[svn-commits] bebuild: tag 1.8.12.0-rc1 r361157 - /tags/1.8.12.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Apr 4 13:16:46 CDT 2012
Author: bebuild
Date: Wed Apr 4 13:16:40 2012
New Revision: 361157
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=361157
Log:
Importing files for 1.8.12.0-rc1 release.
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tags/1.8.12.0-rc1/.version (with props)
tags/1.8.12.0-rc1/ChangeLog (with props)
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+2012-04-04 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.12.0-rc1 Released.
+
+2012-04-04 16:29 +0000 [r361090-361142] Jonathan Rose <jrose at digium.com>
+
+ * main/app.c, pbx/pbx_realtime.c, apps/app_externalivr.c,
+ channels/chan_iax2.c, apps/app_milliwatt.c, main/channel.c,
+ pbx/pbx_loopback.c, addons/chan_ooh323.c, channels/chan_sip.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)
+
+ * 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/
+
+2012-04-03 20:08 +0000 [r360987-361040] 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)
+
+ * 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)
+
+2012-03-30 21:26 +0000 [r360933] 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
+
+2012-03-29 23:32 +0000 [r360862-360884] 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)
+
+ * 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)
+
+2012-03-28 19:06 +0000 [r360712] Terry Wilson <twilson at digium.com>
+
+ * cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c,
+ channels/chan_gtalk.c, channels/chan_jingle.c,
+ addons/chan_ooh323.c: Destroy configs when they are no longer
+ used https://reviewboard.asterisk.org/r/1834/
+
+2012-03-27 16:59 +0000 [r360625] 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.
+
+2012-03-27 14:32 +0000 [r360488-360574] 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
+
+ * 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)
+
+2012-03-26 18:37 +0000 [r360471-360474] Paul Belanger <pabelanger at digium.com>
+
+ * CHANGES: Update CHANGES for r360471
+
+ * CHANGES: Fix Asterisk version typo
+
+ * 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.
+
+2012-03-24 23:46 +0000 [r360356-360413] Russell Bryant <russell at russellbryant.com>
+
+ * funcs/func_curl.c: func_curl: Fix leak of an ast_str in error
+ handling code path.
+
+ * 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.
+
+ * apps/app_jack.c: app_jack: fix datastore memory leak in error
+ handling path.
+
+ * res/ael/ael.tab.h, main/ast_expr2.c, main/ast_expr2.h,
+ res/ael/ael.tab.c, main/ast_expr2f.c, res/ael/ael_lex.c: 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.
+
+ * main/ast_expr2.y: 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.
+
+2012-03-24 00:35 +0000 [r360262-360309] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c, main/channel.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.
+
+ * channels/chan_sip.c: Add missing initialization of
+ update_redirecting in chan_sip.c
+
+2012-03-21 14:51 +0000 [r360138] 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)
+
+2012-03-21 13:19 +0000 [r360087] 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>
+
+2012-03-21 13:19 +0000 [r360086] 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
+
+2012-03-20 20:32 +0000 [r360033] 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.
+
+2012-03-20 17:21 +0000 [r359979] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/manager.h, main/manager.c: 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/
+
+2012-03-16 20:13 +0000 [r359892] 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/
+
+2012-03-16 08:22 +0000 [r359809] 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/
+
+2012-03-15 19:01 +0000 [r359656-359706] 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
+
+ * 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
+
+2012-03-15 18:17 +0000 [r359609] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_queue.c, apps/app_dial.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/
+
+2012-03-15 00:52 +0000 [r359452-359558] 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.
+
+ * channels/chan_gtalk.c: chan_gtalk: Fix use of uninitialized vars
+ in config handling. Fix potential use of context, parkinglot, and
+ prefs before they are initialized.
+
+ * 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.
+
+ * 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.
+
+ * 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).
+
+ * 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.
+
+2012-03-14 22:20 +0000 [r359451] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/channel.h, main/channel.c,
+ channels/chan_agent.c: 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.
+
+2012-03-14 17:32 +0000 [r359356] 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)
+
+2012-03-14 17:17 +0000 [r359344] 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/
+
+2012-03-14 10:52 +0000 [r359050-359259] 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.
+
+ * 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.
+
+ * main/sched.c: Fix inaccurate sizeof() in sched.c. This code just
+ needed sizeof(int), not sizeof(int *).
+
+ * utils/astman.c: Fix incorrect sizeof() in astman.
+
+ * 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.
+
+ * res/res_adsi.c: Fix broken usage of sizeof() in res_adsi.
+
+ * 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.
