[svn-commits] bebuild: tag 1.8.14.0-rc1 r368702 - /tags/1.8.14.0-rc1/

SVN commits to the Digium repositories svn-commits at lists.digium.com
Fri Jun 8 10:24:35 CDT 2012


Author: bebuild
Date: Fri Jun  8 10:24:32 2012
New Revision: 368702

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=368702
Log:
Importing files for 1.8.14.0-rc1 release.

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+2012-06-08  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.14.0-rc1 Released.
+
+2012-06-06 21:27 +0000 [r368644]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, channels/sig_analog.c: Fix POTS flash hook
+	  to orignate a second call deadlock. A deadlock can occur when a
+	  POTS phone tries to flash hook to originate a second call for
+	  3-way or transfer. If another process is scanning the channels
+	  container when the POTS line flash hooks then a deadlock will
+	  occur. * Release the channel and private locks when creating a
+	  new channel as a result of a flash hook. (closes issue
+	  ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett
+
+2012-06-06 19:13 +0000 [r368625]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Fix a specific scenario where ACKs are not
+	  matched. If a dialog-starting INVITE contains a to-tag, then
+	  Asterisk will respond with a 481. In this case, the resulting
+	  incoming ACK would not be matched, so Asterisk would continue
+	  retransmitting the 481 until the transaction times out. There
+	  were two issues. Asterisk, upon creating a sip_pvt would generate
+	  a local tag. However, when the time came to transmit the 481,
+	  since there was a to-tag in the INVITE, Asterisk would place this
+	  original to-tag in the 481 response. When the ACK came in,
+	  Asterisk would attempt to match the to-tag in the ACK to the
+	  generated local tag. Unfortunately, Asterisk never actually
+	  transmitted a response with the generated local tag, so the
+	  to-tag in the ACK would not match. The other problem was that
+	  when the 481 was sent, nothing was set on the sip_pvt to indicate
+	  what CSeq is expected in the ACK. To fix the first problem, we
+	  zero out the to-tag seen in the incoming INVITE. This way,
+	  Asterisk, when time to send a response, will send its generated
+	  local tag instead. To fix the second problem, we set the
+	  sip_pvt's pendinginvite to the CSeq of the INVITE when we send a
+	  481. (closes issue ASTERISK-19892) Reported by Mark Michelson
+
+2012-06-06 17:20 +0000 [r368604]  Matthew Jordan <mjordan at digium.com>
+
+	* build_tools/make_version: Add feature modifier to versions
+	  produced from branches Certain branches, such as Certified
+	  Asterisk, may have a modifier added to them that specifies the
+	  features available in that branch. For branches, this modifier is
+	  expected to be reflected in the location of the branch in
+	  subversion. For example, a subversion of URL of
+	  /certified/branches/1.8.11 would have a feature modifier of
+	  'certified'. This is slightly different then how features are
+	  determined for tags, where the feature is part of the actual tag
+	  name, e.g., "10.5.0-digiumphones". In keeping with the
+	  nomenclature used for tags, the feature specifier for branches is
+	  translated and placed after the revision numbers. For the example
+	  given previously, this would result in a branch version of
+	  "Asterisk SVN-branch-1.8.11-cert-rXXXXXX".
+
+2012-06-06 16:07 +0000 [r368586]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/chan_sip.c: Ensure overlapping hold flags do not
+	  conflict When changing between different modes of hold, the flags
+	  were not being cleared out properly causing a failure to change
+	  hold states. (closes issue ASTERISK-19919) Patch-by: Morten
+	  Tryfoss Reported-by: Morten Tryfoss
+
+2012-06-06 01:08 +0000 [r368567]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/features.c: Fix parked call performing a DTMF blind transfer
+	  after being retrieved. When a parked call was retrieved from the
+	  parking lot, it could not do a blind transfer because it caused
+	  the involved calls to be hung up unconditionally. * Made the
+	  ParkedCall application return the ast_bridge_call() return value.
+	  (closes issue ABE-2862) Reported by: Vlad Povorozniuc
+
+2012-06-05 15:26 +0000 [r368520-368533]  Kinsey Moore <kmoore at digium.com>
+
+	* apps/app_minivm.c: Resolve some build warnings My newly upgraded
+	  compiler caught these usages of uninitialized values. They
+	  weren't actually used.
+
+	* apps/app_voicemail.c: Ensure that pages and emails are sent using
+	  RFC822-compliant date format When localization was added to
+	  app_voicemail, these headers were altered when they should have
+	  remained in en_US format for RFC compliance. This reverts the
+	  changes to those two lines. (closes issue ASTERISK-19876)
+
+2012-06-04 21:56 +0000 [r368498]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Relay proper SIP responses on calling side.
