[svn-commits] bebuild: tag 10.4.0-rc1 r361168 - /tags/10.4.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Apr 4 13:21:03 CDT 2012
Author: bebuild
Date: Wed Apr 4 13:20:57 2012
New Revision: 361168
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=361168
Log:
Importing files for 10.4.0-rc1 release.
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tags/10.4.0-rc1/.version (with props)
tags/10.4.0-rc1/ChangeLog (with props)
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+2012-04-04 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.4.0-rc1 Released.
+
+2012-04-04 16:38 +0000 [r361091-361143] Jonathan Rose <jrose at digium.com>
+
+ * main/channel.c, pbx/pbx_loopback.c, addons/chan_ooh323.c, /,
+ channels/chan_sip.c, main/app.c, pbx/pbx_realtime.c,
+ apps/app_externalivr.c, channels/chan_iax2.c,
+ res/res_fax_spandsp.c, apps/app_milliwatt.c: Replace GNU
+ old-style field designator extensions to fix clang warnings
+ (issue ASTERISK-19540) Reported by: Makoto Dei Patches:
+ clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
+ ........ Also add from the patch the portion in res_fax_spandsp
+ that didn't apply to 1.8 Merged revisions 361142 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 (closes issue
+ ASTERISK-19540)
+
+ * /, apps/app_meetme.c: Make the MeetMeAdmin N command (mute all
+ nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported
+ by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/
+ ........ Merged revisions 361090 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-03 20:08 +0000 [r360993-361041] Kinsey Moore <kmoore at digium.com>
+
+ * /, apps/app_transfer.c: Fix the display of documentation for
+ Transfer This came up while fixing documentation generation for
+ many other cases where the argument separator was not being
+ displayed properly. Now that it is displayed properly, it shows
+ up in the wrong place for Transfer since the '/' is only required
+ if Tech is present. (related to issue ASTERISK-18168) ........
+ Merged revisions 361040 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Stop sending out RTCP if RTP is inactive
+ This change prevents Asterisk from sending RTCP receiver reports
+ during a remote bridge since it is no longer receiving media and
+ should not be reporting anything. (related to ASTERISK-19366)
+ ........ Merged revisions 360987 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-30 21:29 +0000 [r360934] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/logger.c: Fix logger deadlock on Asterisk shutdown. The
+ logger_thread() had an exit path that failed to release the
+ logmsgs list lock. * Make logger_thread() exit path unlock the
+ logmsgs list lock. * Made ast_log() not queue any messages to the
+ logmsgs list if the close_logger_thread flag is set. (issue
+ ASTERISK-19463) Reported by: Matt Jordan ........ Merged
+ revisions 360933 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-29 23:33 +0000 [r360863-360885] Mark Michelson <mmichelson at digium.com>
+
+ * /, main/features.c: Fix potential race condition during call
+ pickup. Prior to this patch, a connected line update was queued
+ during call pickup and then an answer frame was queued. The
+ original caller would presumably then have his connected line
+ updated and then the call would be answered. In actuality, the
+ answer frame was not how the call ended up being answered.
