[asterisk-commits] bebuild: tag 1.8.14.0-rc1 r368702 - /tags/1.8.14.0-rc1/
SVN commits to the Asterisk project
asterisk-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.
Added:
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tags/1.8.14.0-rc1/.version (with props)
tags/1.8.14.0-rc1/ChangeLog (with props)
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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
[... 39599 lines stripped ...]
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