+	  Revision 351130 broke corect HANGUPCAUSE setting for the 404 case
+	  in chan_sip. Other cases were also potentially broken. This patch
+	  fixes the relaying of causes to be what they used to be. (closes
+	  issue ASTERISK-19914) Reported by Pavel Troller Tested by Walter
+	  Doekes (via a reviewboard test to be committed later) Patches:
+	  chan_sip.diff uploaded by Pavel Troller (license #6302)
+
+2012-06-04 21:10 +0000 [r368405-368469]  Richard Mudgett <rmudgett at digium.com>
+
+	* UPGRADE.txt: Document BLINDTRANSFER behavior change. (issue
+	  ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call
+
+	* main/channel.c: Fix potential deadlock between masquerade and
+	  chan_local. * Restructure ast_do_masquerade() to not hold channel
+	  locks while it calls ast_indicate(). * Simplify many calls to
+	  ast_do_masquerade() since it will never return a failure now. If
+	  it does fail internally because a channel driver callback
+	  operation failed, the only thing ast_do_masquerade() can do is
+	  generate a warning message about strange things may happen and
+	  press on. * Fixed the call to ast_bridged_channel() in
+	  ast_do_masquerade(). This change fixes half of the deadlock
+	  reported in ASTERISK-19801 between masquerades and chan_iax.
+	  (closes issue ASTERISK-19537) Reported by: rmudgett Tested by:
+	  rmudgett Review: https://reviewboard.asterisk.org/r/1915/
+
+2012-06-01 23:21 +0000 [r368308]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_stack.c: Fix deadlock when Gosub used with alternate
+	  dialplan switches. Attempting to remove a channel from
+	  autoservice with the channel lock held will result in deadlock. *
+	  Restructured gosub_exec() to not call ast_parseable_goto() and
+	  ast_exists_extension() with the channel lock held. (closes issue
+	  ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett
+
+2012-06-01 18:18 +0000 [r368218]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_sip.c: Improve SDP parsing warning messages *
+	  'Unsupported media type' is only reported when that is in fact
+	  the case, not when a supported media type is included in an 'm'
+	  line that has an invalid format. * All warning messages related
+	  to parsing 'm' lines now include the 'm' line contents. * (minor
+	  bugfix) newline added to port-number-zero warning messages. *
+	  Warning messages improved to use RFC-specified terminology for
+	  various items. * Warnings for offers that include more than one
+	  port for a single media type now include the media type. Review:
+	  https://reviewboard.asterisk.org/r/1811/
+
+2012-06-01 03:25 +0000 [r368092]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* funcs/func_channel.c: Add documentation to function CHANNEL for
+	  options echocan_mode and buffers The ability to set
+	  "echocan_mode" and "buffers" through the dialplan was added to
+	  chan_dahdi some time ago. This patch adds some documentation to
+	  func_channel. (Closes issue ASTERISK-19911) Reported by: Dale
+	  Noll Tested by: Michael L. Young Patches:
+	  asterisk-19911-branch18.diff uploaded by Michael L. Young
+	  (license 5026) Review: https://reviewboard.asterisk.org/r/1949/
+
+2012-05-31 18:00 +0000 [r367906-368039]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/db1-ast/btree/bt_open.c, apps/app_queue.c,
+	  channels/chan_iax2.c, pbx/pbx_config.c, res/ael/pval.c,
+	  main/tcptls.c, main/manager.c, res/res_config_odbc.c,
+	  channels/chan_sip.c, channels/chan_agent.c, funcs/func_math.c,
+	  main/features.c: Coverity Report: Fix issues for error type
+	  REVERSE_INULL (core modules) * Fixes findings:
+	  0-2,5,7-15,24-26,28-31 (issue ASTERISK-19648) Reported by: Matt
+	  Jordan
+
+	* channels/sig_pri.c, channels/sig_ss7.c: Use the
+	  DEADLOCK_AVOIDANCE() macro instead. (issue ASTERISK-19854)
+
+	* channels/sig_pri.c, channels/sig_ss7.c: Fix deadlock when
+	  executing CLI "pri show channels" and "ss7 show channels"
+	  commands. * Fix sig_pri_lock_owner() to avoid deadlock properly.
+	  * Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid
+	  deadlock properly. * Code ss7_grab() better. (closes issue
+	  ASTERISK-19854) Reported by: Jaxon Patches:
+	  jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded
+	  by rmudgett (Modified to do the same thing to sig_ss7) Tested by:
+	  Jaxon
+
+	* apps/app_meetme.c: Coverity Report: Fix issues for error type
+	  REVERSE_INULL (deprecated modules) * Fix only issue pointed out
+	  by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user().