+ Rather, an odd section in app_dial that checks if the called
+ channel's state is up. The result is that the order of the
+ connected line update and the answer were variable. In most
+ cases, this wasn't actually a bad thing. However, if the 'I'
+ option was passed to dial, the connected line update would be
+ inhibited. The fix is to queued the connected line after the
+ answer frame is queued. This way the race in app_dial is between
+ two conditions resulting in an answer. This way the connected
+ line update occurs after the answer every time. (closes issue
+ ASTERISK-19183) Reported by: Thomas Arimont Tested by: Thomas
+ Arimont Mark Michelson Patches: ASTERISK-19183.patch uploaded by
+ Mark Michelson (license 5049) ........ Merged revisions 360884
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Improve accuracy of identifying
+ information sent in dialog-info SIP NOTIFY requests. This change
+ makes use of connected party information in addition to caller ID
+ in order to populate local and remote XML elements in the
+ dialog-info NOTIFYs. (closes issue ASTERISK-16735) Reported by:
+ Maciej Krajewski Tested by: Maciej Krajewski Patches:
+ local_remote_hint2.diff uploaded by Mark Michelson (license 5049)
+ ........ Merged revisions 360862 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-28 19:20 +0000 [r360717] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_jingle.c, addons/chan_ooh323.c, /,
+ cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c,
+ channels/chan_gtalk.c: Destroy configs when they are no longer
+ used https://reviewboard.asterisk.org/r/1834/ ........ Merged
+ revisions 360712 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-27 18:23 +0000 [r360672] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Make a debug message regarding
+ subscription changes more accurate. I was getting confused during
+ some testing why Asterisk was saying that a subscription was
+ being added when it was clearly being removed. This fixes that
+ confusion. ........ Merged revisions 360625 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-27 14:35 +0000 [r360489-360575] Jonathan Rose <jrose at digium.com>
+
+ * /, configure: Updates config with bootstrap where I changed
+ configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon
+ Clark ........ Merged revisions 360574 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, configure.ac: Fix BETTER_BACKTRACES library detection for
+ Fedora/RedHat/CentOS (closes ASTERISK-17842) Reported by: Bryon
+ Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman
+ Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch
+ uploaded by Bryon Clark (license 6157) ........ Merged revisions
+ 360488 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-26 18:41 +0000 [r360472-360476] Paul Belanger <pabelanger at digium.com>
+
+ * /, CHANGES: Update CHANGES for r360471 ........ Merged revisions
+ 360474 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/dnsmgr.c, /: Increase verbosity level for ast_verb messages
+ While this does not fix the issue of the CLI being flooded by
+ 'doing dnsmgr_lookup' messages, increasing the verbosity level
+ above 5 should help minimize it. ........ Merged revisions 360471
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-24 23:47 +0000 [r360358-360414] Russell Bryant <russell at russellbryant.com>
+
+ * funcs/func_curl.c, /: func_curl: Fix leak of an ast_str in error
+ handling code path. ........ Merged revisions 360413 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_page.c: app_page: Fix a memory leak on every Page().
+ dial_list is a dynamically allocated array that is allocated at
+ the beginning of Page() based on how many devices will be dialed.
+ This was never being freed. ........ Merged revisions 360363 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_jack.c, /: app_jack: fix datastore memory leak in error
+ handling path. ........ Merged revisions 360360 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/ast_expr2.c, /, main/ast_expr2.h, res/ael/ael.tab.c,
+ main/ast_expr2.y, main/ast_expr2f.c, res/ael/ael_lex.c,
+ res/ael/ael.tab.h: Multiple revisions 360356-360357 ........
+ r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012)
+ | 6 lines expression parser: Fix (theoretical) memory leak. Fix a
+ memory leak that is very unlikely to actually happen. If a
+ malloc() succeeded, but the following strdup() failed, the memory
+ from the original malloc() would be leaked. ........ r360357 |
+ russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines
+ Rebuild parsers. This is needed to include the last fix to
+ main/ast_expr2.y. The changes look much bigger as this
+ regeneration of the code was done with newer versions of flex and
+ bison. ........ Merged revisions 360356-360357 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-24 00:37 +0000 [r360263-360310] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, /, channels/sig_pri.c: Make number not available
+ presentation also set screening to network provided. Q.951
+ indicates that when the presentation indicator is "Number not
+ available due to interworking" for a number then the screening
+ indicator field should be "Network provided". * Made
+ ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
+ when the presentation is "Number not available due to
+ interworking". This fix makes Asterisk consistent and it also
+ makes it consistent with earlier branches as far as this
+ presentation value is concerned. * Made pri_to_ast_presentation()
+ and ast_to_pri_presentation() conversions handle the "Number not
+ available due to interworking" case better in sig_pri.c. This
+ change is possible because the minimum required libpri version
+ (v1.4.11) has the necessary defines in libpri.h. ........ Merged
+ revisions 360309 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Add missing initialization of
+ update_redirecting in chan_sip.c ........ Merged revisions 360262
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-21 14:52 +0000 [r360139] Jonathan Rose <jrose at digium.com>
+
+ * contrib/scripts/install_prereq, /: Update install_prereq script
+ to include missing GSM library for debian amd move SQLite3.
