--- /dev/null
+2012-12-10 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.12.0-digiumphones-rc1 Released.
+
+2012-12-03 20:25 +0000 [r377068-377134] Automerge script <automerge@asterisk.org>
+
+ * /: automerge cancel
+
+ * main/cli.c, main/cdr.c, /: Merged revisions 377070,377074 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r377070 | rmudgett | 2012-12-03 12:41:28 -0600
+ (Mon, 03 Dec 2012) | 15 lines Cleanup CDR resources on exit. *
+ Simplify do_reload() return handling since it never returned
+ anything other than 0. (issue ASTERISK-20649) Reported by: Corey
+ Farrell Patches: cdr-cleanup-1_8.patch (license #5909) patch
+ uploaded by Corey Farrell cdr-cleanup-10-11-trunk.patch (license
+ #5909) patch uploaded by Corey Farrell Modified ........ Merged
+ revisions 377069 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r377074 | rmudgett | 2012-12-03 13:14:14 -0600
+ (Mon, 03 Dec 2012) | 12 lines Cleanup CLI resources on exit and
+ CLI command registration errors. (issue ASTERISK-20649) Reported
+ by: Corey Farrell Patches: cli-leaks-1_8-10.patch (license #5909)
+ patch uploaded by Corey Farrell cli-leaks-11-trunk.patch (license
+ #5909) patch uploaded by Corey Farrell Modified ........ Merged
+ revisions 377073 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, main/ccss.c: Merged revisions 377038 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r377038 | rmudgett | 2012-12-03 11:06:44 -0600
+ (Mon, 03 Dec 2012) | 10 lines Fix CCSS CLI commands and logger
+ level not unregistered. (issue ASTERISK-20649) Reported by: Corey
+ Farrell Patches: ccss-cleanup-all.patch (license #5909) patch
+ uploaded by Corey Farrell ........ Merged revisions 377037 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-11-30 22:25 +0000 [r376656-376982] Automerge script <automerge@asterisk.org>
+
+ * channels/misdn/isdn_lib.c, /: Merged revisions 376951 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376951 | rmudgett | 2012-11-30 15:33:38 -0600
+ (Fri, 30 Nov 2012) | 18 lines chan_misdn: Fix sending
+ RELEASE_COMPLETE in response to SETUP. Fix sending a
+ RELEASE_COMPLETE in response to a SETUP if chan_misdn does not
+ have a B channel available to assign to the call. (closes issue
+ ABE-2869) Reported by: Guenther Kelleter Patches:
+ setup-reject_2.diff (license #6372) patch uploaded by Guenther
+ Kelleter Modified ........ Merged revision 376949 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 376950 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c, funcs/func_volume.c: Merged revisions
+ 376916,376920 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376916 | mmichelson | 2012-11-30 10:23:46 -0600
+ (Fri, 30 Nov 2012) | 23 lines Fix potential crashes during SIP
+ attended transfers. The principal behind this patch is simple.
+ During a transfer, we manipulate channels that are owned by a
+ separate thread than the one we currently are running in, so it
+ makes sense that we need to grab a reference to the channels so
+ that they cannot disappear out from under us. In the wild,
+ crashes were sometimes seen when the transferring party would
+ hang up the call before the transfer target answered the call.
+ The most common place to see the crash occur was when attempting
+ to send a connected line update to the transferer channel.
+ (closes issue ASTERISK-20226) Reported by Jared Smith Patches:
+ ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
+ Tested by: Jared Smith ........ Merged revisions 376901 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r376920 | seanbright | 2012-11-30 11:06:21 -0600
+ (Fri, 30 Nov 2012) | 5 lines Minor spelling fix to the VOLUME
+ documentation. ........ Merged revisions 376919 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/chan_local.c, /, channels/chan_sip.c: Merged revisions
+ 376865,376869 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376865 | rmudgett | 2012-11-29 16:30:26 -0600
+ (Thu, 29 Nov 2012) | 7 lines Fix compile error. (issue
+ ASTERISK-20724) ........ Merged revisions 376864 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r376869 | rmudgett | 2012-11-29 16:58:28 -0600
+ (Thu, 29 Nov 2012) | 7 lines chan_local: Fix local_pvt ref leak
+ in local_devicestate(). Regression introduced by ASTERISK-20390
+ fix. ........ Merged revisions 376868 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 376835 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376835 | elguero | 2012-11-29 15:51:50 -0600
+ (Thu, 29 Nov 2012) | 19 lines Improve Code Readability And Fix
+ Setting natdetected Flag For 1.8, 10, 11 and trunk we are are
+ improving the code readability. For 11 and trunk, auto nat
+ detection was added. The natdetected flag was being set to 1 when
+ the host address in the VIA header did not specifiy a port. This
+ patch fixes this by setting the port on the temporary sock
+ address used to SIP_STANDARD_PORT in order for the sock address
+ comparison to work properly. (closes issue ASTERISK-20724)
+ Reported by: Michael L. Young Patches:
+ asterisk-20724-set-port-v2.diff uploaded by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2206/
+ ........ Merged revisions 376834 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/astmm.c, main/asterisk.c, /: Merged revisions 376789 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376789 | rmudgett | 2012-11-28 18:45:11 -0600
+ (Wed, 28 Nov 2012) | 26 lines Add MALLOC_DEBUG atexit unreleased
+ malloc memory summary. * Adds the following CLI commands to
+ control MALLOC_DEBUG reporting of unreleased malloc memory when
+ Asterisk is shut down. memory atexit list on memory atexit list
+ off memory atexit summary byline memory atexit summary byfunc
+ memory atexit summary byfile memory atexit summary off * Made
+ check all remaining allocated region blocks atexit for fence
+ violations. * Increased the allocated region hash table size by
+ about three times. It still isn't large enough considering the
+ number of malloced blocks Asterisk uses. * Made CLI "memory show
+ allocations anomalies" use regions_check_all_fences(). Review:
+ https://reviewboard.asterisk.org/r/2196/ ........ Merged
+ revisions 376788 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/astmm.c, /: Merged revisions 376759 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376759 | rmudgett | 2012-11-28 18:05:25 -0600
+ (Wed, 28 Nov 2012) | 19 lines Enhance MALLOC_DEBUG CLI commands.
+ * Fixed CLI "memory show allocations" misspelling of anomalies
+ option. The command will still accept the original misspelling. *
+ Miscellaneous tweaks to CLI "memory show allocations" command
+ output format. * Made CLI "memory show summary" summarize by line
+ number instead of by function if a filename is given. * Made CLI
+ "memory show summary" sort its output by filename or
+ function-name/line-number depending upon request. * Miscellaneous
+ tweaks to CLI "memory show summary" command output format.
+ ........ Merged revisions 376758 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/manager.c, /: Merged revisions 376726 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376726 | jrose | 2012-11-28 10:30:27 -0600
+ (Wed, 28 Nov 2012) | 16 lines manager: Make challenge work with
+ allowmultiplelogin=no Prior to this patch, challenge would yield
+ a multiple logins error if used without providing the username
+ (which isn't really supposed to be an argument to challenge) if
+ allowmultiplelogin was set to no because allowmultiplelogin finds
+ a user with a zero length login name. This check is simply
+ disabled for the challenge action when the username is empty by
+ this patch. (closes issue ASTERISK-20677) Reported by: Vladimir
+ Patches: challenge_action_nomultiplelogin.diff uploaded by
+ Jonathan Rose (license 6182) ........ Merged revisions 376725
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * UPGRADE.txt, main/pbx.c, /: Merged revisions 376689 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376689 | rmudgett | 2012-11-27 17:58:23 -0600
+ (Tue, 27 Nov 2012) | 33 lines Fix extension matching with the '-'
+ char. The '-' char is supposed to be ignored by the dialplan
+ extension matching. Unfortunately, it's treatment is not handled
+ consistently throughout the extension matching code. * Made the
+ old exten matching code consistently ignore '-' chars. * Made the
+ old exten matching code consistently handle case in the matching.
+ * Made ignore empty character sets. * Fixed ast_extension_cmp()
+ to return -1, 0, or 1 as documented. The only user of it in
+ pbx_lua.c was testing for -1. It was originally returning the
+ strcmp() value for less than which is not usually going to be -1.
+ * Fix character set sorting if the sets have the same number of
+ characters and start with the same character. Character set [0-9]
+ now sorts before [02-9a] as originally intended. * Updated some
+ extension label and priority already in use warnings to also
+ indicate if the extension is aliased. (closes issue
+ ASTERISK-19205) Reported by: Philippe Lindheimer, Birger "WIMPy"
+ Harzenetter Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2201/ ........ Merged
+ revisions 376688 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_celgenuserevent.c, pbx/pbx_dundi.c,
+ addons/res_config_mysql.c, /: Merged revisions 376658 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376658 | rmudgett | 2012-11-27 14:36:45 -0600
+ (Tue, 27 Nov 2012) | 21 lines Remove unnecessary channel module
+ references. * Removed call to ast_module_user_hangup_all() in
+ res_config_mysql.c since it is effectively a noop. No channels
+ can attach a reference to that module. * Removed call to
+ ast_module_user_hangup_all() in app_celgenuserevent.c. The caller
+ of unload_module() has already called it. * Removed redundant
+ channel module references in pbx_dundi.c. The registered dialplan
+ function callback dispatchers for the read/read2/write callbacks
+ already reference the module before calling. * pbx_dundi: Moved
+ unregistering CLI commands, DUNDi switch, and dialplan functions
+ to the first thing the unload_module() does. This will reduce the
+ chance of new channels using DUNDi services while the module is
+ being torn down. ........ Merged revisions 376657 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * include/asterisk/linkedlists.h, /: Merged revisions 376628 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376628 | rmudgett | 2012-11-27 11:35:54 -0600
+ (Tue, 27 Nov 2012) | 7 lines Made AST_LIST_REMOVE() simpler and
+ use better names. * Update doxygen of AST_LIST_REMOVE(). ........