+	  * Change use of %i to %d in sscanf() in find_user(). The use of
+	  %i gives unexpected parsing because it can accept hex, octal, and
+	  decimal integer formats. * Changed other uses of %i in
+	  app_meetme() to use %d for consistency. (issue ASTERISK-19648)
+	  Reported by: Matt Jordan
+
+2012-05-29 18:30 +0000 [r367843]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/chan_skinny.c: AST-2012-008: Fix remote crash
+	  vulnerability in chan_skinny When a skinny session is
+	  unregistered, the corresponding device pointer is set to NULL in
+	  the channel private data. If the client was not in the on-hook
+	  state at the time the connection was closed, the device pointer
+	  can later be dereferenced if a message or channel event attempts
+	  to use a line's pointer to said device. The patches prevent this
+	  from occurring by checking the line's pointer in message handlers
+	  and channel callbacks that can fire after an unregistration
+	  attempt. (closes issue ASTERISK-19905) Reported by: Christoph
+	  Hebeisen Tested by: mjordan, Damien Wedhorn Patches:
+	  AST-2012-008-1.8.diff uploaded by mjordan (license 6283)
+	  AST-2012-008-10.diff uploaded by mjordan (license 6283)
+
+2012-05-25 16:28 +0000 [r367781]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_iax2.c: AST-2012-007: Fix IAX receiving HOLD
+	  without suggested MOH class crash. * Made schedule_delivery() set
+	  the received frame f->data.ptr to NULL if the datalen is zero. *
+	  Fix queue_signalling() memcpy() size error. * Made
+	  queue_signalling() not use C++ keyword variable names. (closes
+	  issue ASTERISK-19597) Reported by: mgrobecker Patches:
+	  jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by
+	  rmudgett Tested by: rmudgett, Michael L. Young
+
+2012-05-25 02:27 +0000 [r367730]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* channels/chan_sip.c: Fix pvt_sip for inbound call to use peer's
+	  allowtransfer setting The pvt_sip allowtransfer was not being set
+	  to that of the peer's setting. Therefore, the global
+	  allowtransfer setting was being used instead which would lead to
+	  calls not being transfered if the global setting was set to 'no'
+	  despite the setting on the peer being 'yes' and vice versa, calls
+	  would be allowed to transfer even if the peer's setting was 'no'
+	  but the global setting was 'yes'. (Closes issue ASTERISK-19856)
+	  Reported by: Jacek Tested by: Michael L. Young, Jacek Patches:
+	  issue-asterisk-19856-branch10-v3.diff uploaded by Michael L.
+	  Young (license 5026) Review:
+	  https://reviewboard.asterisk.org/r/1923/
+
+2012-05-24 22:21 +0000 [r367469-367678]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_queue.c, apps/app_dial.c: Fix Dial I option ignored if
+	  dial forked and one fork redirects. The Dial and Queue I option
+	  is intended to block connected line updates and redirecting
+	  updates. However, it is a feature that when a call is locally
+	  redirected, the I option is disabled if the redirected call runs
+	  as a local channel so the administrator can have an opportunity
+	  to setup new connected line information. Unfortunately, the Dial
+	  and Queue I option is disabled for *all* forked calls if one of
+	  those calls is redirected. * Make the Dial and Queue I option
+	  apply to each outgoing call leg independently. Now if one
+	  outgoing call leg is locally redirected, the other outgoing calls
+	  are not affected. * Made Dial not pass any redirecting updates
+	  when forking calls. Redirecting updates do not make sense for
+	  this scenario. * Made Queue not pass any redirecting updates when
+	  using the ringall strategy. Redirecting updates do not make sense
+	  for this scenario. * Fixed deadlock potential with chan_local
+	  when Dial and Queue send redirecting updates for a local
+	  redirect. * Converted the Queue stillgoing flag to a boolean
+	  bitfield. (closes issue ASTERISK-19511) Reported by: rmudgett
+	  Tested by: rmudgett Review:
+	  https://reviewboard.asterisk.org/r/1920/
+
+	* main/pbx.c: Fix WaitExten(x,m(musicclass)) string termination.
+	  The AST_CONTROL_HOLD MOH class from the WaitExten application can
+	  now be queued onto a channel, passed over local channels with the
+	  /m option, and passed over IAX channels.
+
+2012-05-23 20:27 +0000 [r367416]  Mark Michelson <mmichelson at digium.com>
+
+	* main/tcptls.c: Only call SSL_CTX_free if DO_SSL is defined.
+	  Thanks to Paul Belanger for pointing out this error.
+
+2012-05-23 13:06 +0000 [r367362]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/chan_sip.c: Update a peer's LastMsgsSent when the peer
+	  is notified of waiting messages Previously, MWI logic utilized a
+	  counter called 'lastmsgssent' to know whether or not MWI NOTIFY
+	  requests had been sent to a specific peer. When MWI notifications
+	  were changed to use the internal event framework, this value was
+	  no longer needed for its original purpose. Hence, it was no
+	  longer updated with the new/old message counts for a peer.