+ (closes issue ASTERISK-19367) Reported by: Andrew Latham Patches:
+ debian_install_prereq.diff uploaded by Andrew Latham (license
+ 5985) ........ Merged revisions 360138 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-21 14:21 +0000 [r360098] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * /, configure, configure.ac: Also detect gmime 2.6 Also detect
+ gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen
+ (License #5035) <tzafrir.cohen at xorcom.com> ........ Merged
+ revisions 360087 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-21 13:28 +0000 [r360088] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: Ensure Asterisk sends a BYE when pending
+ on the final response to a re-INVITE When Asterisk detects a
+ hangup and cannot send a BYE due to a pending INVITE, it sets the
+ pendingbye flag and waits for the final response to that INVITE.
+ When the response is received, it transmits the BYE. If, however,
+ that INVITE request is a pending re-INVITE, it needs to first
+ send a CANCEL request to terminate the pending re-INVITE. In that
+ circumstance, Asterisk was, in some scenarios, clearing the
+ pendingbye flag after processing the CANCEL request and not
+ checking for a pending BYE when receiving the final 487 response
+ to the INVITE. This patch ensures that if the pendingbye flag is
+ set, it is honored regardless of the nature of the INVITE request
+ currently in flight. (closes issue ASTERISK-19365) Reported by:
+ Thomas Arimont Tested by: Thomas Arimont Patches:
+ bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license
+ 6283) Review: https://reviewboard.asterisk.org/r/1807 ........
+ Merged revisions 360086 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-20 20:37 +0000 [r360034] Kinsey Moore <kmoore at digium.com>
+
+ * /, apps/app_echo.c: Prevent Echo() from relaying control, null,
+ and modem frames Echo()'s description states that it echoes
+ audio, video, and DTMF except for # while it actually echoes any
+ frame that it receives other than DTMF #. This was causing frame
+ storms in the test suite in some circumstances where Echo() was
+ attached to both ends of a pair of local channels and control
+ frames were being periodically generated. Echo()'s behavior and
+ description have been modifed so that it only echoes media and
+ non-# DTMF frames. ........ Merged revisions 360033 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-20 18:11 +0000 [r359982] Sean Bright <sean at malleable.com>
+
+ * channels/chan_iax2.c: chan_iax2: Emit Port alongside Post in
+ PeerStatus AMI Event. The PeerStatus event for IAX2 channels
+ currently includes a header named Post which should have been
+ Port. So include Port along with Post when emitting the event.
+ We'll remove Post in trunk.
+
+2012-03-20 17:25 +0000 [r359980] Richard Mudgett <rmudgett at digium.com>
+
+ * main/manager.c, /, include/asterisk/manager.h: Allow AMI action
+ callback to be reentrant. Fix AMI module reload deadlock
+ regression from ASTERISK-18479 when it tried to fix the race
+ between calling an AMI action callback and unregistering that
+ action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change.
+ Locking the ao2 object guaranteed that there were no active
+ callbacks that mattered when ast_manager_unregister() was called.
+ Unfortunately, this causes the deadlock situation. The patch
+ stops locking the ao2 object to allow multiple threads to invoke
+ the callback re-entrantly. There is no way to guarantee a module
+ unload will not crash because of an active callback. The code
+ attempts to minimize the chance with the registered flag and the
+ maximum 5 second delay before ast_manager_unregister() returns.