+ Merged revisions 376627 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-11-23 00:27 +0000 [r376614] Automerge script <automerge@asterisk.org>
+
+ * main/logger.c, include/asterisk/lock.h, main/lock.c, /: Merged
+ revisions 376587 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376587 | mjordan | 2012-11-22 17:56:00 -0600
+ (Thu, 22 Nov 2012) | 23 lines Re-initialize logmsgs mutex upon
+ logger initialization to prevent lock errors Similar to the patch
+ that moved the fork earlier in the startup sequence to prevent
+ mutex errors in the recursive mutex surrounding the read/write
+ thread registration lock, this patch re-initializes the logmsgs
+ mutex. Part of the start up sequence before forking the process
+ into the background includes reading asterisk.conf; this has to
+ occur prior to the call to daemon in order to read startup
+ parameters. When reading in a conf file, log statements can be
+ generated. Since this can't be avoided, the mutex instead is
+ re-initialized to ensure a reset of any thread tracking
+ information. This patch also includes some additional debugging
+ to catch errors when locking or unlocking the recursive mutex
+ that surrounds locks when the DEBUG_THREADS build option is
+ enabled. DO_CRASH or THREAD_CRASH will cause an abort() if a
+ mutex error is detected. (issue ASTERISK-19463) Reported by:
+ mjordan Tesetd by: mjordan ........ Merged revisions 376586 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-11-20 17:24 +0000 [r376497-376539] Automerge script <automerge@asterisk.org>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+ revisions 376522 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376522 | mmichelson | 2012-11-20 11:01:04 -0600
+ (Tue, 20 Nov 2012) | 14 lines Add "Require: timer" to 200 OK
+ responses when appropriate. The method by which the Require
+ header is added to 200 responses is inspired by the method that
+ Olle Johansson uses in his darjeeling-prack branch. (closes issue
+ ASTERISK-20570) Reported by Matt Jordan, at the behest of Olle
+ Johansson Review: https://reviewboard.asterisk.org/r/2172
+ ........ Merged revisions 376521 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/indications.c, /, channels/chan_sip.c,
+ main/security_events.c: Merged revisions 376470 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376470 | wdoekes | 2012-11-19 13:44:58 -0600
+ (Mon, 19 Nov 2012) | 11 lines Fix most leftover non-opaque
+ ast_str uses. Instead of calling str->str, one should use
+ ast_str_buffer(str). Same goes for str->used as
+ ast_str_strlen(str) and str->len as ast_str_size(str). Review:
+ https://reviewboard.asterisk.org/r/2198 ........ Merged revisions
+ 376469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-11-18 20:23 +0000 [r376427-376446] Automerge script <automerge@asterisk.org>
+
+ * main/utils.c, main/stdtime/localtime.c, main/asterisk.c, /:
+ Merged revisions 376431 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376431 | mjordan | 2012-11-18 14:18:24 -0600
+ (Sun, 18 Nov 2012) | 49 lines Reorder startup sequence to prevent
+ lockups when process is sent to background Although it is very
+ rare and timing dependent, the potential exists for the call to
+ 'daemon' to cause what appears to be a deadlock in Asterisk
+ during startup. This can occur when a recursive mutex is obtained
+ prior to the daemon call executing. Since daemon uses fork to
+ send the process into the background, any threading primitives
+ are unsafe to re-use after the call. Implementations of pthread
+ recursive mutexes are highly likely to store the thread
+ identifier of the thread that previously obtained the mutex. If
+ the mutex was locked prior to the fork, a subsequent unlock
+ operation will potentially fail as the thread identifier is no
+ longer valid. Since the mutex is still locked, all subsequent
+ attempts to grab the mutex by other threads will block. This
+ behavior exhibited itself most often when DEBUG_THREADS was
+ enabled, as this compile time option surrounds the mutexes in
+ Asterisk with another recursive mutex that protects the storage
+ of thread related information. This made it much more likely that
+ a recursive mutex would be obtained prior to daemon and unlocked
+ after the call. This patch does the following: a) It backports a
+ patch from Asterisk 11 that prevents the spawning of the
+ localtime monitoring thread. This thread is now spawned after
+ Asterisk has fully booted. b) It re-orders the startup sequence
+ to call daemon earlier during Asterisk startup. This limits the
+ potential of threading primitives being accessed by
+ initialization calls before daemon is called. c) It removes calls
+ to ast_verbose/ast_log/etc. prior to daemon being called.
+ Developers should send error messages directly to stderr prior to
+ daemon, as calls to ast_log may access recursive mutexes that
+ store thread related information. d) It reorganizes when thread
+ local storage is created for storing lock information during the
+ creation of threads. Prior to this patch, the read/write lock
+ protecting the list of threads in ast_register_thread would
+ utilize the lock in the thread local storage prior to it being
+ initialized; this patch prevents that. On a very related note,
+ this patch will *greatly* improve the stability of the Asterisk
+ Test Suite. Review: https://reviewboard.asterisk.org/r/2197
+ (closes issue ASTERISK-19463) Reported by: mjordan Tested by:
+ mjordan ........ Merged revisions 376428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/confbridge/conf_state.c, /: Merged revisions 376414 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ........ r376414 | mjordan | 2012-11-18 08:22:39 -0600 (Sun, 18
+ Nov 2012) | 8 lines Add a test event that reports changes in
+ ConfBridge state This patch adds a test event to ConfBridge that
+ reports transitions between states in ConfBridge. This is used by
+ tests in the Asterisk Test Suite that verify state changes based
+ on the entering/leaving of conference participants. ........
+
+2012-11-16 20:23 +0000 [r376338-376408] Automerge script <automerge@asterisk.org>
+
+ * res/res_monitor.c, /: Merged revisions 376390 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376390 | jrose | 2012-11-16 13:41:55 -0600
+ (Fri, 16 Nov 2012) | 17 lines monitor: prevent attempts to
+ move/remove recordings skipped with 'i' and 'o'. The i and o
+ options for monitor skip the input and output sides of a
+ recording respectively. This patch addresses a problem in those
+ options when monitor is called without specifying a specific
+ filename where monitor will try to move the recording that was
+ skipped. Since this usually doesn't exist when these options are
+ used, it would produce a warning when it does this in most cases,
+ but it is conceivable that there are use cases where this could
+ result in moving/removing a file unintentionally. (closes issue
+ ASTERISK-20641) Reported by: Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/2190/ ........ Merged
+ revisions 376389 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * utils/extconf.c, /: Merged revisions 376342 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376342 | dlee | 2012-11-15 18:08:56 -0600 (Thu,
+ 15 Nov 2012) | 9 lines Fixed extconf.c breakage introduced in
+ r376306. To quote wdoekes: > Note that I'm not confirming
+ legitimacy of having that file in tree at > all. Is anyone using
+ aelparse/conf2ael? ........ Merged revisions 376340 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * utils/hashtest.c (removed), tests/test_hashtab_thrash.c (added),
+ utils/hashtest2.c (removed), include/asterisk/hashtab.h,
+ utils/Makefile, tests/test_astobj2_thrash.c (added),
+ utils/utils.xml, /, apps/app_meetme.c: Merged revisions
+ 376308,376315 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376308 | jrose | 2012-11-15 16:55:04 -0600
+ (Thu, 15 Nov 2012) | 17 lines app_meetme: Fix channels lingering
+ when hung up under certain conditions Channels would get stuck
+ and MeetMe would repeatedly display an Unable to write frame to
+ channel error in the conf_run function if hung up during certain
+ sound prompts such as during user count announcements. This patch
+ fixes that by reintroducing a hangup check in the meetme's main
+ loop (also in conf_run). (closes issue ASTERISK-20486) Reported
+ by: Michael Cargile Review:
+ https://reviewboard.asterisk.org/r/2187/ Patches:
+ meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan
+ Rose (license 6182) ........ Merged revisions 376307 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r376315 | dlee | 2012-11-15 17:17:54 -0600 (Thu,
+ 15 Nov 2012) | 28 lines Migrate hashtest/hashtest2 to be unit
+ tests. Both hashtest and hashtest2 are manual testing apps that
+ thrash hash tables (hashtab and ao2 containers, respectively), by
+ spinning up several threads that randomly insert, delete, lookup
+ and iterate over the hash table. If the app doesn't crash, the
+ hash table probably passes the test. Those utils are not a part
+ of the typical Asterisk build, so they do not usually get
+ compiled. This all makes them less that useful. This patch
+ removes those manual test programs and replaces them with
+ Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It
+ also attempts to make the tests more deterministic. * Rather than
+ spinning up some number of threads that operate on the hash table
+ randomly, spin up four threads that concurrenly add, remove,
+ lookup and iterate over the hash table. * Each thread checks the
+ state of the hash table both during and after execution, and
+ indicates a test failure if things are not as expected. * Each
+ thread times out after 60 seconds to prevent deadlocking the unit
+ test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2189/ ........ Merged
+ revisions 376306 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-11-15 23:06 +0000 [r376309-376311] David M. Lee <dlee@digium.com>
+
+ * utils/hashtest.c (added), tests/test_hashtab_thrash.c (removed),
+ utils/hashtest2.c (added), include/asterisk/hashtab.h,
+ utils/Makefile, tests/test_astobj2_thrash.c (removed),
+ utils/utils.xml, /: Reverted r376309; merged to wrong branch
+
+ * utils/hashtest.c (removed), tests/test_hashtab_thrash.c (added),
+ utils/hashtest2.c (removed), include/asterisk/hashtab.h,
+ utils/Makefile, tests/test_astobj2_thrash.c (added),
+ utils/utils.xml, /: Migrate hashtest/hashtest2 to be unit tests.
