[asterisk-commits] bebuild: tag 10.6.0-digiumphones-rc1 r368804 - /tags/10.6.0-digiumphones-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jun 12 09:52:19 CDT 2012
Author: bebuild
Date: Tue Jun 12 09:52:15 2012
New Revision: 368804
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=368804
Log:
Importing files for 10.6.0-digiumphones-rc1 release.
Added:
tags/10.6.0-digiumphones-rc1/.lastclean (with props)
tags/10.6.0-digiumphones-rc1/.version (with props)
tags/10.6.0-digiumphones-rc1/ChangeLog (with props)
Added: tags/10.6.0-digiumphones-rc1/.lastclean
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+2012-06-12 Asterisk Development Team <asteriskteam at digium.com>
+
+ * 10.6.0-digiumphones-rc1 Released.
+
+2012-06-12 14:03 +0000 [r368791-368792] Matthew Jordan <mjordan at digium.com>
+
+ * /: Update merge property info
+
+ * channels/chan_sip.c: Fix deadlock in SIP transfers that involve a
+ REFER request In r367163, "send to voicemail" functionality was
+ added to the SIP channel driver. This required updating the party
+ redirecting information for the channel based on the headers
+ provided in the REFER request. When the redirecting party
+ information is updated on the channel, a call to
+ ast_indicate_data occurs. Because handle_request_refer still had
+ the sip_pvt locked, a deadlock could occur between the pbx_thread
+ and the do_monitor thread servicing the REFER request. This patch
+ preserves the proper locking order between the channel and the
+ sip_pvt by ensuring that the sip_pvt is unlocked prior to
+ updating the party redirecting information on the channel.
+ (closes issue AST-903) Reported by: Matt Jordan patches:
+ jira_ast_903_trunk.patch by rmudgett (license 5621)
+
+2012-06-11 22:49 +0000 [r368781-368783] Jason Parker <jparker at digium.com>
+
+ * /: Fix merge prop.
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/chan_sip.c: Multiple revisions 368629,368645 ........
+ r368629 | mmichelson | 2012-06-06 14:18:20 -0500 (Wed, 06 Jun
+ 2012) | 31 lines 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 ........
+ r368645 | rmudgett | 2012-06-06 16:32:09 -0500 (Wed, 06 Jun 2012)
+ | 17 lines 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 ........ Merged
+ revisions 368629,368645 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/app_record.c, include/asterisk/channel.h,
+ res/res_calendar_caldav.c, pbx/dundi-parser.c,
+ apps/app_followme.c, main/cel.c, apps/app_queue.c, main/enum.c,
+ channels/iax2-parser.c, res/res_calendar_ews.c, main/config.c,
+ main/editline/tokenizer.c, channels/chan_dahdi.c,
+ channels/sig_analog.c, main/editline/readline.c, main/event.c,
+ channels/sip/config_parser.c, res/ael/ael.flex,
+ apps/app_chanspy.c, apps/app_stack.c, res/res_odbc.c,
+ res/res_calendar.c, channels/chan_sip.c, channels/chan_agent.c,
+ funcs/func_math.c, channels/iax2-provision.c, UPGRADE.txt,
+ addons/ooh323c/src/h323/H323-MESSAGES.h, channels/chan_iax2.c,
+ res/res_monitor.c, main/channel.c, addons/ooh323c/src/ooh323.c,
+ main/cdr.c, res/ael/pval.c, main/manager.c, main/app.c,
+ pbx/pbx_dundi.c, addons/ooh323c/src/h323/H323-MESSAGESEnc.c,
+ addons/ooh323c/src/ooq931.c, main/netsock2.c,
+ res/res_rtp_asterisk.c, apps/app_meetme.c, /,
+ channels/sip/reqresp_parser.c, main/acl.c, res/res_musiconhold.c,
+ include/asterisk/tcptls.h, channels/sig_pri.c, res/res_srtp.c,
+ res/res_config_odbc.c, funcs/func_odbc.c, funcs/func_cdr.c,
+ funcs/func_channel.c, apps/app_minivm.c, main/features.c,
+ apps/app_confbridge.c, codecs/codec_dahdi.c, pbx/pbx_config.c,
+ apps/app_voicemail.c, apps/app_dial.c, funcs/func_speex.c,
+ res/res_calendar_exchange.c, funcs/func_dialgroup.c,
+ apps/app_page.c, include/asterisk/cel.h, main/say.c,