+	  However, the value was still presented when, either by AMI or
+	  CLI, a 'sip show peer [peer]' command was executed. The output of
+	  the command would always display the erroneous value of
+	  32767/65535 for 'LastMsgsSent'. This patch makes it so that the
+	  value of lastmsgssent is updated appropriately. The value should
+	  now display the new/old message counts for a particular peer.
+	  (closes issue ASTERISK-17866) Reported by: Steve Davies patches
+	  by: ast-17866-rb1272.patch (License #5041 by irroot) Modified
+	  slightly for this commit Review:
+	  https://reviewboard.asterisk.org/r/1939
+
+2012-05-22 17:14 +0000 [r367266-367292]  Terry Wilson <twilson at digium.com>
+
+	* include/asterisk/channel.h, main/cel.c, main/asterisk.c,
+	  main/channel.c, include/asterisk/cel.h: Fix race condition for
+	  CEL LINKEDID_END event This patch fixes to situations that could
+	  cause the CEL LINKEDID_END event to be missed. 1) During a core
+	  stop gracefully, modules are unloaded when ast_active_channels ==
+	  0. The LINKDEDID_END event fires during the channel destructor.
+	  This means that occasionally, the cel_* module will be unloaded
+	  before the channel is destroyed. It seemed generally useful to
+	  wait until the refcount of all channels == 0 before unloading, so
+	  I added a channel counter and used it in the shutdown code. 2)
+	  During a masquerade, ast_channel_change_linkedid is called. It
+	  calls ast_cel_check_retire_linkedid which unrefs the linkedid in
+	  the linkedids container in cel.c. It didn't ref the new linkedid.
+	  Now it does. Review: https://reviewboard.asterisk.org/r/1900/
+
+	* channels/chan_sip.c: Resolve crash in subscribing for MWI
+	  notifications ASTOBJ_UNREF sets the variable to NULL after
+	  unreffing it, so the variable should definitely not be used after
+	  that. To solve this in the two cases that affect subscribing for
+	  MWI notifications, we instead save the ref locally, and unref
+	  them in the error conditions. (closes issue ASTERISK-19827)
+	  Reported by: B. R Review:
+	  https://reviewboard.asterisk.org/r/1940/
+
+2012-05-18 17:47 +0000 [r367002-367027]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_dahdi.c, main/say.c: Address MISSING_BREAK static
+	  analysis reports some more. This addresses core findings 4 and 6.
+	  Moises Silva helped me by stating that a break could be safely
+	  added to the case where it is added in chan_dahdi.c In say.c, I
+	  have added a comment indicating that static analysis complains
+	  but that it is currently unknown if this is correct. This fixes
+	  all core findings of this type. (closes issue ASTERISK-19662)
+	  reported by Matthew Jordan
+
+	* include/asterisk/tcptls.h, main/tcptls.c, channels/chan_sip.c:
+	  Fix memory leak of SSL_CTX structures in TLS core. SSL_CTX
+	  structures were allocated but never freed. This was a bigger
+	  issue for clients than servers since new SSL_CTX structures could
+	  be allocated for each connection. Servers, on the other hand,
+	  typically set up a single SSL_CTX for their lifetime. This is
+	  solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an
+	  ssl_ctx on it, it is freed so that a new one can take its place.
+	  2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
+	  been added so that servers can properly free their SSL_CTXs.
+	  (issue ASTERISK-19278)
+
+2012-05-18 15:42 +0000 [r366944]  Matthew Jordan <mjordan at digium.com>
+
+	* main/cli.c, channels/chan_sip.c, funcs/func_odbc.c: Fix more
+	  memory leaks This patch adds to what was fixed in r366880.
+	  Specifically, it addresses the following: * chan_sip: dispose of
+	  an allocated frame in off nominal code paths in sip_rtp_read *
+	  func_odbc: when disposing of an allocated resultset, ensure that
+	  any rows that were appended to that resultset are also disposed
+	  of * cli: free the created return string buffer in another off
+	  nominal code path (issue ASTERISK-19665) Reported by: Matt Jordan
+	  Review: https://reviewboard.asterisk.org/r/1922/
+
+2012-05-18 14:16 +0000 [r366882]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/sip/config_parser.c: Reorder and renumber tests
+	  appropriately It appears that a patch did not apply properly when
+	  adding tests 12 and 13 and test 11 was duplicated. These tests
+	  have been reordered and renumbered such that they make sense.