+ Both hashtest and hashtest2 are manual testing apps that thrash
+ hash tables (hashtab and ao2 containers, respectively), by
+ spinning up several threads that randomly insert, delete, lookup
+ and iterate over the hash table. If the app doesn't crash, the
+ hash table probably passes the test. Those utils are not a part
+ of the typical Asterisk build, so they do not usually get
+ compiled. This all makes them less that useful. This patch
+ removes those manual test programs and replaces them with
+ Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It
+ also attempts to make the tests more deterministic. * Rather than
+ spinning up some number of threads that operate on the hash table
+ randomly, spin up four threads that concurrenly add, remove,
+ lookup and iterate over the hash table. * Each thread checks the
+ state of the hash table both during and after execution, and
+ indicates a test failure if things are not as expected. * Each
+ thread times out after 60 seconds to prevent deadlocking the unit
+ test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2189/ ........ Merged
+ revisions 376306 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-15 02:22 +0000 [r376260-376281] Automerge script <automerge@asterisk.org>
+
+ * apps/app_voicemail.c, /: Merged revisions 376263 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376263 | newtonr | 2012-11-14 19:50:54 -0600
+ (Wed, 14 Nov 2012) | 10 lines (issue ASTERISK-20280) (closes
+ issue ASTERISK-20280) Reported by: Tomo Takebe Tested by: Rusty
+ Newton Patches: asterisk20280.patch uploaded by Rusty Newton
+ (license 5829) ........ Merged revisions 376262 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * pbx/pbx_spool.c, /: Merged revisions 376233 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376233 | rmudgett | 2012-11-14 13:50:52 -0600
+ (Wed, 14 Nov 2012) | 19 lines Fix call files when astspooldir is
+ relative. Future dated call files are ignored when astspooldir is
+ relative to the current directory. The queue_file() assumed that
+ the qdir needed to be prepended if the given filename did not
+ start with a '/'. If astspooldir is relative it is not going to
+ start from the root directory obviously so it will not start with
+ a '/'. The filename used in queue_file() ultimately results in
+ qdir prepended multiple times. * Made queue_file() not prepend
+ qdir if the filename contains a '/'. (closes issue
+ ASTERISK-20593) Reported by: James Le Cuirot Patches:
+ 0004-Fix-future-call-files-from-relative-directories.patch
+ (license #6439) patch uploaded by James Le Cuirot ........ Merged
+ revisions 376232 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-11-13 18:22 +0000 [r376165-376216] Automerge script <automerge@asterisk.org>
+
+ * main/channel.c, /: Merged revisions 376208 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376208 | beagles | 2012-11-13 12:20:13 -0600
+ (Tue, 13 Nov 2012) | 14 lines Patch to prevent stopping the
+ active generator when it is not the silence generator. This patch
+ introduces an internal helper function to safely check whether
+ the current generator is the one that is expected before
+ deactivating it. The current externally accessible
+ ast_channel_stop_generator() function has been modified to be
+ implemented in terms of the new function. (closes issue
+ ASTERISK-19918) Reported by: Eduardo Abad ........ Merged
+ revisions 376199 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/pbx.c, /: Merged revisions 376167 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376167 | file | 2012-11-12 14:44:56 -0600 (Mon,
+ 12 Nov 2012) | 14 lines Properly check if the "Context" and
+ "Extension" headers are empty in a ShowDialPlan action. The code
+ which handles the ShowDialPlan action wrongly assumed that a
+ non-NULL return value from the function which retrieves headers
+ from an action indicates that the header has a value. This is
+ incorrect and the contents must be checked to see if they are
+ blank. (closes issue ASTERISK-20628) Reported by: jkroon Patches:
+ asterisk-showdialplan-incorrect-error.patch uploaded by jkroon
+ ........ Merged revisions 376166 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/pbx.c, /: Merged revisions 376143 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376143 | elguero | 2012-11-12 14:15:27 -0600
+ (Mon, 12 Nov 2012) | 20 lines Fix Dynamic Hints Variable
+ Substition - Underscore Problem When adding a dynamic hint, if an
+ extension contains an underscore no variable subsitution is being
+ performed. This patch changes from checking if the extension
+ contains an underscore to checking if the extension begins with
+ an underscore. (closes issue ASTERISK-20639) Reported by: Steven
+ T. Wheeler Tested by: Steven T. Wheeler, Michael L. Young
+ Patches: asterisk-20639-dynamic-hint-underscore.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2188/ ........ Merged
+ revisions 376142 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-11-08 22:25 +0000 [r375992-376117] Automerge script <automerge@asterisk.org>
+
+ * /, res/res_fax.c: Merged revisions 376088 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376088 | mmichelson | 2012-11-08 15:59:13 -0600
+ (Thu, 08 Nov 2012) | 12 lines Fix a "set but not used" warning on
+ newer gccs. Turns out the "helpful" setting of ms and res in this
+ macro is completely useless after the timeout antipattern fix. If
+ you're a new guy looking to write code, don't write a macro like
+ this one. ........ Merged revisions 376087 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/sig_ss7.c, /: Merged revisions 376059 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376059 | rmudgett | 2012-11-08 15:07:09 -0600
+ (Thu, 08 Nov 2012) | 16 lines chan_dahdi/SS7: Made reject
+ incoming call for an in-alarm or blocked channel. If a SS7 call
+ comes in requesting a CIC that is in-alarm, the call is accepted
+ and connects if the extension exists in the dialplan. The call
+ does not have any audio. * Made release the call immediately with
+ circuit congestion cause. (closes issue ASTERISK-20204) Reported
+ by: Tuan Le Patches: jira_asterisk_20204_v1.8.patch (license
+ #5621) patch uploaded by rmudgett ........ Merged revisions
+ 376058 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/utils.c, main/astmm.c, main/asterisk.c,
+ include/asterisk/utils.h, include/asterisk/astmm.h, /: Merged
+ revisions 376030 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r376030 | rmudgett | 2012-11-08 11:08:39 -0600
+ (Thu, 08 Nov 2012) | 35 lines Add MALLOC_DEBUG enhancements. *
+ Makes malloc() behave like calloc(). It will return a memory
+ block filled with 0x55. A nonzero value. * Makes free() fill the
+ released memory block and boundary fence's with 0xdeaddead. Any
+ pointer use after free is going to have a pointer pointing to
+ 0xdeaddead. The 0xdeaddead pointer is usually an invalid memory
+ address so a crash is expected. * Puts the freed memory block
+ into a circular array so it is not reused immediately. * When the
+ circular array rotates out a memory block to the heap it checks
+ that the memory has not been altered from 0xdeaddead. * Made the
+ astmm_log message wording better. * Made crash if the DO_CRASH
+ menuselect option is enabled and something is found. * Fixed a
+ potential alignment issue on 64 bit systems. struct
+ ast_region.data[] should now be aligned correctly for all
+ platforms. * Extracted region_check_fences() from
+ __ast_free_region() and handle_memory_show(). * Updated
+ handle_memory_show() CLI usage help. Review:
+ https://reviewboard.asterisk.org/r/2182/ ........ Merged
+ revisions 376029 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_record.c, channels/chan_agent.c, main/utils.c,
+ include/asterisk/channel.h, apps/app_queue.c, channels/sig_pri.c,
+ channels/chan_iax2.c, main/channel.c, channels/chan_dahdi.c,
+ channels/sig_analog.c, apps/app_waitforring.c,
+ include/asterisk/time.h, apps/app_jack.c, apps/app_dial.c,
+ main/pbx.c, main/rtp_engine.c, apps/app_meetme.c, /,
+ res/res_fax.c: Merged revisions 375995 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375995 | mmichelson | 2012-11-07 11:16:24 -0600
+ (Wed, 07 Nov 2012) | 41 lines Multiple revisions 375993-375994
+ ........ r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed,
+ 07 Nov 2012) | 30 lines Fix misuses of timeouts throughout the
+ code. Prior to this change, a common method for determining if a
+ timeout was reached was to call a function such as
+ ast_waitfor_n() and inspect the out parameter that told how many
+ milliseconds were left, then use that as the input to
+ ast_waitfor_n() on the next go-around. The problem with this is
+ that in some cases, submillisecond timeouts can occur, resulting
+ in the out parameter not decreasing any. When this happens
+ thousands of times, the result is that the timeout takes much
+ longer than intended to be reached. As an example, I had a
+ situation where a 3 second timeout took multiple days to finally
+ end since most wakeups from ast_waitfor_n() were under a
+ millisecond. This patch seeks to fix this pattern throughout the
+ code. Now we log the time when an operation began and find the
+ difference in wall clock time between now and when the event
+ started. This means that sub-millisecond timeouts now cannot play
+ havoc when trying to determine if something has timed out. Part
+ of this fix also includes changing the function ast_waitfor() so
+ that it is possible for it to return less than zero when a
+ negative timeout is given to it. This makes it actually possible
+ to detect errors in ast_waitfor() when there is no timeout.
+ (closes issue ASTERISK-20414) reported by David M. Lee Review:
+ https://reviewboard.asterisk.org/r/2135/ ........ r375994 |
+ mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3
+ lines Remove some debugging that accidentally made it in the last
+ commit. ........ Merged revisions 375993-375994 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/features.c, include/asterisk/channel.h, .cleancount,
+ include/asterisk/features.h, main/channel.c, /: Merged revisions
+ 375965 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375965 | rmudgett | 2012-11-06 12:27:19 -0600
+ (Tue, 06 Nov 2012) | 21 lines Fix stuck DTMF when bridge is
+ broken. When a bridge is broken by an AMI Redirect action or the
+ ChannelRedirect application, an in progress DTMF digit could be
+ stuck sending forever. * Made simulate a DTMF end event when a
+ bridge is broken and a DTMF digit was in progress. (closes issue
+ ASTERISK-20492) Reported by: Jeremiah Gowdy Patches:
+ bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by
+ Jeremiah Gowdy Modified to jira_asterisk_20492_v1.8.patch
+ jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2169/ ........ Merged
+ revisions 375964 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-12-10 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.11.0-digiumphones Released.
+
+2012-12-06 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.11.0-digiumphones-rc3 Released.
+
+ * chan_local: Fix local_pvt ref leak in local_devicestate().
+
+ Regression introduced by ASTERISK-20390 fix.
+
+ (closes issue ASTERISK-20769)
+ Reported by: rmudgett
+
+2012-12-05 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.11.0-digiumphones-rc2 Released.
+
+ * Fix a SIP request memory leak with TLS connections.
+
+ During the TLS re-work in chan_sip some TLS specific code was moved
+ into a separate function. This function operates on a copy of the
+ incoming SIP request. This copy was never deinitialized causing a
+ memory leak for each request processed.
+
+ This function is now given a SIP request structure which it can use
+ to copy the incoming request into. This reduces the amount of memory
+ allocations done since the internal allocated components are reused
+ between packets and also ensures the SIP request structure is
+ deinitialized when the TLS connection is torn down.
+
+ (closes issue ASTERISK-20763)
+ Reported by: deti
+
+2012-11-06 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.11.0-digiumphones-rc1 Released.
