=yes/force_rport as it is the most commonly used option and to
+	  strongly discourage setting nat per-peer/user when at all
+	  possible. For more discussion of the issue, please see:
+	  http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
+	  (closes issue ASTERISK-18862) Review:
+	  https://reviewboard.asterisk.org/r/1591/ ........ Merged
+	  revisions 345776 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged
+	  revisions 345800 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+
+2011-11-19 15:08 +0000 [r345682]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* main/db.c: Update the documentation to better clarify how the
+	  existing commands work. Review:
+	  https://reviewboard.asterisk.org/r/1593/
+
+2011-11-17 17:06 +0000 [r345546]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.c: Remove dead code since pri_grab() can never
+	  fail. Dead code makes programmers sick. I am sick of looking at
+	  it.
+
+2011-11-17 17:04 +0000 [r345545]  Jason Parker <jparker at digium.com>
+
+	* apps/app_confbridge.c: Fix documentation of 's' option. The menu
+	  key is #, not *. Reported by p3nguin on #asterisk.
+
+2011-11-16 14:42 +0000 [r345487]  Jonathan Rose <jrose at digium.com>
+
+	* apps/app_voicemail.c: Guarantee messages go into the right
+	  folders with multiple recipients Before, using the U flag in
+	  Voicemail with multiple recipients would put urgent messages in
+	  the INBOX folder for all users past the first thanks to a bug
+	  with the message copying function. This would also cause messages
+	  to fail to be sent if the INBOX directory hadn't been created for
+	  that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt
+	  Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan
+	  Review: https://reviewboard.asterisk.org/r/1589/
+
+2011-11-15 20:09 +0000 [r345219-345431]  Richard Mudgett <rmudgett at digium.com>
+
+	* res/res_agi.c: Make FastAGI HANGUP show up in AGI debug output. *
+	  Change from using send() to ast_agi_send() so the HANGUP shows up
+	  in the AGI debug output. (closes issue ASTERISK-18723) Reported
+	  by: James Van Vleet Patches: jira_asterisk_18723_v1.8.patch
+	  (license #5621) patch uploaded by rmudgett
+
+	* channels/sig_pri.c: Fix typo in sig_pri using wrong structure
+	  name. It is fortunate that the typo does not alter generated code
+	  since the e->restart.channel and e->ring.channel members are in
+	  the same position. (closes issue ASTERISK-18868) Reported by:
+	  zvision Patches: sig_pri.c.diff (License #5755) patch uploaded by
+	  zvision
+
+	* apps/app_queue.c: Make queue log indicate if ADDMEMBER is paused
+	  for AMI and realtime. * Add parameter to queue log ADDMEMBER to
+	  indicate if the member is paused. (closes issue ASTERISK-18645)
+	  Reported by: garlew Patches: paused.diff (License #5337) patch
+	  uploaded by garlew Tested by: rmudgett, garlew Review:
+	  https://reviewboard.asterisk.org/r/1469/
+
+	* UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h,
+	  channels/chan_sip.c: Restore SIP DTMF overlap dialing method. The
+	  recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
+	  working correctly removed a long standing ability to do overlap
+	  dialing using DTMF in the early media phase of a call. See
+	  ASTERISK-18702 it has a very good description of the issue. I
+	  started with Pavel Troller's chan_sip.diff patch on issue
+	  ASTERISK-18702. * Added 'dtmf' enum value to sip.conf
+	  allowoverlap config option. The new option value causes the
+	  Incomplte application to not send anything with chan_sip so the
+	  caller can supply more digits via DTMF. * Renames
+	  SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
+	  since that is what it really means. * Fixed get_destination()
+	  inconsistency with the pickup extension matching. * Fixed
+	  initialization of PAGE3 of global_flags in reload_config().