+ The trunk version of the patch changes the API to fix the race
+ condition correctly to prevent the module code from unloading
+ from memory while an action callback is active. * Don't hold the
+ lock while calling the AMI action callback. (closes issue
+ ASTERISK-19487) Reported by: Philippe Lindheimer Review:
+ https://reviewboard.asterisk.org/r/1818/ Review:
+ https://reviewboard.asterisk.org/r/1820/ ........ Merged
+ revisions 359979 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-16 20:20 +0000 [r359898] Jonathan Rose <jrose at digium.com>
+
+ * /, apps/app_chanspy.c: Prevent chanspy from binding to zombie
+ channels This patch addresses a bug with chanspy on local
+ channels which roughly 50% of the time would create a situation
+ where chanspy can latch onto a zombie channel, keeping the zombie
+ alive forever and causing the channel doing the spying to never
+ be able to hang up. (closes issue ASTERISK-19493) Reported by:
+ lvl Review: https://reviewboard.asterisk.org/r/1819/ ........
+ Merged revisions 359892 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-16 08:24 +0000 [r359810] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * /, channels/sip/include/sip.h: Missed lastinvite CSeq int to
+ uint32_t change from Review:
+ https://reviewboard.asterisk.org/r/1699/ ........ Merged
+ revisions 359809 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-15 19:06 +0000 [r359694-359707] Matthew Jordan <mjordan at digium.com>
+
+ * /, main/utils.c: Fix remotely exploitable stack overflow in HTTP
+ manager There exists a remotely exploitable stack buffer overflow
+ in HTTP digest authentication handling in Asterisk. The
+ particular method in question is only utilized by HTTP AMI. When
+ parsing the digest information, the length of the string is not
+ checked when it is copied into temporary buffers allocated on the
+ stack. This patch fixes this behavior by parsing out pre-defined
+ key/value pairs and avoiding unnecessary copies to the stack.
+ (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested
+ by: Matt Jordan ........ Merged revisions 359706 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_milliwatt.c: Fix remotely exploitable stack overrun
+ in Milliwatt Milliwatt is vulnerable to a remotely exploitable
+ stack overrun when using the 'o' option. This occurs due to the
+ milliwatt_generate function not accounting for
+ AST_FRIENDLY_OFFSET when calculating the maximum number of
+ samples it can put in the output buffer. This patch resolves this
+ issue by taking into account AST_FRIENDLY_OFFSET when determining
+ the maximum number of samples allowed. Note that at no point is
+ remote code execution possible. The data that is written into the
+ buffer is the pre-defined Milliwatt data, and not custom data.
+ (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested
+ by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by
+ Russell Bryant (license 6283) Note that this patch was written by
+ Russell, even though Matt uploaded it ........ Merged revisions
+ 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+ ........ Merged revisions 359656 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-15 18:22 +0000 [r359620] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_dial.c, /, apps/app_queue.c: Add missing connected line
+ macro calls to initial dial for Dial and Queue apps. The
+ connected line interception macros do not get executed when the
+ outgoing channel is initially created and that channel's
+ caller-id is implicitly imported into the incoming channel's
+ connected line data. If you are using the interception macros,
+ you would expect that they get run for every change to a
+ channel's connected line information outside of normal dialplan
+ execution. Review: https://reviewboard.asterisk.org/r/1817/
+ ........ Merged revisions 359609 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-15 00:53 +0000 [r359454-359559] Russell Bryant <russell at russellbryant.com>
+
+ * /, channels/chan_iax2.c: chan_iax2: Fix use of uninitialized
+ sockaddr_in in try_transfer(). Initialize a struct sockaddr_in in
+ try_transfer() so that the code isn't (potentially) trying to
+ read from it while uninitialized. ........ Merged revisions
+ 359558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_gtalk.c: chan_gtalk: Fix potential use of
+ uninitialized variable. Avoid potential use of idroster in
+ gtalk_alloc() before it has been initialized. ........ Merged
+ revisions 359508 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_chanisavail.c: app_chanisavail: Fix use of
+ uninitialized variable. Ensure that status is set before it is
+ used by resetting it during each loop iteration. This could have
+ resulted in incorrect results from this app. ........ Merged
+ revisions 359486 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/udptl.c, /: udptl: Ensure fec[] in udptl_build_packet() is
+ initialized. Scan results indicated that this array could be used
+ uninitialized. At a quick look, it looks correct. In any case,
+ initializing it is a Good Thing (tm). ........ Merged revisions
+ 359457 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * include/asterisk/app.h, /: app.h: Always initialize
+ AST_DECLARE_APP_ARGS(). This patch ensures that the struct
+ defined by AST_DECLARE_APP_ARGS() is always fully initialized.