+
+2012-05-18 13:58 +0000 [r366880]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_calendar_caldav.c, res/res_musiconhold.c,
+	  res/res_jabber.c, apps/app_queue.c, channels/chan_iax2.c,
+	  main/enum.c, main/editline/term.c, main/config.c, res/res_srtp.c,
+	  main/editline/tokenizer.c, main/cli.c, channels/chan_dahdi.c,
+	  main/data.c, funcs/func_odbc.c, apps/app_minivm.c,
+	  main/features.c, main/editline/readline.c,
+	  channels/sip/config_parser.c, main/xmldoc.c, res/res_calendar.c,
+	  apps/app_voicemail.c, res/res_rtp_asterisk.c, main/netsock2.c,
+	  res/res_calendar_icalendar.c, res/res_calendar_exchange.c,
+	  main/pbx.c, apps/app_page.c, channels/chan_sip.c,
+	  funcs/func_dialgroup.c, apps/app_record.c: Fix a variety of
+	  memory leaks This patch addresses a number of memory leaks in a
+	  variety of modules that were found by a static analysis tool. A
+	  brief summary of the changes: * app_minivm: free ast_str objects
+	  on off nominal paths * app_page: free the ast_dial object if the
+	  requested channel technology cannot be appended to the dialing
+	  structure * app_queue: if a penalty rule failed to match any
+	  existing rule list names, the created rule would not be inserted
+	  and its memory would be leaked * app_read: dispose of the created
+	  silence detector in the presence of off nominal circumstances *
+	  app_voicemail: dispose of an allocated unique ID field for MWI
+	  event un-subscribe requests in off nominal paths; dispose of
+	  configuration objects when using the secret.conf option *
+	  chan_dahdi: dispose of the allocated frame produced by
+	  ast_dsp_process * chan_iax2: properly unref peer in CLI command
+	  "iax2 unregister" * chan_sip: dispose of the allocated frame
+	  produced by sip_rtp_read's call of ast_dsp_process; free memory
+	  in parse unit tests * func_dialgroup: properly deref ao2 object
+	  grhead in nominal path of dialgroup_read * func_odbc: free
+	  resultset in off nominal paths of odbc_read * cli: free
+	  match_list in off nominal paths of CLI match completion * config:
+	  free comment_buffer/list_buffer when configuration file load is
+	  unchanged; free the same buffers any time they were created and
+	  config files were processed * data: free XML nodes in various
+	  places * enum: free context buffer in off nominal paths *
+	  features: free ast_call_feature in off nominal paths of
+	  applicationmap config processing * netsock2: users of
+	  ast_sockaddr_resolve pass in an ast_sockaddr struct that is
+	  allocated by the method. Failures in ast_sockaddr_resolve could
+	  result in the users of the method not knowing whether or not the
+	  buffer was allocated. The method will now not allocate the
+	  ast_sockaddr struct if it will return failure. * pbx: cleanup
+	  hash table traversals in off nominal paths; free ignore pattern
+	  buffer if it already exists for the specified context * xmldoc:
+	  cleanup various nodes when we no longer need them *
+	  main/editline: various cleanup of pointers not being freed before
+	  being assigned to other memory, cleanup along off nominal paths *
+	  menuselect/mxml: cleanup of value buffer for an attribute when
+	  that attribute did not specify a value * res_calendar*: responses
+	  are allocated via the various *_request method returns and should
+	  not be allocated in the various write_event methods; ensure
+	  attendee buffer is freed if no data exists in the parsed node;
+	  ensure that calendar objects are de-ref'd appropriately *
+	  res_jabber: free buffer in off nominal path * res_musiconhold:
+	  close the DIR* object in off nominal paths * res_rtp_asterisk: if
+	  we run out of ports, close the rtp socket object and free the rtp
+	  object * res_srtp: if we fail to create the session in libsrtp,
+	  destroy the temporary ast_srtp object (issue ASTERISK-19665)
+	  Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/1922
+
+2012-05-17 14:40 +0000 [r366791]  Jonathan Rose <jrose at digium.com>
+
+	* channels/chan_sip.c: chan_sip: Fix missed locking of opposing pvt
+	  for directmedia acl from r366547 It also required deadlock
+	  avoidance since two sip_pvts structs needed to be locked
+	  simultaneously. Trunk handles it differently, so this is a 1.8
+	  and 10 patch only. (issue AST-876)
+
+2012-05-17 12:51 +0000 [r366740]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_calendar_ews.c, channels/chan_dahdi.c: Fix checking
+	  bounds of array index after using it; improper sizeof This patch
+	  fixes two problems pointed out by a static analysis tool. * In
+	  chan_dahdi, when an event is handled the index of the sub channel
+	  is first obtained. In very off nominal cases, the method that
+	  determines the index can return a negative value. In the event
+	  handling code, whether or not the index returned is valid was
+	  being checked after that value was used to index into an array.
