[asterisk-commits] bebuild: tag 10.4.0-digiumphones-rc1 r365240 - /tags/10.4.0-digiumphones-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu May 3 14:33:34 CDT 2012
Author: bebuild
Date: Thu May 3 14:33:29 2012
New Revision: 365240
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=365240
Log:
Importing files for 10.4.0-digiumphones-rc1 release.
Added:
tags/10.4.0-digiumphones-rc1/.lastclean (with props)
tags/10.4.0-digiumphones-rc1/.version (with props)
tags/10.4.0-digiumphones-rc1/ChangeLog (with props)
Added: tags/10.4.0-digiumphones-rc1/.lastclean
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+2012-05-03 Asterisk Development Team <asteriskteam at digium.com>
+
+ * 10.4.0-digiumphones-rc1 Released.
+
+2012-05-03 19:14 +0000 [r365214] Jason Parker <jparker at digium.com>
+
+ * main/app.c, pbx/pbx_realtime.c, apps/app_externalivr.c,
+ channels/chan_iax2.c, res/res_fax_spandsp.c,
+ apps/app_milliwatt.c, main/channel.c, pbx/pbx_loopback.c,
+ addons/chan_ooh323.c, main/manager.c, /, channels/chan_sip.c,
+ channels/chan_skinny.c: Multiple revisions
+ 361143,363103-363104,363107,363156,364707 ........ r361143 |
+ jrose | 2012-04-04 11:38:12 -0500 (Wed, 04 Apr 2012) | 12 lines
+ 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) ........ r363103 | mjordan | 2012-04-23 08:40:23
+ -0500 (Mon, 23 Apr 2012) | 19 lines AST-2012-005: Fix remotely
+ exploitable heap overflow in keypad button handling When handling
+ a keypad button message event, the received digit is placed into
+ a fixed length buffer that acts as a queue. When a new message
+ event is received, the length of that buffer is not checked
+ before placing the new digit on the end of the queue. The
+ situation exists where sufficient keypad button message events
+ would occur that would cause the buffer to be overrun. This patch
+ explicitly checks that there is sufficient room in the buffer
+ before appending a new digit. (closes issue ASTERISK-19592)
+ Reported by: Russell Bryant ........ Merged revisions 363100 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 363102 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363104 | mjordan | 2012-04-23 08:48:48 -0500 (Mon, 23 Apr 2012)
+ | 10 lines Reference skinny_subchannel object instead of
+ skinny_device for r363103 The check-in to resolve ASTERISK-19592
+ (r363103) failed to switch to the skinny_subchannel object
+ instead of the skinny_device when attempting to reference the
+ buffer for the keypad digits. This patch fixes that. (issue
+ ASTERISK-19592) Reported by: Russell Bryant ........ r363107 |
+ mjordan | 2012-04-23 09:07:29 -0500 (Mon, 23 Apr 2012) | 19 lines
+ AST-2012-006: Fix crash in UPDATE handling when no channel owner
+ exists If Asterisk receives a SIP UPDATE request after a call has
+ been terminated and the channel has been destroyed but before the
+ SIP dialog has been destroyed, a condition exists where a
+ connected line update would be attempted on a non-existing
+ channel. This would cause Asterisk to crash. The patch resolves
+ this by first ensuring that the SIP dialog has an owning channel
+ before attempting a connected line update. If an UPDATE request
+ is received and no channel is associated with the dialog, a 481
+ response is sent. (closes issue ASTERISK-19770) Reported by:
+ Thomas Arimont Tested by: Matt Jordan Patches:
+ ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license
+ 6283) ........ Merged revisions 363106 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363156 | jrose | 2012-04-23 09:39:48 -0500 (Mon, 23 Apr 2012) |
+ 23 lines AST-2012-004: Fix an error that allows AMI users to run
+ shell commands sans authorization. As detailed in the advisory,
+ AMI users without write authorization for SYSTEM class AMI
+ actions were able to run system commands by going through other
+ AMI commands which did not require that authorization.