+
+ * main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final().
+
+ * 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.
+
+ * res/ais/ais.h, res/res_ais.c, res/ais/clm.c, res/ais/evt.c: Dump
+ cache of published events when a node joins the cluster. Also use
+ a more reliable method for stopping the poll() thread.
+
+ * makeopts.in, apps/app_rpt.c (removed), channels/chan_usbradio.c
+ (removed), channels/xpmr (removed),
+ build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: 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/
+
+2012-03-13 20:31 +0000 [r358943-358978] 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/
+
+ * main/devicestate.c, include/asterisk/devicestate.h,
+ channels/chan_sip.c, tests/test_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/
+
+2012-03-13 16:54 +0000 [r358810-358859] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * UPGRADE.txt, CHANGES: Requested changes documenting the fixed AEL
+ functionality.
+
+ * utils/ael_main.c, apps/app_stack.c, utils/conf2ael.c,
+ res/ael/pval.c, funcs/func_dialplan.c, tests/test_gosub.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/
+
+2012-03-08 16:39 +0000 [r358643] 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
+
+2012-03-07 18:25 +0000 [r358530] 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)
+
+2012-03-07 16:11 +0000 [r358484] 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>
+
+2012-03-07 15:16 +0000 [r358435-358438] Terry Wilson <twilson at digium.com>
+
+ * configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in
+ cdr_adaptive_odbc.conf.sample
+
+ * cdr/cdr_adaptive_odbc.c, cel/cel_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.
+
+2012-03-06 17:44 +0000 [r358260-358377] 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.
+
+ * 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
+
+ * channels/sig_ss7.c, channels/chan_dahdi.c, channels/sig_ss7.h:
+ 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
+
+2012-03-05 18:49 +0000 [r358214] 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/
+
+2012-03-05 16:41 +0000 [r358162] 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)
+
+2012-03-05 15:54 +0000 [r358115] 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
+
+2012-03-02 23:27 +0000 [r357986-358029] Terry Wilson <twilson at digium.com>
+
+ * channels/xpmr/xpmr.c, channels/chan_usbradio.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.
+
+ * channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c,
+ channels/chan_misdn.c: Correct some set-but-unused variable
+ warnings in the mISDN library. (from kpfleming's commit to trunk
+ r356292)
+
+ * channels/xpmr/xpmr.c: Make chan_usbradio compile under dev mode
+ x=++x and x=x=1? Really?
+
+2012-03-02 21:02 +0000 [r357940] Kinsey Moore <kmoore at digium.com>
+
+ * main/event.c, include/asterisk/strings.h, main/ccss.c,
+ tests/test_event.c: 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 within
+ CCSS itself that prevented it from functioning correctly after
+ the fix to the events system. This adds a unit test to verify
+ that the event system works as expected. (closes issue
+ ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/
+
+2012-03-02 18:34 +0000 [r357894] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c, main/channel.c: Remove ISDN hold restriction
+ for non-bridged calls. The check if an ISDN call is bridged
+ before it could be placed on hold is not necessary and is overly
+ restrictive. The check was originally done to prevent problems
+ with call transfers in case a user tried to transfer a call
+ connected to an application to another call connected to an
+ application. The ISDN transfer code has not required this
+ restriction for quite some time because ECT could transfer any
+ two active calls to each other. * Remove ISDN hold restriction
+ for calls connected to applications. * Made
+ ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
+ AST_CONTROL_UNHOLD instead of generating a warning message.
+ (closes issue ASTERISK-19388) Reported by: Birger Harzenetter
+ Tested by: rmudgett
+
+2012-03-02 15:58 +0000 [r357811] Sean Bright <sean at malleable.com>
+
+ * channels/chan_iax2.c: The default value for mohinterpret is the
+ empty string, so when resetting to default values don't
+ explicitly set the value to "default."