+	  This patch makes it so the value is checked before any indexing
+	  is done. * In res_calendar_ews, sizeof was being passed a pointer
+	  instead of the struct to determine the amount of memory to
+	  allocate. (issue ASTERISK-19651) Reported by: Matt Jordan (closes
+	  issue ASTERISK-19671) Reported by: Matt Jordan
+
+2012-05-16 15:52 +0000 [r366597-366650]  Mark Michelson <mmichelson at digium.com>
+
+	* main/http.c: Fix incorrect default port number for HTTP server.
+	  Thanks to Tzafrir Cohen for bringing this up on the Asterisk
+	  developers mailing list.
+
+	* channels/chan_sip.c: Correct misuse of ast_strip_quoted() when
+	  getting a Diversion header's reason parameter. The use here was
+	  assuming that the pointer would be updated, but the updated
+	  string is actually returned by ast_strip_quoted() instead.
+
+2012-05-15 20:14 +0000 [r366547]  Jonathan Rose <jrose at digium.com>
+
+	* channels/chan_sip.c: chan_sip: Check the right channel's host
+	  address for directmediapermit/deny Prior to this patch, when
+	  checking the addresses for directmediapermit and directmediadeny,
+	  Asterisk would check the host address of the channel permit/deny
+	  was specified, which differs from the expectations of both our
+	  users and the development team. Instead, directmediapermit/deny
+	  now checks against the address of the channel that the peer with
+	  the ACL is connected to. (issue AST-876) Review:
+	  https://reviewboard.asterisk.org/r/1899/
+
+2012-05-14 19:57 +0000 [r366389-366409]  Mark Michelson <mmichelson at digium.com>
+
+	* pbx/dundi-parser.c: Fix two more coverity constant expression
+	  result findings. These correspond to findings 0 and 1 in the core
+	  findings of ASTERISK-19649. After contacting Mark Spencer, he was
+	  unsure of what the intent behind these lines of code were, so
+	  they are being axed. For Asterisk 1.8 and 10, the output of
+	  debugging DUNDi frames will not be changed, but for trunk the
+	  "Retry" portion will be omitted since it does not properly
+	  distinguish retransmissions from initial frames. (closes issue
+	  ASTERISK-19649) Reported by Matthew Jordan
+
+	* channels/chan_sip.c: Fix broken reinvite glare scenario. To make
+	  a long story short, reinvite glares were broken because Asterisk
+	  would invert the To and From headers when ACKing a 491 response.
+	  The reason was because the initreq of the dialog was being
+	  changed to the incoming glared reinvite instead of being set to
+	  the outgoing glared reinvite. This change has three parts * In
+	  handle_incoming, we never will reject an ACK because it has a
+	  to-tag present, even if we think the request may be out of
+	  dialog. * In handle_request_invite, we do not change the initreq
+	  when receiving a reinvite to which we will respond with a 491. *
+	  In handle_request_invite, several superflous settings up
+	  pendinginvite have been removed since this is dones automatically
+	  by transmit_response_reliable Review:
+	  https://reviewboard.asterisk.org/r/1911
+
+2012-05-11 23:53 +0000 [r366296]  Russell Bryant <russell at russellbryant.com>
+
+	* addons/format_mp3.c: format_mp3: Fix a possible crash mp3_read().
+	  This patch fixes a potential crash in mp3_read() by not assuming
+	  that dbuf has enough data to finish filling up the output buffer.
+	  The patch also makes sure that the dbuf state gets reset after we
+	  know we read everything out of it already. In passing, this patch
+	  includes some other cleanups of this module, including stripping
+	  trailing whitespace, formatting fixes based on coding guidelines,
+	  and removing a number of unused members from the private state
+	  struct. (closes issue ASTERISK-19761) Reported by: Chris
+	  Maciejewsk Tested by: Chris Maciejewsk
+
+2012-05-10 23:38 +0000 [r366240]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/channel.c: * Made ast_change_name() hold the channels
+	  container lock while changing the channel name. * Eliminate
+	  redundant list not empty check in clone_variables().