+
+2012-11-05 23:26 +0000 [r375858-375921] Automerge script <automerge@asterisk.org>
+
+ * res/res_timing_dahdi.c, res/res_timing_timerfd.c,
+ bridges/bridge_softmix.c, funcs/func_jitterbuffer.c,
+ include/asterisk/timing.h, res/res_musiconhold.c,
+ channels/chan_iax2.c, res/res_fax_spandsp.c,
+ res/res_timing_kqueue.c, main/timing.c, main/channel.c, /,
+ res/res_timing_pthread.c: Merged revisions 375894 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375894 | mjordan | 2012-11-05 17:00:32 -0600
+ (Mon, 05 Nov 2012) | 28 lines Refactor ast_timer_ack to return an
+ error and handle the error in timer users Currently, if an
+ acknowledgement of a timer fails Asterisk will not realize that a
+ serious error occurred and will continue attempting to use the
+ timer's file descriptor. This can lead to situations where errors
+ stream to the CLI/log file. This consumes significant resources,
+ masks the actual problem that occurred (whatever caused the timer
+ to fail in the first place), and can leave channels in odd
+ states. This patch propagates the errors in the timing resource
+ modules up through the timer core, and makes users of these
+ timers handle acknowledgement failures. It also adds some
+ defensive coding around the use of timers to prevent using bad
+ file descriptors in off nominal code paths. Note that the patch
+ created by the issue reporter was modified slightly for this
+ commit and backported to 1.8, as it was originally written for
+ Asterisk 10. Review: https://reviewboard.asterisk.org/r/2178/
+ (issue ASTERISK-20032) Reported by: Jeremiah Gowdy patches:
+ jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license
+ 6358) ........ Merged revisions 375893 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/loader.c, /: Merged revisions 375863 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375863 | rmudgett | 2012-11-05 15:39:00 -0600
+ (Mon, 05 Nov 2012) | 10 lines Add safety NULL pointer check in
+ module user references. Made __ast_module_user_remove() check for
+ NULL pointers. ........ Merged revision 375860 from C.3 ........
+ Merged revisions 375862 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * UPGRADE.txt, /: Merged revisions 375846 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r375846 | jrose | 2012-11-05 11:55:13 -0600 (Mon, 05 Nov 2012) |
+ 9 lines chan_sip: Document a change to user-field encoding
+ introduced with r303509 The change in question was added to
+ improve compliance with RFC3261, but at the time of commit, it
+ wasn't adequately documented in the UPGRADE notes. (closes issue
+ ASTERISK-20561) Reported by: Deniz Review:
+ https://reviewboard.asterisk.org/r/2177/ ........
+
+2012-11-04 03:25 +0000 [r375612-375828] Automerge script <automerge@asterisk.org>
+
+ * main/manager.c, /, res/res_fax.c: Merged revisions
+ 375794,375797,375801 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375794 | mjordan | 2012-11-03 21:30:30 -0500
+ (Sat, 03 Nov 2012) | 15 lines Properly clean up manager resources
+ on exit This patch does two things: 1) It properly unregisters
+ the manager CLI commands 2) It cleans up AMI users on exit. Prior
+ to this patch, the AMI users were not being disposed of properly,
+ resulting in a memory leak. (closes issue ASTERISK-20646)
+ Reported by: Corey Farrell patches: manager_shutdown.patch
+ uploaded by Corey Farrell (license 5909) ........ Merged
+ revisions 375793 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r375797 | mjordan | 2012-11-03 21:42:43 -0500
+ (Sat, 03 Nov 2012) | 9 lines Only deref a reserved gateway
+ session if we actually reserved one Its perfectly acceptable to
+ have a gateway session unreserved when we go to first allocate
+ one. Unreffing the reserved gateway session - when its NULL -
+ will result in an assertion error. This problem was caught by the
+ Asterisk Test Suite (once we had enough of the debugging flags
+ enabled) ................ r375801 | mjordan | 2012-11-03 22:08:12
+ -0500 (Sat, 03 Nov 2012) | 17 lines Don't attempt to purge
+ sessions when no sessions exist Manager's tcp/tls objects have a
+ periodic function that purge old manager sessions periodically.
+ During shutdown, the underlying container holding those sessions
+ can be disposed of and set to NULL before the tcp/tls periodic
+ function is stopped. If the periodic function fires, it will
+ attempt to iterate over a NULL container. This patch checks for
+ whether or not the sessions container exists before attempting to
+ purge sessions out of it. If the sessions container is NULL, we
+ simply return. Note that this error was also caught by the
+ Asterisk Test Suite. ........ Merged revisions 375800 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/db.c, main/xmldoc.c, /: Merged revisions 375759,375761 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375759 | mjordan | 2012-11-03 19:55:19 -0500
+ (Sat, 03 Nov 2012) | 18 lines Fix memory leak when unloading XML
+ documentation This patch is a modified version of a patch
+ originally committed for the Asterisk 11 branch in r375756. A
+ portion of that patch, that fixed the memory leak during
+ unloading XML documentation, applies to branches 1.8 and 10 as
+ well. The patch for this issue was modified for these two
+ branches. (issue ASTERISK-20648) Reported by: Corey Farrell
+ Tested by: mjordan patches: xmldoc-memory_leak.patch uploaded by
+ Corey Farrell (license 5909) ........ Merged revisions 375758
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r375761 | mjordan | 2012-11-03 20:13:37 -0500
+ (Sat, 03 Nov 2012) | 15 lines Properly finalize prepared SQLite3
+ statements to prevent memory leak The AstDB uses prepared SQLite3
+ statements to retrieve data from the SQLite3 database. These
+ statements should be finalized during Asterisk shutdown so that
+ the SQLite3 database can be properly closed. Failure to finalize
+ the statements results in a memory leak and a failure when
+ closing the database. This patch fixes those issues by ensuring
+ that all prepared statements are properly finalized at shutdown.
+ (closes issue ASTERISK-20647) Reported by: Corey Farrell patches:
+ astdb-sqlite3_close.patch uploaded by Corey Farrell (license
+ 5909) ................
+
+ * main/cdr.c, /: Merged revisions 375728 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375728 | mjordan | 2012-11-03 18:51:43 -0500
+ (Sat, 03 Nov 2012) | 16 lines Prevent multiple CDR batches from
+ conflicting when scheduling the CDR write The Asterisk Test Suite
+ caught an error condition where a scheduled CDR batch write can
+ be deleted twice if two channels attempt to post their CDRs at
+ the same time. The batch CDR mutex is locked while the CDRs are
+ appended to the current batch list; however, it is unlocked prior
+ to actually scheduling the CDR write. As such, two threads can
+ attempt to remove the currently scheduled batch write at the same
+ time, resulting in an assertion error. This patch extends the
+ time that the mutex is locked to encompass actually scheduling
+ the write. This prevents two threads from unscheduling the
+ currently scheduled write at the same time. ........ Merged
+ revisions 375727 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * README, include/asterisk/doxyref.h, /: Merged revisions 375699
+ via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375699 | lathama | 2012-11-02 22:15:30 -0500
+ (Fri, 02 Nov 2012) | 9 lines Doxygen Updates Replace links to
+ missing text files removed in the 1.6.x series with links to the
+ wiki. Doxygen can handle URLs fine, don't atempt to quote them.
+ Also update the wiki link in the Readme to get everyone on the
+ same page. (issue ASTERISK-20259) ........ Merged revisions
+ 375698 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/format_pref.c, main/channel.c, channels/chan_misdn.c, /,
+ main/ccss.c: Merged revisions 375659 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375659 | rmudgett | 2012-11-02 15:53:53 -0500
+ (Fri, 02 Nov 2012) | 5 lines Things don't need to be that const.
+ ........ Merged revisions 375658 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged
+ revisions 375626 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375626 | rmudgett | 2012-11-02 13:42:23 -0500
+ (Fri, 02 Nov 2012) | 127 lines Multiple revisions 375519-375524
+ ........ r375519 | rmudgett | 2012-10-30 16:06:15 -0500 (Tue, 30
+ Oct 2012) | 11 lines chan_misdn: Timer primitives must be handled
+ first. The frm->addr is a different "address space" than the
+ stack/instance address of other Lx primitives. The test for B
+ channel instance address could fail. Patches: patch01_timers.diff
+ (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2888
+ ........ r375520 | rmudgett | 2012-10-30 16:14:58 -0500 (Tue, 30
+ Oct 2012) | 10 lines chan_misdn: Free memory in error paths and
+ other memory leaks. The one line commented with BUG is not easily
+ fixable because there is no de-init function one can call.
+ Patches: patch02_memory.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2888 ........ r375521 | rmudgett |
+ 2012-10-30 16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines
+ chan_misdn: ISDN NT L2 de-establish/establish * An NT-PTMP cannot
+ de/establish L2 since it doesn't know the TEIs. * On NT-PTP L2 is
+ started when L1 is finally active in handle_l1. * L2 deactivation
+ logging cleanup. * L2 aggregate link status is unknown for
+ NT-PTMP, show as "UNKN". * Removed unused functions and code for
+ L2 handling. Patches: patch03_L2estab.diff (license #6372) patch
+ uploaded by Guenther Kelleter Modified JIRA ABE-2888 ........
+ r375522 | rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012)
+ | 22 lines chan_misdn: Fix broken upper_id/lower_id usage.
+ Sending PH prim via lower_id layer (3 or 1) simply does not work.
+ For TE (3) it returns an error (len=-6) which is not evaluated by
+ handle_l1(), so the L1 layer status ends up wrong. Instead PH
+ must be sent via L4, only then does it reach L1 without an error
+ message. And NT PH prims only reach L1 when they are sent to
+ layer 2 id. --> use upper_id to send PH primitives. * Check for
+ errors in PH_(DE)ACTIVATE | CONFIRM. * Debug messages are
+ improved. * The lower_id is now not used for anything, except:
+ Why is lower_id layer deleted when it wasn't created? I removed
+ this code since it looks very wrong. Patches:
+ patch04_l1activation.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2888 ........ r375523 | rmudgett |
+ 2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines
+ chan_misdn: Fix loss of B channels if L1 is down. If you make 2
+ calls out an NT PTMP port which is not connected to any phone,
+ the B channel associated with that call becomes unusable until
+ Asterisk is restarted. The problem is the EVENT_SETUP is queued
+ when L1 is not up in misdn_lib_send_event(). If L1 cannot be
+ activated the event won't be dequeued. It gets even worse when
+ the call is hung up. The queued EVENT_SETUP will be overwritten
+ by an EVENT_DISCONNECT. The reserved B channel then will never be
+ freed. If later someone connects a phone to the port, L1 will
+ eventually activate and the queued EVENT_DISCONNECT is sent down
+ the stack. However, it is ignored because it is the wrong call
+ state. The real fix would be that activation and queueing for a
+ new SETUP is done by the NT stack. But since it doesn't, the
+ workaround must be removed because it doesn't always work. Fix:
+ The event is no longer queued but immediately sent to the stack.
+ If L1 cannot be activated, the L3 state machine that was started
+ by the EVENT_SETUP will do its work, i.e. a timeout will release
+ the B channel properly. The SETUP possibly cannot be sent the
+ first time but is resent by T303 in case L1 could be activated.
