[asterisk-commits] bebuild: tag 10.6.0-rc1 r368706 - /tags/10.6.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jun 8 10:28:14 CDT 2012
Author: bebuild
Date: Fri Jun 8 10:28:10 2012
New Revision: 368706
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=368706
Log:
Importing files for 10.6.0-rc1 release.
Added:
tags/10.6.0-rc1/.lastclean (with props)
tags/10.6.0-rc1/.version (with props)
tags/10.6.0-rc1/ChangeLog (with props)
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+2012-06-08 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.6.0 Released.
+
+2012-06-06 21:32 +0000 [r368645] 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
+ ........ Merged revisions 368644 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-06 19:18 +0000 [r368629] 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
+ ........ Merged revisions 368625 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-06 17:21 +0000 [r368605] 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". ........ Merged
+ revisions 368604 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-06 16:09 +0000 [r368587] 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 ........ Merged revisions
+ 368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-06 01:10 +0000 [r368568] 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
+ ........ Merged revisions 368567 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-05 15:27 +0000 [r368524-368536] 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. ........ Merged revisions 368533 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * 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)
+ ........ Merged revisions 368520 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-04 22:02 +0000 [r368499] 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) ........ Merged revisions 368498 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-04 21:11 +0000 [r368407-368470] Richard Mudgett <rmudgett at digium.com>
+
+ * /, UPGRADE.txt: Document BLINDTRANSFER behavior change. (issue
+ ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call
+ ........ Merged revisions 368469 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * 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/
+ ........ Merged revisions 368405 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-01 23:24 +0000 [r368310] 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
+ ........ Merged revisions 368308 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-01 20:22 +0000 [r368267] 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/ ........ Merged
+ revisions 368218 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-01 03:28 +0000 [r368093] 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/
+ ........ Merged revisions 368092 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-31 18:20 +0000 [r368042] Richard Mudgett <rmudgett at digium.com>
+
+ * 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,
+ apps/app_queue.c, channels/chan_iax2.c, pbx/pbx_config.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 ........ Merged
+ revisions 368039 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-30 18:07 +0000 [r367907-367981] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/sig_pri.c, channels/sig_ss7.c: Use the
+ DEADLOCK_AVOIDANCE() macro instead. (issue ASTERISK-19854)
+ ........ Merged revisions 367980 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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 ........ Merged revisions 367976 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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 ........ Merged revisions 367906 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-29 18:33 +0000 [r367844] 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 dereferened 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 (licesen 6283)
+
+2012-05-25 16:30 +0000 [r367782] 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 ........ Merged
+ revisions 367781 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-25 02:29 +0000 [r367731] 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/ ........ Merged
+ revisions 367730 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-24 22:29 +0000 [r367679] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_dial.c, /, apps/app_queue.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/ ........ Merged
+ revisions 367678 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-24 13:32 +0000 [r367562] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_confbridge.c: Fix crash in ConfBridge when user
+ announcement is played for more than 2 users A patch introduced
+ in r354938 made it so that ConfBridge would not attempt to play
+ sound files if those files did not exist. Unfortunately,
+ ConfBridge uses the same underlying function, play_sound_helper,
+ to playback both sound files and numbers to callers. When a
+ number is being played back, the name of the sound file is
+ expected to be NULL. This NULL value was passed into a function
+ that tested for the existance of a sound file and is not tolerant
+ to NULL file names, causing a crash. This patch fixes the
+ behavior, such that if a sound file does not exist we do not
+ attempt to play it, but we only attempt that check if the a sound
+ file was specified in the first place. If a sound file was not
+ specified, we use the 'play number' logic in the helper function.
+ (closes issue ASTERISK-19899) Reported by: Florian Gilcher Tested
+ by: Florian Gilcher patches: asterisk-19899.diff uploaded by
+ mjordan (license 6283)
+
+2012-05-23 23:16 +0000 [r367470] Richard Mudgett <rmudgett at digium.com>
+
+ * 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. ........ Merged
+ revisions 367469 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-23 20:29 +0000 [r367417] 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. ........