+	  (closes issue ASTERISK-18702) Reported by: Pavel Troller Review:
+	  https://reviewboard.asterisk.org/r/1517/ Review:
+	  https://reviewboard.asterisk.org/r/1582/
+
+	* main/pbx.c: Fix Progress spelling error in main/pbx.c. (closes
+	  issue ASTERISK-18857) Reported by: David M Patches:
+	  mainpbx-trivial.patch (License #6326) patch uploaded by David M
+
+2011-11-14 19:05 +0000 [r345163]  Terry Wilson <twilson at digium.com>
+
+	* main/channel.c: Don't read past end of input when calling write()
+	  int blah = 1; ... write(chan->alertpipe[1], &blah, new_frames *
+	  sizeof(blah)) != (new_frames * sizeof(blah))) is only valid when
+	  new_frames == 1. Otherwise we start reading into adjacent
+	  variables declared on the stack. The read end discards what is
+	  read, so the values don't matter but it's not a good idea to read
+	  past where we want even though new_frames is almost always 1 and
+	  should never be large. This patch is basically taken out of
+	  kpfleming's eventfd branch, as he mentioned that he remembered
+	  fixing it there when I talked to him about this issue. Review:
+	  https://reviewboard.asterisk.org/r/1583/
+
+2011-11-14 19:00 +0000 [r345160]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* channels/sip/include/reqresp_parser.h: Update reqresp_parser
+	  parse_uri doxygen comments. The issue mentioned in the bug report
+	  had been fixed recently by twilson. The reporter included this
+	  documentation fix. (closes issue ASTERISK-18572) Reported by:
+	  Richard Miller Patch by: Richard Miller (modified)
+
+2011-11-14 15:08 +0000 [r345063]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/chan_sip.c: Ensure that a null vmexten does not cause a
+	  segfault When sip_send_mwi_to_peer was modified recently to avoid
+	  deadlocks, vmexten was not expected to be null. This change
+	  handles that situation to avoid a segfault.
+
+2011-11-14 15:00 +0000 [r345062]  Jonathan Rose <jrose at digium.com>
+
+	* apps/app_voicemail.c: Moves voicemail setup password entry to the
+	  end of the setup process. This change was made because
+	  forcegreeting and forcename settings in voicemail could be
+	  circumvented by hanging up after entering a password, because the
+	  only way voicemail currently observes whether a mailbox is new or
+	  not is by checking to see if the password is the same as the
+	  mailbox number or not. (closes issue ASTERISK-18282) Reported by:
+	  Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/
+
+2011-11-12 16:05 +0000 [r344965]  Gregory Nietsky <gregory at distrotech.co.za>
+
+	* channels/chan_misdn.c: mISDN Round Robin break when no channel is
+	  available Prevent channels been parsed repetitively.
+
+2011-11-12 00:24 +0000 [r344899]  Terry Wilson <twilson at digium.com>
+
+	* res/res_musiconhold.c: Don't forget to rescan MOH files for
+	  cached realtime classes Realtime MOH class caching was
+	  implemented because without it, you would build a completely new
+	  MOH class and would start the music over at the beginning each
+	  time hold was pressed in a conversation. Unfortunately, this
+	  broke re-scanning for file changes for realtime MOH classes. This
+	  patch corrects that issue. (closes issue ASTERISK-18039) Review:
+	  https://reviewboard.asterisk.org/r/1579/
+
+2011-11-11 21:54 +0000 [r344835-344843]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* main/utils.c, include/asterisk/stringfields.h,
+	  include/asterisk/utils.h: Use __alignof__ instead of sizeof for
+	  stringfield length storage. Kevin P Fleming suggested that
+	  r343157 should use __alignof__ instead of sizeof. For most
+	  systems this won't be an issue, but better fix it now while it's
+	  still fresh. Review: https://reviewboard.asterisk.org/r/1573
+
+	* channels/sip/reqresp_parser.c: Remove unneeded if(params) checks
+	  in reqresp_parser. Nick Lewis added them in
+	  https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent
+	  reason. There is no way that params could become NULL in that
+	  piece of code, so I removed these excess checks again.
+
+	* main/manager.c: Fix bad quoting of multiline mxml opaque_data
+	  that caused invalid xml. The opaque_data was added and enclosed
+	  in single quotes, assuming it would be only a single line. The
+	  rest of the lines were appended after the closing quote. (closes
+	  issue ASTERISK-18852) Reported by: peep_ on IRC Review:
+	  https://reviewboard.asterisk.org/r/1577
+
+2011-11-11 20:42 +0000 [r344823]  Matthew Jordan <mjordan at digium.com>
+
+	* main/file.c: Video format was treated as audio when removed from
+	  the file playback scheduler This patch fixes the format type
+	  check in ast_closestream and filestream_destructor. Previously a
+	  comparison operator was used, but since audio formats are no
+	  longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats
+	  that have a value greater than the video formats), a bitwise AND
+	  operation is used instead. Duplicated code was also moved to
+	  filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo
+	  Bedrij Tested by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/1580/
+
+2011-11-11 20:10 +0000 [r344769]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/chan_sip.c: Fix regression introduced by SDP fixups If
+	  capability is adjusted when switching to UDPTL during fax
+	  transmission, fax teardown fails. Make sure capability is only
+	  touched if RTP is active. This regression was introduced in

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