+ Patches: patch05_bchan-loss.diff (license #6372) patch uploaded
+ by Guenther Kelleter Modified JIRA ABE-2888 ........ r375524 |
+ rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13
+ lines chan_misdn: Remove some calls to exit(). Try proper cleanup
+ when something goes wrong in misdn_lib_init(). Especially do not
+ call exit()! * Fix memory leak because stack_destroy() does not
+ free the stack struct. Patches: patch06_cleanup-init.diff
+ (license #6372) patch uploaded by Guenther Kelleter Modified JIRA
+ ABE-2888 ........ Merged revisions 375519-375524 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 375625 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 375601 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375601 | elguero | 2012-11-02 12:19:33 -0500
+ (Fri, 02 Nov 2012) | 14 lines Fix Wrong Result In Debug Message
+ For SDP Origin Processing While looking at some debug logs, I
+ noticed that it was being reported that the SDP origin line was
+ unsupported or failed. Upon looking into this on my local
+ machine, I found that I too was getting this debug message yet
+ everything seemed to be getting processed properly. What was
+ discovered is, that, the variable to determine what is displayed
+ in the debug message for the SDP line that was processed, was not
+ being set for the origin line when the result was successful.
+ This patch fixes this and was tested on local machine. ........
+ Merged revisions 375594 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-10-31 15:26 +0000 [r375387-375558] Automerge script <automerge@asterisk.org>
+
+ * res/res_calendar_ews.c, /: Merged revisions 375531 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375531 | mjordan | 2012-10-31 09:34:42 -0500
+ (Wed, 31 Oct 2012) | 24 lines Properly extract the Body
+ information of an EWS calendar item Unlike all other calendar
+ modules, res_calendar_ews fails to extract the Body information
+ for a calendar item. This is due, in part, to a quirk in the
+ schema in the XML - not only does a CalendarItem contain a Body
+ element, but the CalendarItem exists as a descendant of a
+ different Body element. The neon parser was erroneously skipping
+ all Body elements. This patch fixes that by bypassing Body
+ elements that are not a child of CalendarItem, and parsing the
+ Body element out if it is a child. Note that the original patch
+ by Terry Wilson only needed slight modifications to make it
+ properly pull the Body information out; as such, while I've
+ linked to the patch that I uploaded for Dmitry, I've attributed
+ the patch to Terry. (closes issue ASTERISK-19738) Reported by:
+ Dmitry Burilov Tested by: Dmitry Burilov patches:
+ calendar_ews_body_2012_10_29.diff uploaded by Terry Wilson
+ (license 6283) ........ Merged revisions 375528 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * bridges/bridge_softmix.c, apps/app_mixmonitor.c, /: Merged
+ revisions 375485,375496 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375485 | jrose | 2012-10-30 13:55:58 -0500
+ (Tue, 30 Oct 2012) | 8 lines mixmonitor: Add a test event This
+ test event is being used to fix the mixmonitor_audiohook_inherit
+ test. ........ Merged revisions 375484 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r375496 | rmudgett | 2012-10-30 14:20:28 -0500
+ (Tue, 30 Oct 2012) | 8 lines Fix ConfBridge crash if no timing
+ module loaded. (closes issue ASTERISK-19448) Reported by: feyfre
+ Patches: smfix.patch (license #6099) patch uploaded by feyfre
+ Modified for coding guidelines. ................
+
+ * apps/app_confbridge.c, /: Merged revisions 375470 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10 ........
+ r375470 | jrose | 2012-10-30 09:42:29 -0500 (Tue, 30 Oct 2012) |
+ 7 lines confbridge: Fix a bug which made conferences not record
+ with AMI/CLI commands (closes issue ASTERISK-20601) Reported by:
+ Vilius Patches: confbridge_mixmonitor.diff uploaded by Jonathan
+ Rose (license 6182) ........
+
+ * apps/app_queue.c, /: Merged revisions 375451 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375451 | mjordan | 2012-10-29 21:14:01 -0500
+ (Mon, 29 Oct 2012) | 14 lines Ensure that the Queue application
+ tracks busy members in off nominal situations There are a few
+ code paths where the Queue application fails to count a paused or
+ in use queue member as being 'busy'. This can cause callers to
+ get stuck in the Queue until a paused agent unpauses themselves.
+ (closes issue ASTERISK-20623) Reported by: Bryan Walters patches:
+ app_queue.patch uploaded by Bryan Walters (license 5851) ........
+ Merged revisions 375450 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 375417 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375417 | mmichelson | 2012-10-29 16:09:18 -0500
+ (Mon, 29 Oct 2012) | 23 lines Prevent resetting of NATted
+ realtime peer address on reload. If a "sip reload" is issued for
+ a SIP peer, then his IP address will be cleared, thus resulting
+ in forgetting the public IP address. Asterisk will then attempt
+ to route SIP traffic to the private IP address. The fix here is
+ to make "sip reload" ignore realtime peers when "host = dynamic"
+ is spotted. Realtime peers can now only have their IP address
+ reset if they have gone from being not dynamic to being dynamic.
+ (closes issue ASTERISK-18203) reported by daren ferreira (closes
+ issue ASTERISK-20572) reported by JoshE Patches:
+ fix_nat_realtime.diff uploaded by JoshE (license #6075) ........
+ Merged revisions 375415 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/features.c, /: Merged revisions 375389 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375389 | rmudgett | 2012-10-29 14:28:38 -0500
+ (Mon, 29 Oct 2012) | 16 lines Fix the Park 'r' option when a
+ channel parks itself. When a channel uses the Park appliation to
+ park itself with the 'r' option, the channel hears music-on-hold
+ instead of the requested ringing. * Added a missing check for the
+ 'r' option when a channel parks itself. (closes issue
+ ASTERISK-19382) Reported by: James Stocks Patches by: dsessions
+ Review: https://reviewboard.asterisk.org/r/2148/ ........ Merged
+ revisions 375388 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/chan_dahdi.c, /: Merged revisions 375362 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375362 | rmudgett | 2012-10-29 10:51:24 -0500
+ (Mon, 29 Oct 2012) | 15 lines chan_dahdi: Fix segfault
+ dereferencing a NULL tech_pvt. The tech support customer was
+ using the AMI Redirect action shortly after a call was placed.
+ While the channel tried to do an ast_read(), the masquerade
+ resulting from the channel redirect took place. The masquerade in
+ the middle of the ast_read() resulted in the segfault. (closes
+ issue AST-1025) Reported by: Trey Blancher Patches:
+ jira_ast_1025_v1.8_v2.patch (license #5621) patch uploaded by
+ rmudgett ........ Merged revisions 375361 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-10-23 17:25 +0000 [r375290-375350] Automerge script <automerge@asterisk.org>
+
+ * contrib/scripts/ast_tls_cert, /: Merged revisions 375326 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375326 | jrose | 2012-10-23 11:21:22 -0500
+ (Tue, 23 Oct 2012) | 10 lines ast_tls_cert script: Better
+ response for various exit conditions to openssl (closes issue
+ ASTERISK-20260) Reported by: Daniel O'Connor Patches:
+ ast_tls_cert-update.diff uploaded by Daniel O'Connor (license
+ 6419) ........ Merged revisions 375325 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/app.c, /: Merged revisions 375300 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375300 | jrose | 2012-10-22 14:56:20 -0500
+ (Mon, 22 Oct 2012) | 14 lines core: Fix a memory leak in app.c
+ from an early return ast_app_group_match_get_count allocates
+ memory with the regcomp function and we previously forgot to free
+ it when bailing out due to a regex compilation failure against
+ category. (closes issue AST-1018) Reported by: Guenther Kelleter
+ Patches: regcomp_memleak.diff uploaded by Guenther Kelleter
+ (license 6372) ........ Merged revisions 375299 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * codecs/gsm/src/code.c, /: Merged revisions 375273 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375273 | jrose | 2012-10-22 12:08:49 -0500
+ (Mon, 22 Oct 2012) | 10 lines GSM: Fix encoding problems with GSM
+ (closes issue ASTERISK-20457) Reported by: Richard Miller
+ Patches: code.patch uploaded by Richard Miller (license 5685)
+ ........ Merged revisions 375272 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-10-18 21:25 +0000 [r375214-375241] Automerge script <automerge@asterisk.org>
+
+ * apps/app_queue.c, /: Merged revisions 375217 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375217 | jrose | 2012-10-18 16:09:29 -0500
+ (Thu, 18 Oct 2012) | 15 lines app_queue: Make ordering of
+ rrmemory/rrordered persist over add/remove members Prior to this
+ patch, adding, removing or reloading members to rrmemory would
+ cause the order to become completely jumbled. Now it behaves more
+ or less like rrordered other than the fact that it stores the
+ members on a hash table rather than a linked list. This patch
+ also prevents removal of members and member reloads from jumbling
+ rrordered queues. (issue AST-989) Reported by: Thomas Arimont
+ Review: https://reviewboard.asterisk.org/r/2164/ ........ Merged
+ revisions 375216 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * configure.ac, makeopts.in, Makefile, /, build_tools/make_version,
+ configure: Merged revisions 375190 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375190 | rmudgett | 2012-10-18 14:53:08 -0500
+ (Thu, 18 Oct 2012) | 36 lines build_tools: Allow Asterisk to
+ report git SHAs in version string. Make git more attractive for
+ managing work-in-progress. Especially convenient when a potential
+ patch set needs to be tested on multiple platforms since one can
+ use git to keep all the test environments in sync independent of
+ a subversion server. Now the Asterisk version will show the exact
+ git SHA5 that was used when building (still appended by "M" if
+ there are local modifications) from a git clone of the Asterisk
+ repository so the developer can more easily know what is actually
+ under test. You will now get this: $ asterisk -V Asterisk
+ GIT-1698298 Instead of this: $ asterisk -V Asterisk
+ UNKNOWN__and_probably_unsupported This has zero impact for those
+ not using git with the exception of an extra test in the
+ configure script to gather git's path. This is necessary to
+ prevent "sudo make install" from failing since git may not be in
+ the path in make's shell environment. (closes issue
+ ASTERISK-20483) Reported by: Shaun Ruffell Patches:
+ 0001-build_tools-Allow-Asterisk-to-report-git-SHAs-in-ver.patch
+ (license #5417) patch uploaded by Shaun Ruffell Modified ........