+ Specifically, GetVar and Status allowed users to do this by
+ setting their variable/s options to the SHELL or EVAL functions.
+ Also, within 1.8, 10, and trunk there was a similar flaw with the
+ Originate action that allowed users with originate permission to
+ run MixMonitor and supply a shell command in the Data argument.
+ That flaw is fixed in those versions of this patch. (closes issue
+ ASTERISK-17465) Reported By: David Woolley Patches:
+ 162_ami_readfunc_security_r2.diff uploaded by jrose (license
+ 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose (license
+ 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose (license
+ 6182) ........ Merged revisions 363117 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 363141 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364707 | mmichelson | 2012-04-30 14:42:35 -0500 (Mon, 30 Apr
+ 2012) | 17 lines Revert improved identities sent in dialog-info
+ NOTIFY requests in r360862 Revision 360862 was intended to
+ improve identities sent in dialog-info NOTIFY requests. Some
+ users reported that hint became broken once this was done. It's
+ not clear exactly what part of the patch has caused this
+ regression, but broken hints are bad. For now, this revision is
+ being reverted so that the next releases of Asterisk do not have
+ bad behavior in them. The original reported issue will have to be
+ fixed differently in the next version of Asterisk. (issue
+ ASTERISK-16735) ........ Merged revisions 364706 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361143,363103-363104,363107,363156,364707 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-02 15:57 +0000 [r364766-365009] Jason Parker <jparker at digium.com>
+
+ * channels/chan_sip.c: Save the address on which a MESSAGE was
+ received, so it can be used in MESSAGE() This is useful in cases
+ where chan_sip may be listening on multiple addresses. (issue
+ DPMA-127)
+
+ * main/manager.c, /: Prevent a potential crash when using manager
+ hooks. Found by me while poking at DPMA-127. ........ Merged
+ revisions 364841 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364842 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * tests/test_voicemail_api.c, include/asterisk/app_voicemail.h,
+ apps/app_voicemail.c, /: Remove folder_dir from voicemail
+ snapshots API. It was both unused (except in tests, where it was
+ fudged) and unnecessary. (closes issue AST-842) ........ Merged
+ revisions 364761 from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+2012-04-19 20:35 +0000 [r362674] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: The 10-digiumphones version of revision
+ 362673 to 1.8-digiumphones
+
+2012-04-13 17:48 +0000 [r362134] Jason Parker <jparker at digium.com>
+
+ * /: init merge prop
+
+2012-04-13 17:29 +0000 [r362133] Matthew Jordan <mjordan at digium.com>
+
+ * /: Add branch-10-merged property
+
+2012-04-09 21:32 +0000 [r361748] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.exports.in, tests/test_voicemail_api.c
+ (added), include/asterisk/app_voicemail.h, apps/app_voicemail.c,
+ main/manager.c: Fix bugs in voicemail APIs and add unit tests.
+
+2012-04-06 19:16 +0000 [r361503] Richard Mudgett <rmudgett at digium.com>
+
+ * main/message.c: Update Func MESSAGE() and AMI MessageSend
+ documentation. * Document MESSAGE(custom_data) * Update AMI
+ MessageSend documentation * Eliminate a shadowed variable name in
+ msg_func_write() for custom_data.
+
+2012-04-05 17:48 +0000 [r361140-361293] Mark Michelson <mmichelson at digium.com>
+
+ * tests/test_config.c, funcs/func_presencestate.c: Add unit tests
+ for func_presencestate and config hooks. These tests were
+ originally written for the trunk merge of Digium phone support to
+ Asterisk, but this branch can benefit from these tests too.
+
+ * main/pbx.c: Fix a compiler error in pbx.c This places a function
+ declaration before its first invocation.
+
+ * main/pbx.c: Only add the device portion of a hint when adding a
+ hintdevice.