+
+2012-03-02 15:45 +0000 [r357809] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_chanspy.c: Fix channel reference leak in ChanSpy. * Fix
+ next_channel() channel reference leak in ChanSpy. (closes issue
+ ASTERISK-19461) Reported by: Irontec Patches:
+ app_chanspy_iteartor_next_unref.patch (license #6213) patch
+ uploaded by Irontec (issue ASTERISK-17515)
+
+2012-03-02 00:59 +0000 [r357760-357761] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Fix race condition that can cause important
+ control frames (such as a hangup) to be missed. This takes two
+ actions. 1. Move the reading of the alertpipe in __ast_read() to
+ immediately before the removal of frames from the readq. This
+ means we won't do something silly like read from the alertpipe,
+ then ignore the fact that there's a frame to get from the readq
+ since channel's fdno is the AST_TIMING_FD. 2. When
+ ast_settimeout() sets the rate to 0 and the timingfunc to NULL,
+ if the channel's fdno is the AST_TIMING_FD, then set the fdno to
+ -1. This is because if the rate is 0 and the timingfunc is NULL,
+ it means that the channel's timing fd is being invalidated, so
+ any pending reads should not occur. This may actually solve more
+ issues than the referenced one below, but it's not known at this
+ time for sure. (closes issue ASTERISK-19223) reported by
+ Frank-Michael Wittig Review:
+ https://reviewboard.asterisk.org/r/1779
+
+ * main/translate.c: Second attempt to get optimal translation paths
+ when codec_resample is used. This borrows code heavily from
+ changes made in translation code in Asterisk 10. This uses the
+ quality and sample rate change of translation in order to pick
+ paths rather than the computational cost of translations.
+ Computational cost is used solely in determining if a single
+ translation step from a specific translator is better than the
+ same translation step provided by a different translator. (closes
+ issue ASTERISK-16821) reported by Andrew Lindh Review:
+ https://reviewboard.asterisk.org/r/1772
+
+2012-03-01 14:18 +0000 [r357665] Kinsey Moore <kmoore at digium.com>
+
+ * main/acl.c: Prevent outbound SIP NOTIFY packets from displaying a
+ port of 0 In the change from 1.6.2 to 1.8, ast_sockaddr was
+ introduced which changed the behavior of ast_find_ourip such that
+ port number was wiped out. This caused the port in internip
+ (which is used for Contact and Call-ID on NOTIFYs) to be 0. This
+ change causes ast_find_ourip to be port-preserving again. (closes
+ issue ASTERISK-19430)
+
+2012-02-29 19:41 +0000 [r357575] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * apps/app_dial.c: Fix copying of CDR(accountcode) to local
+ channels. In r203638, during the addition of the Channel Event
+ Logging, in mid-2009, this got broken in trunk and ended up in
+ asterisk 1.8 and higher. This fixes so the CDR(accountcode) from
+ the calling channel is available to dialed channels again as well
+ as showing up properly in the CDR's. (closes issue
+ ASTERISK-19384) Patches: accountcode.patch (License #6033) by
+ jamicque Review: https://reviewboard.asterisk.org/r/1775/
+ Reviewed by: Richard Mudgett
+
+2012-02-28 22:27 +0000 [r357455-357490] Jonathan Rose <jrose at digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample: Adding transport=udp to
+ sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE
+ (issue ASTERISK-19352) Reported by: jamicque Patches:
+ asterisk-19352-transport-warning-message-v1.patch uploaded by
+ Michael L. Young (license 5026)
+
+ * cdr/cdr_adaptive_odbc.c: Add additional character type types to
+ supported data types for cdr_adaptive_odbc The reporter was uable
+ to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so this
+ patch adds those along with some other character types to the
+ list of types cdr_adaptive_odbc will work using the varchar
+ conditions. The problem wasn't really UTF8 characters as much as
+ it was a failure to respond to the exact type that was
+ declared/in use on that database. (closes issue ASTERISK-19334)
+ Reported By: Igor Nikolaev Patches: cdr_adaptive_odbc.patch
+ uploaded by Igor Nikolaev (license 6236)
+
+2012-02-28 21:19 +0000 [r357416] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * apps/app_stack.c: Correctly reset the dialplan priority. When the
+ stack frame is allocated, we save the address to which we should
+ return, when the Gosub returns. However, if we just want to
+ restore the priority, then we need to subtract 1 before setting
+ it. Otherwise, when a Gosub goes to a nonexistent address, it
+ will skip a priority in the dialplan. This is because when we
+ return from an application, the PBX increments the priority for
+ us.
+
+2012-02-28 20:57 +0000 [r357407] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c: Use more reasonable cause code when rejecting
+ incoming call waiting calls. (closes issue ASTERISK-19397)
+ Reported by: Birger Harzenetter Patches: nochannel-cause.patch
+ (license #5870) patch uploaded by Birger Harzenetter
+
+2012-02-28 20:26 +0000 [r357356-357386] Jonathan Rose <jrose at digium.com>
+
+ * UPGRADE.txt: Moves UPGRADE.txt notes from r357356 to a new
+ section specific to 1.8.12 (issue ASTERISK-19352) reported by:
+ jamicque
+
+ * UPGRADE.txt: Adds UPGRADE.txt notes to r357266 indicating changes
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