+ I'm not sure if this fixes any real bugs, but it silences a bunch
+ of warnings from coverity, and is generally a good thing to do
+ anyway. ........ Merged revisions 359452 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-14 22:28 +0000 [r359453] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, /, channels/chan_agent.c,
+ include/asterisk/channel.h: Fix deadlock potential with some
+ ast_indicate/ast_indicate_data calls. Calling
+ ast_indicate()/ast_indicate_data() with the channel lock held can
+ result in a deadlock with a local channel because of how local
+ channels need to avoid deadlock. ........ Merged revisions 359451
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-14 17:42 +0000 [r359358] Matthew Jordan <mjordan at digium.com>
+
+ * /, main/jitterbuf.c: Fix incorrect jitter buffer overflow due to
+ missed resynchronizations When a change in time occurs, such that
+ the timestamps associated with frames being placed into an
+ adaptive jitter buffer (implemented in jitterbuf.c) are
+ significantly different then the previously inserted frames, the
+ jitter buffer checks to see if it needs to be resynched to the
+ new time frame. If three consecutive packets break the threshold,
+ the jitter buffer resynchs itself to the new timestamps. This
+ currently only occurs when history is calculated, and hence only
+ on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other
+ hand, are never passed to the history calculations. Because of
+ this, if the jump in time is greater then the maximum allowed
+ length of the jitter buffer, the JB_TYPE_CONTROL frames are
+ dropped and no resynchronization occurs. Alterntively, if the
+ overfill logic is not triggered, the JB_TYPE_CONTROL frame will
+ be placed into the buffer, but with a time reference that is not
+ applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger
+ the overflow logic until reads from the jitter buffer reach the
+ errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL
+ frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames
+ are unlikely to occur in multiples, it perform the
+ resynchronization on any JB_TYPE_CONTROL frame that breaks the
+ resynch threshold. Note that this only impacts chan_iax2, as
+ other consumers of the adaptive jitter buffer use the abstract
+ jitter buffer API, which does not use JB_TYPE_CONTROL frames.
+ Review: https://reviewboard.asterisk.org/r/1814/ (closes issue
+ ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt
+ Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw
+ (license 5722) ........ Merged revisions 359356 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-14 17:24 +0000 [r359355] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_dial.c, main/channel.c, /: Fix Dial m and r options and
+ forked calls generating warnings for voice frames. When connected
+ line support was added, the wait_for_answer() variable single
+ changed its meaning slightly. Unfortunately, the places where
+ single was used did not necessarily get updated to reflect that
+ change. Also audio/video frames were sent to all forked calls
+ when the endpoints were never made compatible. * Don't pass
+ audio/video media frames when the channels have not been made
+ compatible. * Added handling of AST_CONTROL_SRCCHANGE to
+ app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD
+ because that frame can also pass a requested MOH class. (closes
+ issue ASTERISK-16901) Reported by: Chris Gentle (closes issue
+ ASTERISK-17541) Reported by: clint Review:
+ https://reviewboard.asterisk.org/r/1805/ ........ Merged