+
+ * main/file.c, apps/app_queue.c, apps/app_mixmonitor.c,
+ main/message.c, apps/app_voicemail.c, channels/chan_sip.c,
+ channels/chan_skinny.c: Fix compiler errors. Next up is to do
+ some testing to be sure things are sane.
+
+ * channels/sip/include/sip.h, channels/chan_sip.c,
+ channels/chan_skinny.c: Changes to the channels directory for
+ Digium phone support. Next step: Get this poopoo compiling!
+
+ * tests/test_config.c (added), tests/test_custom_control.c (added):
+ Add tests for Digium phone support.
+
+ * apps/app_voicemail.exports.in, apps/app_queue.c,
+ apps/app_mixmonitor.c, apps/app_voicemail.c: apps changes
+ necessary for Digium phone support.
+
+ * funcs/func_frame_trace.c, funcs/func_presencestate.c (added): Add
+ Digium phone changes for the funcs directory.
+
+ * include/asterisk/custom_control_frame.h (added),
+ include/asterisk/message.h, include/asterisk/config.h,
+ include/asterisk/presencestate.h (added),
+ include/asterisk/app_voicemail.h (added), include/asterisk/pbx.h,
+ include/asterisk/frame.h, include/asterisk/manager.h,
+ include/asterisk/file.h, include/asterisk/app.h,
+ include/asterisk/event_defs.h: Add Digium phone changes for the
+ include/asterisk directory
+
+ * main/message.c: Add phone changes for message.c
+
+ * main/features.c, main/file.c, main/app.c, main/event.c,
+ main/custom_control_frame.c (added), main/asterisk.c,
+ main/config.c, main/presencestate.c (added), main/channel.c,
+ main/pbx.c, main/manager.c: Update everything in the main
+ directory for Digium phone additions.
+
+ * configs/manager.conf.sample: Update sample config files. This
+ just involved updating manager.conf.sample to have the new
+ "message" write permission.
+
+ * / (added): Create a Digium phones branch of Asterisk 10.
+
+2012-03-29 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.3.0 Released.
+
+2012-03-26 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.3.0-rc3 Released.
+
+ * AST-2012-003
+
+ * AST-2012-002
+
+ * /main/manager.c, include/asterisk/manager.h: Fix AMI deadlock
+ regression by allowing AMI action callback to be reentrant
+
+ Fix AMI module reload deadlock from ASTERISK-18479 when it tired to
+ fix the race between calling an AMI action callback and
+ unregistering that action. Refixes ASTERISK-13874 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 path 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/
+
+2012-03-06 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.3.0-rc2 Released.
+
+ * 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.
+
+2012-01-30 22:16 +0000 [r353369-353321] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/sip/include/dialog.h, /, channels/chan_sip.c,
+ channels/sip/include/sip.h: Merged revisions 353320 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31
+ Jan 2012) | 18 lines RFC3261 Section 8.1.1.5. The sequence number
+ value MUST be expressible as a 32-bit unsigned integer * fix: use
+ %u instead of %d when dealing with CSeq numbers - to remove
+ possibility of -ve numbers. * fix: change all uses of seqno and
+ friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
+ Summary of CSeq numbers. An initial CSeq number must be less than
+ 2^31 A CSeq number can increase in value up to 2^32-1 An
+ incrementing CSeq number must not wrap around to 0. Tested with
+ Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
+ Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1699/ ........
+
+ * /, channels/chan_sip.c: Merged revisions 353368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31 Jan
+ 2012) | 2 lines prevent debug messsges displaying -ve Cseq
+ numbers. Missed in R353320 ........