+ Merged revisions 375189 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-10-17 19:26 +0000 [r375043-375173] Automerge script <automerge@asterisk.org>
+
+ * main/tcptls.c, /: Merged revisions 375147 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375147 | kmoore | 2012-10-17 13:58:52 -0500
+ (Wed, 17 Oct 2012) | 15 lines Ensure Asterisk fails TCP/TLS SIP
+ calls when certificate checking fails When placing a call to a
+ TCP/TLS SIP endpoint whose certificate is not signed by a
+ configured CA certificate, Asterisk would issue a warning and
+ continue to process the call as if there was not an issue with
+ the certificate. Asterisk now properly fails the call if the
+ certificate fails verification or if the certificate does not
+ exist when certificate checking is enabled (the default
+ behavior). (closes issue ASTERISK-20559) Review:
+ https://reviewboard.asterisk.org/r/2163/ ........ Merged
+ revisions 375146 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 375112 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375112 | wdoekes | 2012-10-16 16:43:29 -0500
+ (Tue, 16 Oct 2012) | 10 lines Fixes to the fd-oriented SIP TCP
+ reads. Don't crash on large user input. Allow SIP headers without
+ space. Optimize code a bit. Review:
+ https://reviewboard.asterisk.org/r/2162 ........ Merged revisions
+ 375111 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 375078 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375078 | wdoekes | 2012-10-16 14:22:44 -0500
+ (Tue, 16 Oct 2012) | 7 lines Update sip_request_call SIP dial
+ string documentation. This was missed when merging review r1859.
+ ........ Merged revisions 375074 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * contrib/scripts/autosupport, /: Merged revisions 375060 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375060 | tzafrir | 2012-10-16 14:16:43 -0500
+ (Tue, 16 Oct 2012) | 10 lines autosupport: fix bashism '==' is
+ bashism (bashspecific, fails when dash is /bin/sh). Anyway, a
+ 'case' works better there. (closes issue ASTERISK-20567) ........
+ Merged revisions 375059 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * include/asterisk/strings.h, channels/chan_iax2.c,
+ apps/app_dial.c, /, main/ccss.c: Merged revisions 375026 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r375026 | mmichelson | 2012-10-15 16:06:42 -0500
+ (Mon, 15 Oct 2012) | 22 lines Fix some potential misuses of
+ ast_str in the code. Passing an ast_str pointer by value that
+ then calls ast_str_set(), ast_str_set_va(), ast_str_append(), or
+ ast_str_append_va() can result in the pointer originally passed
+ by value being invalidated if the ast_str had to be reallocated.
+ This fixes places in the code that do this. Only the example in
+ ccss.c could result in pointer invalidation though since the
+ other cases use a stack-allocated ast_str and cannot be
+ reallocated. I've also updated the doxygen in strings.h to
+ include notes about potential misuse of the functions mentioned
+ previously. Review: https://reviewboard.asterisk.org/r/2161
+ ........ Merged revisions 375025 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-10-14 10:20 +0000 [r374994] Automerge script <automerge@asterisk.org>
+
+ * config.guess, config.sub, /: Merged revisions 374991 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374991 | tzafrir | 2012-10-14 04:40:24 -0500
+ (Sun, 14 Oct 2012) | 12 lines Update config.guess and config.sub:
+ 2012-10-10 Update config.guess and config.sub to revision
+ fb456b34ef4aa02b95dc6be69aaa66fa94a844fb from the
+ savannah.gnu.org git repo. Adds support for e.g. aarch64 (ARM
+ 64bit). config.guess:timestamp='2012-09-25'
+ config.sub:timestamp='2012-10-10' ........ Merged revisions
+ 374977 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-10-12 21:56 +0000 [r374931] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_voicemail.c: Avoid a segfault on invalid format names If
+ a format name was not found by ast_getformatbyname, a NULL
+ pointer would be passed into ast_format_rate and immediately
+ dereferenced. This ensures that a valid pointer is used since the
+ structure is already allocated on the stack. (closes issue
+ DPH-523) Reported-by: Steve Pitts
+
+2012-10-12 16:23 +0000 [r374720-374923] Automerge script <automerge@asterisk.org>
+
+ * include/asterisk/tcptls.h, main/tcptls.c, /, channels/chan_sip.c:
+ Merged revisions 374906 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374906 | mmichelson | 2012-10-12 11:11:30 -0500
+ (Fri, 12 Oct 2012) | 28 lines Do not use a FILE handle when doing
+ SIP TCP reads. This is used to solve an issue where a poll on a
+ file descriptor does not necessarily correspond to the readiness
+ of a FILE handle to be read. This change makes it so that for TCP
+ connections, we do a recv() on the file descriptor instead.
+ Because TCP does not guarantee that an entire message or even
+ just one single message will arrive during a read, a loop has
+ been introduced to ensure that we only attempt to handle a single
+ message at a time. The tcptls_session_instance structure has also
+ had an overflow buffer added to it so that if more than one TCP
+ message arrives in one go, there is a place to throw the excess.
+ Huge thanks goes out to Walter Doekes for doing extensive review
+ on this change and finding edge cases where code could fail.
+ (closes issue ASTERISK-20212) reported by Phil Ciccone Review:
+ https://reviewboard.asterisk.org/r/2123 ........ Merged revisions
+ 374905 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/cdr.c, /: Merged revisions 374844 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374844 | mjordan | 2012-10-11 10:43:19 -0500
+ (Thu, 11 Oct 2012) | 29 lines Fix incorrect billing duration
+ reported when batch mode is enabled Similar to r369351, the
+ billing duration can be skewed when batch mode is enabled. This
+ happened much more rarely than the duration, as it only occured
+ when the call was answered (thereby indicating an actual answer
+ time) and immediately hung up on (indicating a billsec of 0).
+ Since a billing time of '0' can either mean that the call
+ immediately ended or that the CDR was improperly answered, we
+ have to use additional information to know whether or not we can
+ trust the CDR billsec value. Prior to this patch, we looked to
+ see if we had a valid answer time. If we did, and billsec was
+ zero, we used the current time to calculate what billsec value we
+ could from the CDR being written. If batch mode is enabled, this
+ will incorrectly report a billsec value being much greater than
+ the actual duration of the call. Instead of relying on the
+ presence of an answer time to know whether or not we can
+ re-calculate the billsec for the CDR, we now also use the
+ presence of the CDR's end time to know if we need to re-calculate
+ or whether we can trust the billsec value that we have. This
+ prevents erroneous jumps in the billsec value, while still making
+ sure that in the worst case, some billing time will be
+ calculated. (closes issue AST-1016) Reported by: Thomas Arimont
+ Tested by: Thomas Arimont ........ Merged revisions 374843 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_queue.c, /: Merged revisions 374803 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374803 | rmudgett | 2012-10-10 15:55:44 -0500
+ (Wed, 10 Oct 2012) | 30 lines app_queue: Made pass connected line
+ updates from the caller to ringing queue members. Party A calls
+ Party B Party B puts Party A on hold. Party B calls a queue.
+ Ringing queue member D sees Party B identification. Party B
+ transfers Party A to the queue. Queue member D does not get a
+ connected line update for Party A. Queue member D answers the
+ call and still sees Party B information. However, if Party A
+ later transfers the call to Party C then queue member D gets a
+ connected line update for Party C. * Made pass connected line
+ updates from the caller to queue members while the queue members
+ are ringing. (closes issue AST-1017) Reported by: Thomas Arimont
+ (closes issue ABE-2886) Reported by: Thomas Arimont Tested by:
+ rmudgett ........ Merged revisions 374801 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 374802 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/pbx.c, /: Merged revisions 374763 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374763 | rmudgett | 2012-10-09 17:19:26 -0500
+ (Tue, 09 Oct 2012) | 15 lines Fix execution of 'i' extension due
+ to uninitialized variable. The fix for ASTERISK-18243 added code
+ that could potentially use dst_exten[] uninitialized. As a result
+ the 'i' exten may not be executed when it should. (closes issue
+ ASTERISK-20455) Reported by: Richard Miller Patches:
+ pbx-1.8.16.0.diff (license #5685) patch uploaded by Richard
+ Miller Made some cosmetic modifications. ........ Merged
+ revisions 374758 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * configs/chan_dahdi.conf.sample, /: Merged revisions 374728 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374728 | rmudgett | 2012-10-08 17:29:47 -0500
+ (Mon, 08 Oct 2012) | 15 lines dahdi.conf.sample: Add description
+ for "buffers" setting. This contains an edited version of the
+ patch originally uploaded by John Bigelow. (closes issue
+ ASTERISK-14435) Reported by: John Bigelow Patches: buffers.patch
+ (license #5091) patch uploaded by John Bigelow
+ 0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch
+ (license #5417) patch uploaded by Shaun Ruffell Modified ........
+ Merged revisions 374727 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * pbx/pbx_spool.c, /: Merged revisions 374695 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374695 | rmudgett | 2012-10-08 16:11:41 -0500
+ (Mon, 08 Oct 2012) | 34 lines Fix deletion of unopenable spool
+ files. If scan_service() cannot open the spool file, it logs a
+ message saying that it will delete the file and calls
+ remove_from_queue() to do it. However, remove_from_queue() fails
+ to delete the spool file because struct outgoing has not yet been
+ fully initialized. * Merged allocating a new struct outgoing and
+ init_outgoing() into new_outgoing(). Allocation is
+ initialization. * Made apply_outgoing() not initialize the spool
+ filename in struct outgoing. * Made apply_outgoing() call
+ ast_trim_blanks() and ast_skip_blanks() rather than manually
+ inlining them. * Reduced indentation levels in apply_outgoing().
+ * Fixed a garbled comment in remove_from_queue(). * Reworked
+ scan_service() to simplify it. (closes issue ASTERISK-17231)
+ Reported by: David Chappell Patches: spool_open_failure.diff
+ (license #4997) patch uploaded by David Chappell Started with
+ this patch. ........ Merged revisions 374686 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 * Fixed some
+ memory leaks on of nominal paths in init_outgoing() when merging
+ into the new_outgoing() function dealing with o->capabilities.
+ ................
+
+2012-11-06 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.10.0-digiumphones Released.
+
+2012-11-05 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.10.0-digiumphones-rc2 Released.
+
+ * Fix a bug which made ConfBridge not record the conference when
+ the recording was initiated from an AMI/CLI command
+
+ (closes issue ASTERISK-20601)
+ Reported by: Vilius
+
+2012-10-08 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.10.0-digiumphones-rc1 Released.