+ revisions 359344 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-14 10:54 +0000 [r359051-359260] Russell Bryant <russell at russellbryant.com>
+
+ * include/asterisk/logger.h, /, main/logger.c: Fix bogus
+ reads/writes of console log levels in asterisk.c This patch
+ updates the NUMLOGLEVELS define in logger.h to 32, to match the
+ fact that logger.c implements 32 log levels (because of the
+ custom log level stuff). asterisk.c uses this define to size an
+ array of levels per remote console. This array is modified in
+ ast_console_toggle_loglevel(), which is called by the "logger set
+ level" CLI command. While the documentation for the CLI command
+ doesn't make it terribly obvious, you can use this CLI command to
+ toggle a custom log level on a remote console, as well. However,
+ doing so led to an invalid array index in asterisk.c. This array
+ is read from any time a log message is written to a console. So,
+ all custom log level messages resulted in a bogus read if a
+ remote console was connected. ........ Merged revisions 359259
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_externalivr.c, channels/chan_iax2.c: Fix invalid
+ reads/writes due to incorrect sizeof(). These few places in the
+ code used sizeof() on h_addr in struct hostent. This is
+ sizeof(char *). The correct way to get the size of this address
+ is to use h_length. This error would result in reads/writes of 8
+ bytes instead of 4 on 64-bit machines. ........ Merged revisions
+ 359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/sched.c, /: Fix inaccurate sizeof() in sched.c. This code
+ just needed sizeof(int), not sizeof(int *). ........ Merged
+ revisions 359157 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, utils/astman.c: Fix incorrect sizeof() in astman. ........
+ Merged revisions 359116 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, res/res_crypto.c: Fix incorrect usage of sizeof() in
+ res_crypto. In this case, just remove the memset(). There was a
+ redundant memset that is done correctly just 2 lines later.
+ ........ Merged revisions 359110 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, res/res_adsi.c: Fix broken usage of sizeof() in res_adsi.
+ ........ Merged revisions 359088 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/features.c: Fix incorrect sizeof() usage in features.c.
+ This didn't actually result in a bug anywhere, luckily. The only
+ place where the result of these memcpys was used is in app_dial,
+ and the only field that it read out of ast_call_feature was the
+ first one, which is an int, so these memcpys always copied just
+ enough to avoid a problem. ........ Merged revisions 359069 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final().
+ ........ Merged revisions 359059 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/pbx.c, /: Don't use a buffer after it goes out of scope. 's'
+ is set to 'workspace'. Make sure 'workspace' doesn't go out of
+ scope while the reference to it via 's' is still used. ........
+ Merged revisions 359056 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * res/res_ais.c, /, res/ais/clm.c, res/ais/evt.c, res/ais/ais.h:
+ Dump cache of published events when a node joins the cluster.
+ Also use a more reliable method for stopping the poll() thread.
+ ........ Merged revisions 359053 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_usbradio.c (removed), /, channels/xpmr (removed),
+ build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
+ apps/app_rpt.c (removed): Remove chan_usbradio and app_rpt. These
+ modules are being maintained outside of the tree and have been
+ for a long time now, so it doesn't make sense to keep them here.