+
+2012-01-30 23:28 +0000 [r353397] Terry Wilson <twilson at digium.com>
+
+ * main/dnsmgr.c, /, channels/chan_sip.c, include/asterisk/dnsmgr.h:
+ Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr
+ currently takes a pointer to an ast_sockaddr and updates it
+ anytime an address resolves to something different. There are a
+ couple of issues with this. First, the ast_sockaddr is usually
+ the address of an ast_sockaddr inside a refcounted struct and we
+ never bump the refcount of those structs when using dnsmgr. This
+ makes it possible that a refresh could happen after the
+ destructor for that object is called (despite ast_dnsmgr_release
+ being called in that destructor). Second, the module using dnsmgr
+ cannot be aware of an address changing without polling for it in
+ the code. If an action needs to be taken on address update (like
+ re-linking a SIP peer in the peers_by_ip table), then polling for
+ this change negates many of the benefits of having dnsmgr in the
+ first place. This patch adds a function to the dnsmgr API that
+ calls an update callback instead of blindly updating the address
+ itself. It also moves calls to ast_dnsmgr_release outside of the
+ destructor functions and into cleanup functions that are called
+ when we no longer need the objects and increments the refcount of
+ the objects using dnsmgr since those objects are stored on the
+ ast_dnsmgr_entry struct. A helper function for returning the
+ proper default SIP port (non-tls vs tls) is also added and used.
+ This patch also incorporates changes from a patch posted by Timo
+ Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue
+ ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/
+ ........ Merged revisions 353371 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-31 17:21 +0000 [r353463] Richard Mudgett <rmudgett at digium.com>
+
+ * main/manager.c, /, include/asterisk/channel.h: Fix memory leak in
+ error paths for action_originate(). * Fix memory leak of vars in
+ error paths for action_originate(). * Moved struct
+ fast_originate_helper tech and data members to stringfields. *
+ Simplified ActionID header handling for fast_originate(). * Added
+ doxygen note to ast_request() and ast_call() and the associated
+ channel callbacks that the data/addr parameters should be treated
+ as const char *. Review: https://reviewboard.asterisk.org/r/1690/
+ ........ Merged revisions 353454 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-01 00:00 +0000 [r353503] Terry Wilson <twilson at digium.com>
+
+ * res/res_calendar.c, /: Allow res_calendar to be unloaded The
+ calendaring tech modules depend on res_calendar and initially
+ res_calendar just bumped the use count so that it couldn't be
+ unloaded. res_calendar can potentially create many threads and
+ I've seen issues where the Asterisk shutdown has failed where it
+ looked like these threads could be the culprit. This patch adds
+ unload support for res_calendar. Unloading res_calendar will also
+ unload the dependant tech modules as well. (closes issue
+ ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/
+ ........ Merged revisions 353502 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-01 15:05 +0000 [r353551] Matthew Jordan <mjordan at digium.com>
+
+ * /, contrib/init.d/etc_default_asterisk: Added clarification for
+ the VERBOSITY setting to etc_default_asterisk Clarified that
+ using the VERBOSITY setting in etc_default_asterisk is the same
+ as using the -v command line switch, which causes Asterisk to
+ launch in console mode. (closes issue ASTERISK-17030) Reported
+ by: Jonas ........ Merged revisions 353550 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-01 15:51 +0000 [r353599] Sean Bright <sean at malleable.com>
+
+ * /, include/asterisk/audiohook.h: Resolve an overlap in the
+ ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and
+ AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused
+ unintended side effects. This patch moves
+ AST_AUDIOHOOK_TRIGGER_WRITE, and updates
+ AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention.
+ This will affect existing modules that use these flags, so be
+ sure to recompile as necessary. (closes issue ASTERISK-19246)
+ Reported by: feyfre ........ Merged revisions 353598 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-01 21:16 +0000 [r353771-353721] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers
+ for various functions in chan_sip There are a number of cleaner
+ looking wrappers for ast_sockaddr_stringify_fmt available which
+ are slightly more readable than using a direct call to
+ ast_sockaddr_stringify_fmt. This patch switches a number of those
+ calls in chan_sip to use those wrappers and is generally
+ harmless. (Closes issue ASTERISK-16930) Reported by: Michael L.