+
+2012-10-08 14:19 +0000 [r374656] Automerge script <automerge@asterisk.org>
+
+ * apps/confbridge/include/conf_state.h (added),
+ apps/confbridge/conf_state_multi.c (added),
+ apps/app_confbridge.c, apps/confbridge/conf_state_multi_marked.c
+ (added), apps/confbridge/conf_state_empty.c (added),
+ apps/confbridge/conf_state.c (added),
+ apps/confbridge/conf_state_single.c (added),
+ apps/confbridge/conf_state_inactive.c (added),
+ apps/confbridge/conf_state_single_marked.c (added), /,
+ apps/confbridge/include/confbridge.h: Merged revisions 374652 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ........ r374652 | mjordan | 2012-10-08 08:46:27 -0500 (Mon, 08
+ Oct 2012) | 46 lines Resolve issues in ConfBridge regarding
+ marked, waitmarked, and unmarked users Thank's to Neil Tallim
+ (flan)'s tireless testing, issue reporting, and patches it became
+ clear that app_confbridge had some complex logic in how it
+ handled interactions between marked, waitmarked, and unmarked
+ users. In particular, there were some areas in which the
+ interactions between the users resulted in inconsistent behavior,
+ and app_confbridge was missing logic in how to handle some corner
+ cases. Some areas included: * Poor handling of mixing unmarked
+ and waitmarked users * Inconsistencies in how MOH and muting was
+ applied to various users * Handling of various announcements for
+ different user profile options flan's patches seem to fix the
+ various issues, but highlighted how hard the code could be to
+ maintain. In an attempt to make things easier to maintain and to
+ more fully enumerate the various cases that exist, this patch
+ breaks up the logic into a state machine-like setup. Please note
+ that the various state transitioned are documented on the
+ Asterisk wiki:
+ https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes
+ Review: //https://reviewboard.asterisk.org/r/2072/ Note that for
+ the following issues, mjordan uploaded the patch, although it was
+ written by twilson. Any contributor license discrepency is due to
+ that. (closes issue ASTERISK-19562) Reported by: flan Tested by:
+ flan, mjordan, jrose patches:
+ bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
+ twilson (license 6283) (closes issue ASTERISK-19726) Reported by:
+ flan Tested by: flan patches:
+ bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
+ twilson (license 6283) (closes issue ASTERISK-20181) Reported by:
+ Jonathan White Tested by: Jonathan White patches:
+ bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
+ twilson (license 6283) ........
+
+2012-10-05 21:25 +0000 [r374226-374610] Automerge script <automerge@asterisk.org>
+
+ * main/manager.c, /: Merged revisions 374586 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374586 | dlee | 2012-10-05 15:23:14 -0500 (Fri,
+ 05 Oct 2012) | 34 lines Multiple revisions 374570,374581 ........
+ r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) |
+ 22 lines Improve AMI long line error handling In AMI's parser,
+ when it receives a long line (> 1024 characters), it discards
+ that line, but continues to process the message normally.
+ Typically, this is not a problem because a) who has lines that
+ long and b) usually a discarded line results in an invalid
+ message. But if that line is specifying an optional field, then
+ the message will be processed, you get a 'Response: Success', but
+ things don't work the way you expected them to. This patch
+ changes the behavior when a line-too-long parse error occurs. *
+ Changes the log message to avoid way-too-long (and truncated
+ anyways) log messages * Adds a 'parsing' status flag to Response:
+ Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line
+ is too long * Responds with an appropriate error if parsing !=
+ MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow
+ Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581
+ | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line
+ I've committed too much. Reverting part of r374570. ........
+ Merged revisions 374570,374581 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c,
+ channels/misdn/isdn_lib.h, channels/chan_misdn.c, /: Merged
+ revisions 374537 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374537 | rmudgett | 2012-10-05 13:25:20 -0500
+ (Fri, 05 Oct 2012) | 162 lines Merged revisions 374515-374535
+ from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500
+ (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode *
+ Made setup_bc() static. Patches: patch1_unused-code.diff (license
+ #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882
+ ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500
+ (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan
+ states Patches: patch2_unused-states.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012)
+ | 16 lines chan_misdn: Remove unnecessary null pointer checks and
+ checks for stack->nt * cleanup_bc() is always called with valid
+ bc (or it would've crashed before). * Value of stack->nt is known
+ in advance at some places. * Rename handle_event() to
+ handle_event_te(), handle_frm() to handle_frm_te(). Patches:
+ patch3_checks.diff (license #6372) patch uploaded by Guenther
+ Kelleter Modified JIRA ABE-2882 ................ r374518 |
+ rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Fix spelling in log messages Patches:
+ patch4_spelling.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2882 ................ r374519 | rmudgett |
+ 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines
+ chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after
+ calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is
+ emptied, cleaned and set not in use, although
+ misdn_lib_send_event() already did the same. This is bad. When
+ it's not in use we are not allowed to touch it. * Moved log
+ message in front of the resulting actions and fixed it to match
+ the case. Patches: patch5_bccleanup.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012)
+ | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up
+ etc., really bad stuff. * Fix return codes of cb_events() for
+ EVENT_SETUP to use caller's cleanup mechanisms. * Move
+ cl_queue_chan() call after bearer check. Patches:
+ patch6_leaks.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2882 ................ r374521 | rmudgett |
+ 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines
+ chan_misdn: We must initialize cause on sending a DISCONNECT. We
+ must initialize cause on sending a DISCONNECT, so it is later
+ correctly indicated to ast_channel in case the answer
+ (RELEASE/RELEASE_COMPLETE) does not include one. Patches:
+ patch7_hangupcause.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2882 ................ r374522 |
+ rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Remove unused code for upqueue Patches:
+ patch8_unused-upqueue.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2882 ................ r374523 |
+ rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Improve debugging (port number, messages fixed, dups
+ removed) Patches: patch9_debug.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012)
+ | 8 lines chan_misdn: Better debug: we can print_bc_info even if
+ there's no ast leg. Patches: patch10_debug-bc-2.diff (license
+ #6372) patch uploaded by Guenther Kelleter Modified. JIRA
+ ABE-2882 ................ r374534 | rmudgett | 2012-10-05
+ 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn:
+ setup_bc() is called too early for an incoming SETUP on TE. This
+ prevents the B channel from being setup for HDLC mode when
+ requested by the bearer capability and config option hdlc=yes. It
+ violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not
+ connect to the channel until a CONNECT ACKNOWLEDGE message has
+ been received." * Call setup_bc() on receipt of
+ CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for
+ PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by
+ Guenther Kelleter Modified. JIRA ABE-2881 ................
+ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012)
+ | 2 lines chan_misdn: Remove some more deadcode. ................
+ ........ Merged revisions 374536 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * configs/dsp.conf.sample, CHANGES, main/dsp.c, /: Merged revisions
+ 374476,374481 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374476 | alecdavis | 2012-10-04 15:05:14 -0500
+ (Thu, 04 Oct 2012) | 13 lines dsp.c fix incorrect DTMF
+ Digit_Duration. it's always short by 'hits_to_begin*DTMF_GSIZE',
+ or 25.5ms if hitstobegin=2 (issue ASTERISK-16003) Tested by:
+ alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2145/ ........ Merged
+ revisions 374475 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r374481 | alecdavis | 2012-10-04 15:17:16 -0500
+ (Thu, 04 Oct 2012) | 17 lines dsp.c User Configurable
+ DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END Instead of a recompile,
+ allow values to be adjusted in dsp.conf For binary distributions
+ allows easy adjustment for wobbly GSM calls, and other reasons.
+ Defaults to DTMF_HITS_TO_BEGIN=2 and DTMF_MISSES_TO_END=3 (closes
+ issue ASTERISK-17493) Tested by: alecdavis alecdavis (license
+ 585) Review https://reviewboard.asterisk.org/r/2144/ ........
+ Merged revisions 374479 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 374457 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374457 | file | 2012-10-04 12:44:38 -0500 (Thu,
+ 04 Oct 2012) | 17 lines Fix a regression from direct media ACLs
+ where the directrtpsetup option no longer works. A check was
+ added for direct media ACLs that immediately forbid remote
+ bridging if there was no bridged channel. This caused
+ directrtpsetup to no longer function as it needs this information
+ before bridging actually occurs. Logic has now been adjusted so
+ if there is no bridged channel a remote bridge will still be
+ attempted. (closes issue ASTERISK-20511) Reported by: kristoff
+ Review: https://reviewboard.asterisk.org/r/2146/ ........ Merged
+ revisions 374456 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * res/res_agi.c, main/db.c, /: Merged revisions 374427 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374427 | dlee | 2012-10-04 10:37:11 -0500 (Thu,
+ 04 Oct 2012) | 25 lines Fix DBDelTree error codes for AMI, CLI
+ and AGI The AMI DBDelTree command will return Success/Key tree
+ deleted successfully even if the given key does not exist. The
+ CLI command 'database deltree' had a similar problem, but was
+ saved because it actually responded with '0 database entries
+ removed'. AGI had a slightly different error, where it would
+ return success if the database was unavailable. This came from
+ confusion about the ast_db_deltree retval, which is -1 in the
+ event of a database error, or number of entries deleted
+ (including 0 for deleting nothing). * Changed some poorly named
+ res variables to num_deleted * Specified specific errors when
+ calling ast_db_deltree (database unavailable vs. entry not found
+ vs. success) * Fixed similar bug in AGI database deltree, where
+ 'Database unavailable' results in successful result (closes issue
+ AST-967) Reported by: John Bigelow Review:
+ https://reviewboard.asterisk.org/r/2138/ ........ Merged
+ revisions 374426 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * configs/dsp.conf.sample, CHANGES, main/dsp.c, /: Merged revisions
+ 374385 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374385 | alecdavis | 2012-10-03 23:41:19 -0500
+ (Wed, 03 Oct 2012) | 36 lines dsp.c User configuration of
+ DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values Asterisk's DTMF
+ Specifications are based on AT&T specs, which may not be
+ compatible in other countries. Various countries have different
+ specifications for the maximum power level differences between
+ the DTMF low group and high group of frequencies. Power level
+ difference between frequencies for different
+ Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to
+ 8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian
+ = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1
+ (2006-03) Now allow 4 variables to be individually configured in
+ dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T
+ specifications Add's the following variables to dsp.conf
+ ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51
+ ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98
+ (closes issue ASTERISK-20442) Reported by: tbsky Tested by:
+ tbsky,alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2141/ ........ Merged
+ revisions 374384 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/dsp.c, /: Merged revisions 374370 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374370 | alecdavis | 2012-10-03 23:18:44 -0500
+ (Wed, 03 Oct 2012) | 15 lines _dsp_init: bring inline with trunk
+ preparation for clean merge of DTMF TWIST patch No functional
+ changes, just style. alecdavis (license 585) Reported by: Alec
+ Davis Tested by: alecdavis related
+ https://reviewboard.asterisk.org/r/2141 ........ Merged revisions
+ 374365 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * res/res_jabber.c, /: Merged revisions 374336 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374336 | mjordan | 2012-10-03 21:11:05 -0500
+ (Wed, 03 Oct 2012) | 31 lines Check for presence of buddy in
+ info/dinfo handlers The res_jabber resource module uses the
+ ASTOBJ library for managing its ref counted objects. After
+ calling ASTOBJ_CONTAINER_FIND to locate a buddy object, the
+ pointer to the object has to be checked to see if the buddy
+ existed. Prior to this patch, the buddy object was not checked
+ for NULL; with this patch in both aji_client_info_handler and
+ aji_dinfo_handler the pointer is checked before used and, if no
+ buddy object was found, the handlers return an error code. This
+ patch does not take the approach that our JID can be used to log
+ in from another resource. If that approach is desired, an
+ improvement could be made to this patch to create the buddy on
+ the fly. This patch seeks only to prevent Asterisk from crashing.