+ Review: https://reviewboard.asterisk.org/r/1764/ ........ Merged
+ revisions 359050 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-13 20:36 +0000 [r358944-358989] Terry Wilson <twilson at digium.com>
+
+ * /, main/features.c: Fix setting CDR variables in the hangup
+ extension A previous CDR fix for setting CDR variables during a
+ bridge via custom dialplan features broke setting CDR variables
+ in the hangup extension. This patch fixes the issue. Review:
+ https://reviewboard.asterisk.org/r/1794/ ........ Merged
+ revisions 358978 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * include/asterisk/devicestate.h, /, channels/chan_sip.c,
+ tests/test_devicestate.c, main/devicestate.c: Make hints for
+ invalid SIP devices return Unavail, not idle This patch
+ drastically simplifies the device state aggegation code. The old
+ method was not only overly complex, but also made it impossible
+ to return AST_DEVICE_INVALID from the aggregation code. The unit
+ test update is as a result of fixing that bug. The SIP change
+ stems from a bug introduced by removing a DNS lookup for
+ hostname-based SIP channels. (closes issue ASTERISK-16702)
+ Review: https://reviewboard.asterisk.org/r/1808/ ........ Merged
+ revisions 358943 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-13 16:58 +0000 [r358811-358860] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * /, UPGRADE.txt, CHANGES: Requested changes documenting the fixed
+ AEL functionality. ........ Merged revisions 358859 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * res/ael/pval.c, funcs/func_dialplan.c, /, tests/test_gosub.c,
+ utils/ael_main.c, apps/app_stack.c, utils/conf2ael.c: Enable
+ macros in 1.8 to find the next highest "h" extension in a
+ context, like in 1.4. This change restores functionality that was
+ present in 1.4, when AEL macros were implemented with the Macro
+ dialplan application. Macros are fraught with functionality
+ issues, because they consume a large portion of the underlying
+ application stack. This limits the ability of AEL users to call
+ many layers of subroutines, an issue which Gosub does not have
+ (originally tested to 100,000 levels deep). Therefore, starting
+ in 1.6.0, AEL macros were implemented with Gosub. However, there
+ were some implicit behaviors of Macro, which were not replicated
+ at the same time as with the transition to Gosub, one of which is
+ documented in the related issue. In particular, the "h" extension
+ is designed to execute not in the Macro context, but in the
+ topmost calling context. Due to legacy issues with a misapplied
+ bugfix many years ago, when a macro exited in 1.4, it looks in
+ all calling contexts, bubbling up from the deepest level until it
+ finds an "h" extension. Since AEL hides the complexity of the
+ underlying dialplan logic from the AEL programmer, it's
+ reasonable to assume that this behavior should not change in the
+ transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we
+ break working AEL configurations in the transition to Asterisk
+ 1.8 LTS. This fix is the result, which implements a search for
+ the "h" extension in all calling Gosub contexts. Fixes
+ ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff
+ (License #5003) by Tilghman Lesher (with slight modifications for
+ 1.8) Tested by: Johan Wilfer Review:
+ https://reviewboard.asterisk.org/r/1776/ ........ Merged
+ revisions 358810 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-08 16:50 +0000 [r358644] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/chan_sip.c: Make transfer not ignore port information
+ with SIP. Attempting to transfer with SIP to an address like
+ 1XXXXX at ip.ad.re.ss:5061 would fail because port would be cut from
+ the host string and ignored. This simply keeps chan_sip from
+ cutting off the port number during these kinds of transfers.
+ (closes issue ASTERISK-19321) Reported by: Federico Alves Review:
+ https://reviewboard.asterisk.org/r/1790/diff/#index_header
+ ........ Merged revisions 358643 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-07 18:28 +0000 [r358531] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/sig_ss7.c: Change directly setting _softhangup in
+ sig_ss7.c to use ast_softhangup_nolock(). Update to: (issue
+ ASTERISK-19372) ........ Merged revisions 358530 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-07 16:13 +0000 [r358485] Sean Bright <sean at malleable.com>
+
+ * /, codecs/codec_dahdi.c: Return g729 and g723.1 frames with the
+ number of samples set properly. If the wctc4xxp returns more than
+ a single packet, we need to update the number of samples in the
+ returned frame accordingly. Acked-by: Shaun Ruffell
+ <sruffell at digium.com> ........ Merged revisions 358484 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-07 15:17 +0000 [r358436-358441] Terry Wilson <twilson at digium.com>
+
+ * /, configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in
+ cdr_adaptive_odbc.conf.sample ........ Merged revisions 358438
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * cel/cel_odbc.c, /, cdr/cdr_adaptive_odbc.c: Add detection for
+ ODBC WCHAR fields Without detecting these types, cel_odbc blows
+ up when the character set for the table is utf8. This also wraps
+ cdr_adaptive_odbc's use of those types in the HAVE_ODBC_WCHAR
+ #ifdef seen in other parts of the code. ........ Merged revisions
+ 358435 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-06 17:46 +0000 [r358261-358378] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Fix ring cadance setup for outgoing
+ calls on FXS ports. * Fix referencing the wrong variable in
+ chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for
+ compiling with -Wshadow and finding this bug. ........ Merged
+ revisions 358377 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/sig_ss7.c: Drop SS7 call if not connected yet when
+ INCOMPLETE/BUSY/CONGESTION. SS7 is a trunk protocol and should
+ clear a failed call as soon as possible. * Made SS7 hangup a call
+ immediately if it has not connected yet for
+ INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate
+ inband tone. (closes issue ASTERISK-19372) Reported by: Igor
+ Nikolaev ........ Merged revisions 358278 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c:
+ Setup DSP when SS7 call is connected or early media is available.