+ Young Patches: chan_sip-broken-registration-1.8.diff uploaded by
+ Michael L. Young (license 5026) ........ Merged revisions 353720
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Fix sip show peers port output, align
+ columns, and fix ami port output. A previous patch I committed
+ from ASTERISK-16930 unexpectedly changed some output for the AMI
+ action "sippeers" which this patch changes back. Also, this
+ aligns the output for the cli command "sip show peers" and fixes
+ another issue that patch introduced by using
+ ast_sockaddr_stringify calls multiple times without immediately
+ using the pointer. I also went ahead and did a little janitorial
+ work to clean up whitespace in _sip_show_peers. (issue
+ ASTERISK-16930) (closes issue ASTERISK-19281) Reported by:
+ Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by
+ Walter Doekes (license 5674) ........ Merged revisions 353769
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-02 18:48 +0000 [r353820] Mark Michelson <mmichelson at digium.com>
+
+ * configs/http.conf.sample, main/manager.c, /, main/http.c,
+ configs/manager.conf.sample, include/asterisk/manager.h: Fix TLS
+ port binding behavior as well as reload behavior: * Removes
+ references to tlsbindport from http.conf.sample and
+ manager.conf.sample * Properly bind to port specified in
+ tlsbindaddr, using the default port if specified. * On a reload,
+ properly close socket if the service has been disabled. A note
+ has been added to UPGRADE.txt to indicate how ports must be set
+ for TLS. (closes issue ASTERISK-16959) reported by Olaf
+ Holthausen (closes issue ASTERISK-19201) reported by Chris
+ Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas
+ Review: https://reviewboard.asterisk.org/r/1709 ........ Merged
+ revisions 353770 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-02 20:11 +0000 [r353868] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c:
+ Restore the 'w' modifier support for ISDN spans.
+ Dial(DAHDI/g0/1234w888) This feature also causes the sending
+ complete ie to be sent for switch types that do not automatically
+ send the ie. (EuroISDN/ETSI) The main difference between dialing
+ Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the
+ sending of the sending complete ie. (closes issue ASTERISK-19176)
+ Reported by: rmudgett Tested by: rmudgett ........ Merged
+ revisions 353867 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-02 22:27 +0000 [r353916] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: Ensure entering T.38 passthrough does not
+ cause an infinite loop After R340970 Asterisk was still polling
+ the RTCP file descriptor after RTCP is shut down and removed. If
+ the descriptor happened to have data ready when the removal
+ occured then Asterisk would go into an infinite loop trying to
+ read data that it can never actually access. This change disables
+ the audio RTCP file descriptor for the duration of the T.38
+ transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan
+ Vrban ........ Merged revisions 353915 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-03 16:22 +0000 [r354000-353962] Jonathan Rose <jrose at digium.com>
+
+ * res/res_fax.c: Fixes a segfault occuring when performing attended
+ transfer with FAXOPT(gateway)=yes (closes issue ASTERISK-19184)
+ Reported by: Alexandr
+
+ * /, channels/chan_agent.c: Fixes deadlocks occuring in chan_agent
+ due to r335976 Bad locking order was added to chan_agent to
+ prevent segfaults from having no locking in a patch by irroot.
+ This patch addresses the bad locking order by releasing locks
+ before getting the right locking order to stop deadlocks from
+ occuring when doing multiple interactions with agents. (closes
+ issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review:
+ https://reviewboard.asterisk.org/r/1708/ ........ Merged
+ revisions 353999 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-06 17:31 +0000 [r354217-354119] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/features.c: Add missing headers to AMI UnParkedCall event
+ to uniquely identify the call. The AMI UnParkedCall event was
+ missing the Parkinglot and Uniqueid headers that the AMI
+ ParkedCall event contains. (closes issue ASTERISK-19240) Reported
+ by: Michael Yara ........ Merged revisions 354116 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, pbx/pbx_config.c: Improved documentation of CLI "dialplan add
+ extension" command. * Documented dialplan add extension
+ <exten>,<priority>,<app(<app-data>)> format. * Allow acceptance
+ of command without the app-data value. There are many
+ applications that do no need any parameters so it is silly to
+ require that field for all commands. * Fixed a couple
+ ast_malloc/ast_free mismatches with ast_add_extension2() calls.