+ funcs/func_lock.c, apps/app_disa.c, main/devicestate.c, CHANGES,
+ res/res_jabber.c, main/editline/term.c, main/cli.c,
+ main/tcptls.c, main/data.c, channels/chan_skinny.c,
+ funcs/func_aes.c, tests/test_config.c, funcs/func_devstate.c,
+ channels/sip/include/sip.h, channels/sig_ss7.c, main/asterisk.c,
+ main/xmldoc.c, res/res_calendar_icalendar.c, main/pbx.c,
+ channels/chan_local.c, addons/format_mp3.c: Multiple revisions
+ 365155,365160,365299,365320,365399,365475,365478,365575,365632,365701,365898,365990,366049,366053,366106,366168,366241,366297,366390,366412,366591,366598,366741,366792,366881,366884,366948,367003,367028,367267,367299,367369,367417,367470,367562,367679,367731,367782,367844,367907,367978,367981,368042,368093,368267,368310,368407,368470,368499,368524,368536,368568,368587
+ ........ r365155 | may | 2012-05-03 09:27:00 -0500 (Thu, 03 May
+ 2012) | 11 lines Fix coverity static analysis warning, allocate
+ full ie structure instead of without data buffer (close issue
+ ASTERISK-19674) Reported by: Matt Jordan Patches:
+ ASTERISK-19674.patch (License #5415) ........ Merged revisions
+ 365143 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r365160 | may | 2012-05-03 10:01:14 -0500 (Thu, 03 May
+ 2012) | 11 lines Fix warning of Coverity Static analysis, change
+ H225ProtocolIdentifier from value to pointer per functions that
+ use this. (close issue ASTERISK-19670) Reported by: Matt Jordan
+ Patches: ASTERISK-19670.patch (License #5415) ........ Merged
+ revisions 365159 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365299 | mmichelson | 2012-05-04 10:51:04 -0500 (Fri, 04 May
+ 2012) | 12 lines 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) ........
+ Merged revisions 365298 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365320 | rmudgett | 2012-05-04 11:28:06 -0500 (Fri, 04 May 2012)
+ | 30 lines 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/ ........
+ Merged revisions 365313 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365399 | kmoore | 2012-05-04 17:15:05 -0500 (Fri, 04 May 2012) |
+ 13 lines 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) ........ Merged revisions 365398 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365475 | mjordan | 2012-05-07 13:39:10 -0500 (Mon, 07 May 2012)
+ | 20 lines 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 ........
+ Merged revisions 365474 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365478 | rmudgett | 2012-05-07 13:43:08 -0500 (Mon, 07 May 2012)
+ | 5 lines Fix type punned compiler warning in test_config.c
+ ........ Merged revisions 365476 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365575 | mmichelson | 2012-05-08 10:51:13 -0500 (Tue, 08 May
+ 2012) | 22 lines 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 ........ Merged
+ revisions 365574 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365632 | rmudgett | 2012-05-08 13:08:01 -0500 (Tue, 08 May 2012)
+ | 13 lines * 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. ........ Merged revisions 365631 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365701 | rmudgett | 2012-05-08 15:25:08 -0500 (Tue, 08 May 2012)
+ | 12 lines * 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
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365898 | mmichelson | 2012-05-09 11:15:28 -0500 (Wed, 09 May
+ 2012) | 29 lines 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 ........
+ r365990 | jrose | 2012-05-09 14:12:32 -0500 (Wed, 09 May 2012) |
+ 18 lines 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 ........
+ r366049 | jrose | 2012-05-10 10:43:06 -0500 (Thu, 10 May 2012) |
+ 9 lines 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 ........