+ Outgoing SS7 calls fail to detect incoming DTMF so any bridged
+ channel that requires out-of-band DTMF will not work. * Added
+ sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
+ The new call converts conditionaled out unconverted code and
+ shows that the code really did something useful. * Improved some
+ chan_dahdi DTMF debug messages to help track DTMF handling.
+ (closes issue ASTERISK-19312) Reported by: Igor Nikolaev ........
+ Merged revisions 358260 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-05 18:58 +0000 [r358215] Jonathan Rose <jrose at digium.com>
+
+ * main/manager.c, /: Eliminate double close of file descriptor in
+ manager.c The process_output function in manager.c attempted to
+ call fclose and close immediately afterwards. Since fclose
+ implies close, this resulted in a potential double free on file
+ descriptors. This patch changes that behavior and also adds error
+ checking to fclose and close depending on which was deemed
+ necessary. Also error messages. Thanks to Rosen Iliev for
+ pointing out the location of the problem. (closes issue
+ ASTERISK-18453) Reported By: Jaco Kroon Review:
+ https://reviewboard.asterisk.org/r/1793/ ........ Merged
+ revisions 358214 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-05 16:42 +0000 [r358163] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Defer sending the connected line reinvite
+ if a reinvite is already in progress. (issue ASTERISK-19355)
+ Reported by: tomaso (closes issue AST-825) ........ Merged
+ revisions 358162 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-05 15:59 +0000 [r358116] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: Ensure Asterisk acknowledges ACKs to 4xx
+ on Replaces errors Asterisk was not setting pendinginvite in the
+ upper half of handle_request_invite such that the 4xx was
+ retransmitted repeatedly even though an ack was received for
+ every retransmission. (closes issue ASTERISK-19303) Patch-by:
+ Jeremiah Gowdy ........ Merged revisions 358115 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-02 23:28 +0000 [r357987-358033] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_usbradio.c, /, channels/xpmr/xpmr.c: Fix
+ unused-but-set-variable warnings All of these were pretty
+ obviously unused. Some were unused because the code that used
+ them was #if 0'd. In those cases, I just commented out the
+ unused-but-set variables. ........ Merged revisions 358029 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c,
+ channels/misdn/isdn_lib.c: Correct some set-but-unused variable
+ warnings in the mISDN library. (from kpfleming's commit to trunk
+ r356292) ........ Merged revisions 358011 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/xpmr/xpmr.c: Make chan_usbradio compile under dev
+ mode x=++x and x=x=1? Really? ........ Merged revisions 357986
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-02 21:03 +0000 [r357941] Kinsey Moore <kmoore at digium.com>
+
+ * /, main/ccss.c, tests/test_event.c, main/event.c,
+ include/asterisk/strings.h: Fix case-sensitivity for
+ device-specific event subscriptions and CCSS This change fixes
+ case-sensitivity for device-specific subscriptions such that the
+ technology identifier is case-insensitive while the remainder of
+ the device string is still case-sensitive. This should also
+ preserve the original case of the device string as passed in to
+ the event system. CCSS is the only feature affected as it is the
+ only consumer of device-specific event subscriptions. The second
+ part of this patch addresses similar case-sensitivity issues
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