+ (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested
+ by: rmudgett ........ Merged revisions 354216 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-07 15:19 +0000 [r354270] Jonathan Rose <jrose at digium.com>
+
+ * /, cdr/cdr_pgsql.c: Fix column duplication bug in module reload
+ for cdr_pgsql. Prior to this patch, attempts to reload
+ cdr_pgsql.so would cause the column list to keep its current data
+ and then add a second copy during the reload. This would cause
+ attempts to log the CDR to the database to fail. This patch also
+ cleans up some unnecessary null checks for ast_free and deals
+ with a few potential locking problems. (closes issue
+ ASTERISK-19216) Reported by: Jacek Konieczny Review:
+ https://reviewboard.asterisk.org/r/1711/ ........ Merged
+ revisions 354263 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-07 21:17 +0000 [r354349] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql:
+ Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing
+ instead of "" 2. Don't set ipaddr or port to the string "(null)"
+ when they are empty 3. Add missing required fields, set default
+ for lastms to 0, and modify the length of the ipaddr field to 45
+ in the Postgresql realtime.sql file. (closes issue
+ ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/
+ ........ Merged revisions 354348 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 02:25 +0000 [r354493] Russell Bryant <russell at russellbryant.com>
+
+ * main/channel.c, /: Remove some unnecessary locking from
+ ast_hangup(). This patch removes some unnecessary locking of the
+ channels container in ast_hangup(). The reason this came up is
+ that this lock can very quickly block the entire system. If any
+ of the channel cleanup code decides to block, it causes a problem
+ for the whole system. For example, when audiohooks get destroyed,
+ if that blocks for a while waiting on the mixmonitor thread to
+ exit because it's busy blocking on some I/O, it causes a problem
+ for many other threads in the meantime. Review:
+ https://reviewboard.asterisk.org/r/1712/ ........ Merged
+ revisions 354492 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 02:54 +0000 [r354496] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_parkandannounce.c, /: Fix crash in ParkAndAnnounce.
+ Well, thats embarrasing. I forgot to initialize the caller_id
+ storage. (closes issue ASTERISK-19311) Reported by: tootai Tested
+ by: rmudgett ........ Merged revisions 354495 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 16:35 +0000 [r354543] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: Fix SIP INFO DTMF handling for
+ non-numeric codes In ASTERISK-18924, SIP INFO DTMF handlingw as
+ changed to account for both lowercase alphatbetic DTMF events, as
+ well as uppercase alphabetic DTMF events. When this occurred, the
+ comparison of the character buffer containing the event code was
+ changed such that the buffer was first compared again '0' and '9'
+ to determine if it was numeric. Unfortunately, since the first
+ character in the buffer will typically be '1' in the case of
+ non-numeric event codes (10-16), this caused those codes to be
+ converted to a DTMF event of '1'. This patch fixes that, and
+ cleans up handling of both application/dtmf-relay and
+ application/dtmf content types. Review:
+ https://reviewboard.asterisk.org/r/1722/ (closes issue
+ ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan ........
+ Merged revisions 354542 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 17:04 +0000 [r354546] Mark Michelson <mmichelson at digium.com>
+
+ * /, res/res_fax.c: Adding reload support to res_fax.so (closes
+ issue ASTERISK-16712) reported by Frank DiGennaro Review:
+ https://reviewboard.asterisk.org/r/1713 ........ Merged revisions
+ 354545 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 17:08 +0000 [r354548] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: Clean-up of minor formatting issues in
+ r354542/3/4 rmudgett pointed out some formatting issues in the
+ check-in for ASTERISK-19290. This cleans those up. Review:
+ https://reviewboards.asterisk.org/r/1722/ ........ Merged
+ revisions 354547 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 19:54 +0000 [r354703-354656] Kinsey Moore <kmoore at digium.com>
+
+ * /, main/config.c: Make the config parser remove escaping
+ backslashes The config parser in Asterisk does not currently
+ remove a backslash that is used to escape a semicolon which would
+ otherwise be interpreted as the start of a comment. The change
+ here causes that backslash to be removed, but does not create a
+ real escape system in the config parser. The biggest complication
+ with a real escape system would be breaking existing configs
+ everywhere (parsing \\ as \ and breaking on escaped non-semicolon
+ characters) even though it would be the "right" way to do things.