+ r366053 | mmichelson | 2012-05-10 11:13:06 -0500 (Thu, 10 May
+ 2012) | 9 lines 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 ........
+ r366106 | jrose | 2012-05-10 11:55:22 -0500 (Thu, 10 May 2012) |
+ 9 lines 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 ........
+ r366168 | kmoore | 2012-05-10 15:54:08 -0500 (Thu, 10 May 2012) |
+ 13 lines 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 ........
+ r366241 | rmudgett | 2012-05-10 18:42:43 -0500 (Thu, 10 May 2012)
+ | 7 lines * 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
+ ........ r366297 | russell | 2012-05-11 18:59:35 -0500 (Fri, 11
+ May 2012) | 19 lines 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
+ ........ r366390 | mmichelson | 2012-05-14 14:16:36 -0500 (Mon,
+ 14 May 2012) | 25 lines 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
+ ........ r366412 | mmichelson | 2012-05-14 15:06:58 -0500 (Mon,
+ 14 May 2012) | 19 lines 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 ........
+ r366591 | jrose | 2012-05-15 15:44:59 -0500 (Tue, 15 May 2012) |
+ 15 lines 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 ........
+ r366598 | mmichelson | 2012-05-15 18:39:06 -0500 (Tue, 15 May
+ 2012) | 8 lines 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 ........
+ r366741 | mjordan | 2012-05-17 07:57:30 -0500 (Thu, 17 May 2012)
+ | 23 lines 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 ........
+ r366792 | jrose | 2012-05-17 09:41:13 -0500 (Thu, 17 May 2012) |
+ 10 lines 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 ........
+ r366881 | mjordan | 2012-05-18 09:01:56 -0500 (Fri, 18 May 2012)
+ | 65 lines 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
+ ........ r366884 | kmoore | 2012-05-18 09:18:47 -0500 (Fri, 18
+ May 2012) | 9 lines 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 ........
+ r366948 | mjordan | 2012-05-18 10:45:42 -0500 (Fri, 18 May 2012)
+ | 20 lines 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 ........
+ r367003 | mmichelson | 2012-05-18 12:00:14 -0500 (Fri, 18 May
+ 2012) | 19 lines 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 ........
+ r367028 | mmichelson | 2012-05-18 12:50:18 -0500 (Fri, 18 May
+ 2012) | 18 lines 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 ........
+ r367267 | twilson | 2012-05-22 11:17:46 -0500 (Tue, 22 May 2012)
+ | 14 lines 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 ........
+ r367299 | twilson | 2012-05-22 12:21:51 -0500 (Tue, 22 May 2012)
+ | 21 lines 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 ........
+ r367369 | mjordan | 2012-05-23 08:25:04 -0500 (Wed, 23 May 2012)
+ | 26 lines 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
+ ........ r367417 | mmichelson | 2012-05-23 15:29:03 -0500 (Wed,
+ 23 May 2012) | 7 lines 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 ........
+ r367470 | rmudgett | 2012-05-23 18:16:49 -0500 (Wed, 23 May 2012)
+ | 9 lines 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
+ ........ r367562 | mjordan | 2012-05-24 08:32:33 -0500 (Thu, 24
+ May 2012) | 24 lines 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) ........ r367679 | rmudgett | 2012-05-24
+ 17:29:23 -0500 (Thu, 24 May 2012) | 34 lines 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 ........
+ r367731 | elguero | 2012-05-24 21:29:26 -0500 (Thu, 24 May 2012)
+ | 20 lines 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 ........
+ r367782 | rmudgett | 2012-05-25 11:30:55 -0500 (Fri, 25 May 2012)
+ | 18 lines 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 ........
+ r367844 | mjordan | 2012-05-29 13:33:20 -0500 (Tue, 29 May 2012)
+ | 21 lines 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) ........ r367907 | rmudgett |
+ 2012-05-29 17:28:55 -0500 (Tue, 29 May 2012) | 17 lines 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
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