+ Merged revisions 367416 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-23 13:25 +0000 [r367369] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c, CHANGES, channels/sip/include/sip.h:
+ Re-add LastMsgsSent value for SIP peers 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. The value was previously removed for Asterisk 10; however,
+ since it was still present in Asterisk 1.8 and still useful for
+ reporting purposes, it was decided to re-add the value. This
+ patch re-adds the 'LastMsgsSent' field in the response to an
+ AMI/CLI 'sip show peer [peer]' command, and 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 ........ Merged revisions
+ 367362 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-22 17:21 +0000 [r367267-367299] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c, /, include/asterisk/cel.h,
+ include/asterisk/channel.h, main/cel.c, main/asterisk.c: 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/
+ ........ Merged revisions 367292 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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/ ........ Merged
+ revisions 367266 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-18 17:50 +0000 [r367003-367028] 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 ........ Merged
+ revisions 367027 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
+ 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) ........ Merged revisions 367002 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-18 15:45 +0000 [r366948] 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/ ........ Merged
+ revisions 366944 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-18 14:18 +0000 [r366884] 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.
+ ........ Merged revisions 366882 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-18 14:01 +0000 [r366881] Matthew Jordan <mjordan at digium.com>
+
+ * main/xmldoc.c, apps/app_voicemail.c, res/res_calendar.c,
+ main/netsock2.c, res/res_rtp_asterisk.c, main/pbx.c,
+ res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
+ apps/app_page.c, /, funcs/func_dialgroup.c, channels/chan_sip.c,
+ apps/app_record.c, res/res_calendar_caldav.c,
+ res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c,
+ res/res_jabber.c, main/editline/term.c, main/enum.c,
+ main/config.c, res/res_srtp.c, main/cli.c,
+ main/editline/tokenizer.c, main/data.c, channels/chan_dahdi.c,
+ funcs/func_odbc.c, main/features.c, apps/app_minivm.c,
+ main/editline/readline.c, channels/sip/config_parser.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 ........ Merged revisions
+ 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-17 14:41 +0000 [r366792] 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) Merged revisions
+ 366791 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-17 12:57 +0000 [r366741] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_dahdi.c, /, res/res_calendar_ews.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 ........ Merged
+ revisions 366740 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-15 23:39 +0000 [r366598] Mark Michelson <mmichelson at digium.com>
+
+ * /, 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.
+ ........ Merged revisions 366597 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-15 20:44 +0000 [r366591] 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
+ denydirectmediadeny, Asterisk would check the host address of the
+ channel permit/deny was specified, which defers 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/
+ ........ Merged revisions 366547 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-14 20:06 +0000 [r366390-366412] 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 ........ Merged
+ revisions 366409 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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 ........ Merged revisions
+ 366389 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-11 23:59 +0000 [r366297] 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
+ ........ Merged revisions 366296 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-10 23:42 +0000 [r366241] 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(). ........
+ Merged revisions 366240 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-10 20:54 +0000 [r366168] Kinsey Moore <kmoore at digium.com>
+
+ * 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, main/devicestate.c,
+ pbx/dundi-parser.c, channels/chan_iax2.c, channels/iax2-parser.c,
+ main/config.c, res/res_monitor.c, main/cdr.c, main/channel.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: 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)
+ ........ Merged revisions 366167 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-10 16:55 +0000 [r366106] Jonathan Rose <jrose at digium.com>
+
+ * 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, channels/iax2-provision.c,
+ apps/app_queue.c, channels/chan_iax2.c, res/ael/ael.flex,
+ funcs/func_devstate.c, main/asterisk.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/ ........ Merged
+ revisions 366094 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-10 16:13 +0000 [r366053] 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 ........ Merged revisions
+ 366052 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-10 15:43 +0000 [r365990-366049] Jonathan Rose <jrose at digium.com>
+
+ * /, apps/app_page.c, funcs/func_cdr.c, main/features.c,
+ apps/app_disa.c, apps/app_chanspy.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/ ........ Merged
+ revisions 366048 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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) ........ Merged
+ revisions 365989 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-09 16:15 +0000 [r365898] 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 ........ Merged revisions 365896 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-08 20:25 +0000 [r365632-365701] 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. ........ Merged revisions 365692 from
[... 24418 lines stripped ...]
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