+ (closes issue ASTERISK-17121) Review:
+ https://reviewboard.asterisk.org/r/1724/ ........ Merged
+ revisions 354655 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Fix parsing of SIP headers where compact
+ and non-compact headers are mixed Change parsing of SIP headers
+ so that compactness of the header no longer influences which
+ header will be chosen. Previously, a non-compact header would be
+ chosen instead of a preceeding compact-form header. (closes issue
+ ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/
+ ........ Merged revisions 354702 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 22:03 +0000 [r354750] Terry Wilson <twilson at digium.com>
+
+ * /, funcs/func_cdr.c: Note that CDRs are immutable once a bridge
+ is torn down CDRs cannot be modified after a bridge is torn down,
+ (e.g. after Dial() returns) even though the CDR() function may be
+ called. Since modifying the CDR code to change this behavior
+ could very easily break all kinds of things, this patch just
+ documents this limitation. (closes issues ASTERISK-16923) Review:
+ https://reviewboard.asterisk.org/r/1720/ ........ Merged
+ revisions 354749 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-10 18:05 +0000 [r354836] Richard Mudgett <rmudgett at digium.com>
+
+ * main/manager.c, /: Fix AMI Redirect ExtraChannel not redirecting
+ to the same exten and context. The astman_get_header() never
+ returns NULL so the check by the code for NULL would never fail.
+ (closes issue ASTERISK-16974) Reported by: Nuno Borges Patches:
+ 0018325.patch (license #6116) patch uploaded by Nuno Borges
+ (modified) ........ Merged revisions 354835 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-10 22:00 +0000 [r354890] Jason Parker <jparker at digium.com>
+
+ * apps/app_voicemail.c, /: Fix a voicemail memory leak with
+ heard/deleted messages. open_mailbox() was changed quite a long
+ time ago to allocate this memory. close_mailbox() should have
+ been changed to be responsible for freeing it. ........ Merged
+ revisions 354889 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-13 16:41 +0000 [r354938] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_confbridge.c: Don't try to play sound files that do not
+ exist. (closes issue ASTERISK-19188) Reported by: slesru
+
+2012-02-13 17:24 +0000 [r354959] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_config_pgsql.c, /, configs/extconfig.conf.sample: Fix
+ reconnecting to pgsql database after connection loss. There can
+ only be one database connection in res_config_pgsql just like
+ res_config_sqlite. If the connection is lost, the connection may
+ not get reestablished to the same database if the res_pgsql.conf
+ and extconfig.conf files are inconsistent. * Made only use the
+ configured database from res_pgsql.conf. * Fixed potential buffer
+ overwrite of last[] in config_pgsql(). (closes issue
+ ASTERISK-16982) Reported by: german aracil boned Review:
+ https://reviewboard.asterisk.org/r/1731/ ........ Merged
+ revisions 354953 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-13 19:51 +0000 [r355010] Joshua Colp <jcolp at digium.com>
+
+ * /, pbx/pbx_config.c: Only allow one 'dialplan reload' to execute
+ at a time as otherwise they would share the same common local
+ context list. (closes issue AST-758) ........ Merged revisions
+ 355009 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-13 22:03 +0000 [r355057] Richard Mudgett <rmudgett at digium.com>
+
+ * pbx/pbx_spool.c, /: Fix occasional incorrectly delayed call-file
+ execution. Since the dir timestamp is available at one second
+ resolution, we cannot know if it was updated within the same
+ second after we scanned it. Therefore, we will force another scan
+ if the dir was just modified. * Changed to force another scan if
+ the directory was just modified. (closes issue ASTERISK-19081)
+ Reported by: Knut Bakke Review:
+ https://reviewboard.asterisk.org/r/1688/ ........ Merged
+ revisions 355056 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 09:49 +0000 [r355137] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: call manager_event only if there is not