+
+2012-05-10 20:50 +0000 [r366167]  Kinsey Moore <kmoore at digium.com>
+
+	* main/devicestate.c, pbx/dundi-parser.c, channels/chan_iax2.c,
+	  channels/iax2-parser.c, main/config.c, res/res_monitor.c,
+	  main/channel.c, main/cdr.c, res/ael/pval.c, main/data.c,
+	  channels/chan_dahdi.c, main/tcptls.c, main/manager.c,
+	  main/features.c, main/app.c, main/event.c, pbx/pbx_dundi.c,
+	  res/res_odbc.c, main/xmldoc.c, apps/app_voicemail.c,
+	  funcs/func_speex.c, main/pbx.c, res/res_calendar_icalendar.c,
+	  channels/chan_sip.c, funcs/func_lock.c, channels/chan_agent.c,
+	  channels/sip/reqresp_parser.c: Resolve FORWARD_NULL static
+	  analysis warnings This resolves core findings from ASTERISK-19650
+	  numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56,
+	  82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding
+	  numbers 26, 33, and 29 were already resolved. Those skipped were
+	  either extended/deprecated or in areas of code that shouldn't be
+	  disturbed. (Closes issue ASTERISK-19650)
+
+2012-05-10 16:47 +0000 [r366094]  Jonathan Rose <jrose at digium.com>
+
+	* channels/iax2-provision.c, apps/app_queue.c,
+	  channels/chan_iax2.c, res/ael/ael.flex, funcs/func_devstate.c,
+	  main/asterisk.c, main/db.c, main/xmldoc.c, apps/app_voicemail.c,
+	  main/pbx.c, channels/sig_analog.c, channels/chan_sip.c,
+	  funcs/func_lock.c, main/features.c, main/acl.c: Coverity Report:
+	  Fix issues for error type CHECKED_RETURN for core (issue
+	  ASTERISK-19658) Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/1905/
+
+2012-05-10 16:10 +0000 [r366052]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Close the proper tcptls_session when session
+	  creation fails. (issue AST-998) Reported by: Thomas Arimont
+	  Tested by: Thomas Arimont
+
+2012-05-10 15:35 +0000 [r365989-366048]  Jonathan Rose <jrose at digium.com>
+
+	* apps/app_chanspy.c, apps/app_page.c, funcs/func_cdr.c,
+	  main/features.c, apps/app_disa.c: Coverity Report: Fix issues for
+	  error type UNINIT in Core supported modules (issue
+	  ASTERISK-19652) Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/1909/
+
+	* codecs/codec_dahdi.c: Block on frameout if the hardware has
+	  enough samples to complete a frame. Fixes some problems with
+	  skipping audio in elaborate scenarios involving multiple codecs
+	  by making codec_dahdi operate in a more synchronous fashion
+	  similar to codec_g729. This change also fixes the use of file
+	  conversion tools from Asterisk's CLI. This change may cause the
+	  thread responsible for transcoding audio to block briefly (Shaun
+	  Ruffell describes this as 'several milliseconds') while waiting
+	  for the hardware transcoder. (closes issue ASTERISK-19643)
+	  reported by: Shaun Ruffell Patches:
+	  0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch
+	  uploaded by Shaun Ruffell (license 5417)
+
+2012-05-09 16:11 +0000 [r365896]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Prevent sip_pvt refleak when an ast_channel
+	  outlasts its corresponding sip_pvt. chan_sip was coded under the
+	  assumption that a SIP dialog with an owner channel will always be
+	  destroyed after the owner channel has been hung up. However,
+	  there are situations where the SIP dialog can time out and auto
+	  destruct before the corresponding channel has hung up. A typical
+	  example of this would be if the 'h' extension in the dialplan
+	  takes a long time to complete. In such cases,
+	  __sip_autodestruct() would complain about the dialog being auto
+	  destroyed with an owner channel still in place. The problem is
+	  that even once the owner channel was hung up, the sip_pvt would
+	  still be linked in its ao2_container because nothing would ever
+	  unlink it. The fix for this is that if __sip_autodestruct() is
+	  called for a sip_pvt that still has an owner channel in place,
+	  the destruction is rescheduled for 10 seconds in the future. This
+	  will continue until the owner channel is finally hung up. (closes
+	  issue ASTERISK-19425) reported by David Cunningham Patches:
+	  ASTERISK-19425.patch uploaded by Mark Michelson (License #5049)
+	  (closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by
+	  Dean Vesvuio
+
+2012-05-08 20:14 +0000 [r365631-365692]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_followme.c: * Fix FollowMe memory leak on error paths in
+	  app_exec(). * Fix FollowMe leaving recorded caller name file on
+	  error paths in app_exec(). * Use correct buffer dimension define
+	  in struct call_followme.moh[] and struct fm_args.namerecloc[].
+	  This fixes unexpected namerecloc filename length restriction.
+
+	* apps/app_followme.c: * Fix accept/decline DTMF buffer overwrite
+	  in FollowMe. * Made use MAX_YN_STRING define to make all
+	  accept/decline DTMF buffers the same size. Just using 20 isn't
+	  good enough when someone didn't get the memo. * Fix stupid use of
+	  a global variable in FollowMe. (ynlongest) * Fix bit field
+	  declarations in FollowMe. * Fix FollowMe n option documentation.