+ null channel structure (Closes issue ASTERISK-19298) Reported by:
+ robinfood Patches: issue19298.patch uploaded by may213 (License
+ #5415) ........ Merged revisions 355136 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 13:33 +0000 [r355183] Sean Bright <sean at malleable.com>
+
+ * /, channels/chan_iax2.c: Clear the high order bit from the
+ destination call number before sending. send_apathetic_reply
+ takes the incoming frame's source call number as the destination
+ call number for the outgoing frame. If the incoming frame was a
+ full frame, then the high order bit of the source call number is
+ set and will be interpreted as a retransmit when sent back out as
+ the destination call number. ........ Merged revisions 355182
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 15:53 +0000 [r355229] Jason Parker <jparker at digium.com>
+
+ * /, configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3
+ CDRs by default in sample configs. ........ Merged revisions
+ 355228 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 16:27 +0000 [r355271] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Properly invert the return of a strncmp
+ call. This was causing identification that should have been made
+ private to be public. (closes issue AST-814) reported by Patrick
+ Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson
+ (license 5430) ........ Merged revisions 355268 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 18:14 +0000 [r355375-355320] Richard Mudgett <rmudgett at digium.com>
+
+ * /, cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock
+ in cel_sqlite_custom reload. (closes issue ASTERISK-19356)
+ Reported by: Alex Villacis Lasso Patches:
+ asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch
+ (license #5617) patch uploaded by Alex Villacis Lasso Review:
+ https://reviewboard.asterisk.org/r/1740/ ........ Merged
+ revisions 355319 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ formats/format_ogg_vorbis.c: Fix voicemail problems when using
+ ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file
+ format because it did not implement the seek and tell format
+ callbacks among other problems. Since we were already using the
+ libvorbis and libvorbisenc libraries we can use libvorbisfile as
+ it is also part of the vorbis library package. * Made use the
+ libvorbisfile to handle the ogg/vorbis file stream. The
+ format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
+ (closes issue ASTERISK-16926) Reported by: sque Patches:
+ ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded
+ by sque ........ Merged revisions 355365 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-15 17:25 +0000 [r355530-355449] Sean Bright <sean at malleable.com>
+
+ * /, channels/chan_iax2.c: Use TRUNK_CALL_START as originally
+ intended. Back in r646, TRUNK_CALL_START was added and defined as
+ 0x4000. That same value was also hard-coded in one part of the
+ IAX2 code instead of using the #define. TRUNK_CALL_START has
+ changed over the years (for dealing with LOW_MEMORY), but the
+ hard-coded usage was never updated to match. This patch fixes
+ that. ........ Merged revisions 355448 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: Only use maxtrunkcall and
+ maxnontrunkcall in chan_iax2 if IAX_OLD_FIND is specified. These
+ variables are only accessed from the IAX_OLD_FIND path, so there
+ is no reason to keep them updated otherwise. ........ Merged
+ revisions 355458 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: When IAX2 debugging is enabled, make
+ sure to log 'apathetic' messages too. ........ Merged revisions
+ 355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-16 18:32 +0000 [r355620-355575] Richard Mudgett <rmudgett at digium.com>
+
+ * /, res/res_monitor.c: Fix AMI Monitor action without File header
+ converting channel name into filename. * Fix potential Solaris
+ crash if Monitor application has a urlbase and no fname_base
+ option. ........ Merged revisions 355574 from
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