+
+2012-05-08 15:48 +0000 [r365574]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Send more accurate identification
+	  information in dialog-info SIP NOTIFYs. This uses the calling
+	  channel's caller ID and connected line information to populate
+	  the remote and local identities in the dialog-info NOTIFY when an
+	  extension is ringing. There is a bit of an oddity here, and that
+	  is that we seed the remote target with the To header of the
+	  outbound call rather than the from header. This is because it was
+	  reported that seeding with the from header caused hints to be
+	  broken with certain SNOM devices. A comment has been added to the
+	  code to explain this. (closes issue ASTERISK-16735) reported by
+	  Maciej Krajewski patches: local_remote_hint2.diff uploaded by
+	  Mark Michelson (license #5049) 16735_tweak1.diff uploaded by Mark
+	  Michelson (license #5049) Tested by Niccolo Belli
+
+2012-05-07 18:40 +0000 [r365476]  Richard Mudgett <rmudgett at digium.com>
+
+	* tests/test_config.c: Fix type punned compiler warning in
+	  test_config.c
+
+2012-05-07 18:36 +0000 [r365474]  Matthew Jordan <mjordan at digium.com>
+
+	* apps/app_voicemail.c, main/pbx.c: Support VoiceMail d() option
+	  when extension does not exist in channel's context The VoiceMail
+	  d([c]) option is documented to accept digits for a new extension
+	  in context <c>, if played during the greeting. This option works
+	  fine if the extension being redirected to has an extension with
+	  the same initial digit in the channel's current context. If that
+	  digit did not happen to exist in some extension, a dialplan match
+	  would fail and the user would not be redirected. This patch fixes
+	  it such that if the <c> option is used, the extensions are
+	  matched in that context as opposed to the caller's original
+	  context. (closes issue ASTERISK-18243) Reported by: mjordan
+	  Tested by: mjordan Review:
+	  https://reviewboard.asterisk.org/r/1892
+
+2012-05-07 16:01 +0000 [r365460]  Mark Michelson <mmichelson at digium.com>
+
+	* main/audiohook.c, res/res_speech.c, channels/sig_analog.c,
+	  main/abstract_jb.c, res/res_agi.c: Fix findings 0-3, 5, and 8 for
+	  Coverity MISSING_BREAK errors. (Issue ASTERISK-19662)
+
+2012-05-04 22:12 +0000 [r365398]  Kinsey Moore <kmoore at digium.com>
+
+	* apps/app_followme.c, channels/chan_iax2.c,
+	  channels/sip/config_parser.c, pbx/pbx_config.c,
+	  apps/app_chanspy.c, apps/app_stack.c, main/config.c,
+	  apps/app_voicemail.c, channels/chan_sip.c, funcs/func_aes.c,
+	  main/features.c: Fix many issues from the NULL_RETURNS Coverity
+	  report Most of the changes here are trivial NULL checks. There
+	  are a couple optimizations to remove the need to check for NULL
+	  and outboundproxy parsing in chan_sip.c was rewritten to avoid
+	  use of strtok. Additionally, a bug was found and fixed with the
+	  parsing of outboundproxy when "outboundproxy=," was set. (Closes
+	  issue ASTERISK-19654)
+
+2012-05-04 16:24 +0000 [r365313]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_local.c: Fix local channel chains optimizing
+	  themselves out of a call. * Made chan_local.c:check_bridge()
+	  check the return value of ast_channel_masquerade(). In long
+	  chains of local channels, the masquerade occasionally fails to
+	  get setup because there is another masquerade already setup on an
+	  adjacent local channel in the chain. * Made the outgoing local
+	  channel (the ;2 channel) flush one voice or video frame per
+	  optimization attempt. * Made sure that the outgoing local channel
+	  also does not have any frames in its queue before the masquerade.
+	  * Made do the masquerade immediately to minimize the chance that
+	  the outgoing channel queue does not get any new frames added and
+	  thus unconditionally flushed. * Made block indication -1 (Stop
+	  tones) event when the local channel is going to optimize itself
+	  out. When the call is answered, a chain of local channels pass
+	  down a -1 indication for each bridge. This blizzard of -1 events
+	  really slows down the optimization process. (closes issue
+	  ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec
+	  Davis Review: https://reviewboard.asterisk.org/r/1894/
+
+2012-05-04 15:48 +0000 [r365298]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_rtp_asterisk.c: Fix core FINDING 2, FINDING 3, and
+	  FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report.
+	  These three all are in RTP code that attempts to print the number
+	  of sequence number cycles in an RTCP RR report. The code was
+	  masking out the upper 16 bits and then shifting the number right
+	  by 16 bits. This led to an all zero result in all cases. The fix
+	  is to do the shift without the bit masking. (issue
+	  ASTERISK-19649)
+
+2012-05-03 14:54 +0000 [r365143-365159]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/ooh323c/src/h323/H323-MESSAGES.h,
+	  addons/ooh323c/src/h323/H323-MESSAGESEnc.c,
+	  addons/ooh323c/src/ooh323.c: Fix warning of Coverity Static

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