+ Note that multiple people have proposed patches for this issue;
+ the patch being committed here is based on those. (closes issue
+ ASTERISK-19532) Reported by: Karsten Wemheuer Tested by: Byron
+ Clark patches: fix-jabber uploaded by Karsten Wemheuer (license
+ #5930) xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark
+ (license #6157) (closes issue ASTERISK-19557) Reported by:
+ ulugutz ........ Merged revisions 374335 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, main/ccss.c: Merged revisions 374300 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r374300 | mjordan | 2012-10-03 12:25:36 -0500 (Wed, 03 Oct 2012)
+ | 10 lines Destroy the generic_monitors container after the
+ core_instances in ccss For each item in core_instances disposed
+ of in the shutdown of ccss, any generic monitor instances
+ referenced by the objects will be removed from generic_monitors
+ during their destruction. Hilarity ensues if generic_monitors no
+ longer exists. Thanks to the Asterisk Test Suite's generic_ccss
+ test for complaining loudly when it ran into this. ........
+
+ * main/asterisk.c, /: Merged revisions 374231 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374231 | mjordan | 2012-10-02 16:12:30 -0500
+ (Tue, 02 Oct 2012) | 9 lines Ensure Shutdown AMI event is still
+ fired during Asterisk shutdown Richard pointed out that having
+ the manager dispose of itself gracefully during shutdown meant
+ that the Shutdown event will no longer get fired. This patch
+ moves the AMI event just prior to running the atexit callbacks.
+ ........ Merged revisions 374230 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/message.c, /: Merged revisions 374210 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r374210 | mjordan | 2012-10-02 12:10:04 -0500 (Tue, 02 Oct 2012)
+ | 10 lines Fix findings from check-in on r374177 Richard pointed
+ out two problems with the check-in from r374177: * The
+ ast_msg_shutdown function declaration doesn't match the prototype
+ in main/message.c. * The ref/alloc function usage in astobj2 (in
+ 11+) can use the ao2_t_* variants of the functions to allow the
+ REF_DEBUG flag to enable/disable their debug counterparts.
+ ........
+
+2012-10-02 16:41 +0000 [r374208-374209] Jason Parker <jparker@digium.com>
+
+ * /: Re-enable automerge.
+
+ * channels/chan_agent.c, main/features.c, main/cel.c,
+ main/format_pref.c, main/indications.c, main/message.c,
+ main/asterisk.c, main/db.c, main/channel.c, main/format.c,
+ main/data.c, main/pbx.c, main/manager.c, /, main/ccss.c: Fix a
+ variety of ref counting issues This patch resolves a number of
+ ref leaks that occur primarily on Asterisk shutdown. It adds a
+ variety of shutdown routines to core portions of Asterisk such
+ that they can reclaim resources allocate duringd initialization.
+ Review: https://reviewboard.asterisk.org/r/2137 ........ Merged
+ revisions 374177 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374178 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-02 01:23 +0000 [r374148-374195] Automerge script <automerge@asterisk.org>
+
+ * /: automerge cancel
+
+ * tests/test_db.c, apps/app_queue.c, main/db.c,
+ include/asterisk/astdb.h, /: Merged revisions 374132,374135 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374132 | seanbright | 2012-10-01 12:27:22 -0500
+ (Mon, 01 Oct 2012) | 2 lines Use ast_copy_string instead of
+ strncpy to guarantee a NUL terminated string. ................
+ r374135 | seanbright | 2012-10-01 12:52:38 -0500 (Mon, 01 Oct
+ 2012) | 23 lines app_queue: Support persisting and loading of
+ long member lists. Greenlight in #asterisk brought up that he was
+ receiving an error message "Could not create persistent member
+ string, out of space" when running app_queue in Asterisk 10.
+ dump_queue_members() made an assumption that 8K would be enough
+ to store the generated string, but with queues that have large
+ member lists this is not always the case. This patch removes the
+ limitation and uses ast_str instead of a fixed sized buffer. The
+ complicating factor comes from the fact that ast_db_get requires
+ a buffer and buffer size argument, which doesn't let us pull back
+ more than what we pass in, so I introduced a new
+ ast_db_get_allocated() which returns an ast_strdup()'d copy of
+ the value from astdb. As an aside, I did some testing on the
+ maximum size of data that we can store in the BDB library we
+ distribute and was able to store a 10MB string and retrieve it
+ with no problems, so I feel this is a safe patch. Review:
+ https://reviewboard.asterisk.org/r/2136/ ........ Merged
+ revisions 374108 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-09-28 19:25 +0000 [r373498-374058] Automerge script <automerge@asterisk.org>
+
+ * res/res_jabber.c, /: Merged revisions 374045 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r374045 | jrose | 2012-09-28 14:21:10 -0500
+ (Fri, 28 Sep 2012) | 12 lines res_jabber: Remove CLI command
+ 'jabber test' The opinion of development was that it is both
+ improper to have Matt's personal email address used in the source
+ and that the command wouldn't be useful without it. (closes issue
+ AST-467) Reported by: Malcolm Davenport ........ Merged revisions
+ 374032 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * res/res_agi.c, /: Merged revisions 373990 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373990 | file | 2012-09-28 07:15:48 -0500 (Fri,
+ 28 Sep 2012) | 8 lines Update documentation to make it explicit
+ that "stream file" will not restart musiconhold. (issue
+ ASTERISK-17367) Reported by: oej ........ Merged revisions 373989
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_senddtmf.c, /: Merged revisions 373946 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373946 | rmudgett | 2012-09-27 17:12:47 -0500
+ (Thu, 27 Sep 2012) | 14 lines Fix SendDTMF crash and channel
+ reference leak using channel name parameter. The SendDTMF channel
+ name parameter has two issues. 1) Crashes if the channel name
+ does not exist. 2) Leaks a channel reference if the channel is
+ the current channel. Problem introduced by ASTERISK-15956. *
+ Updated SendDTMF documentation. * Renamed app to senddtmf_name
+ and tweaked the type. ........ Merged revisions 373945 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/loader.c, /: Merged revisions 373910 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373910 | file | 2012-09-27 11:50:46 -0500 (Thu,
+ 27 Sep 2012) | 24 lines loader: Ensure dependent modules are
+ properly initialized. If an Asterisk module specifies a
+ dependency in ast_module_info.nonoptreq, it is possible for
+ Asterisk to skip calling the modules's .load function. Asterisk
+ was loading and linking the module via load_dynamic_module() but
+ was not adding the module to the resource_heap. Therefore the
+ module was not initialized based on it's priority along with the
+ other modules in the heap. Now use load_resource() instead of
+ load_dynamic_module() for non-optional requirement. This will add
+ the module to the resource_heap so the module can be properly
+ initialized in the correct order. This is required if there are
+ any module global data structures initialized in the .load()
+ callback for the module on platforms which do not support weak
+ references. (issue ASTERISK-20439) Reported by: sruffell Patches:
+ 0001-loader-Ensure-dependent-modules-are-properly-initial.patch
+ uploaded by sruffell (license 5417) ........ Merged revisions
+ 373909 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/chan_local.c, /: Merged revisions 373879 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373879 | file | 2012-09-27 06:32:13 -0500 (Thu,
+ 27 Sep 2012) | 14 lines Fix an issue where Local channels dialed
+ by app_queue are considered in use immediately. The chan_local
+ channel driver returns a device state of in use even if a created
+ Local channel has not yet been dialed. This fix changes the logic
+ to return a state of not in use until the channel itself has been
+ dialed. (closes issue ASTERISK-20390) Reported by: tim_ringenbach
+ Review: https://reviewboard.asterisk.org/r/2116/ ........ Merged
+ revisions 373878 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 373849 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373849 | mmichelson | 2012-09-26 16:11:35 -0500
+ (Wed, 26 Sep 2012) | 8 lines Move handling of 408 response so
+ there is no misleading warning message. (closes issue
+ ASTERISK-20060) Reported by: Walter Doekes ........ Merged
+ revisions 373848 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, apps/app_meetme.c: Merged revisions 373816 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373816 | rmudgett | 2012-09-26 13:15:50 -0500
+ (Wed, 26 Sep 2012) | 18 lines Fixed meetme tab completion and
+ command documentation. * Removed unnecessary case sensitivity in
+ meetme list, lock, unlock, mute, unmute, and kick commands. *
+ Separated meetme lock/unlock, mute/unmute, and kick commands into
+ their own registered commands to simplify tab completion and
+ parameter checking. meetme_lock_cmd(), meetme_mute_cmd(), and
+ meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue
+ AST-1006) Reported by: John Bigelow Tested by: rmudgett ........
+ Merged revisions 373815 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/chan_agent.c, configs/agents.conf.sample, /, main/say.c:
+ Merged revisions 373769,373774 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373769 | mmichelson | 2012-09-25 17:54:13 -0500
+ (Tue, 25 Sep 2012) | 11 lines Remove dead code and documentation
+ for nonexistent feature. multiplelogin was removed from
+ chan_agent back in 1.6.0 when AgentCallbackLogin() was removed.
+ (closes issue AST-948) reported by Steve Pitts ........ Merged
+ revisions 373768 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r373774 | mmichelson | 2012-09-25 18:08:46 -0500
+ (Tue, 25 Sep 2012) | 10 lines Fix saying of date in Dutch. The
+ Dutch say the date before the month. (closes issue
+ ASTERISK-20353) Reported by: Teun Ouwehand ........ Merged
+ revisions 373773 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 373737 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373737 | mmichelson | 2012-09-25 16:12:40 -0500
+ (Tue, 25 Sep 2012) | 11 lines Fix error where improper IMAP
+ greetings would be deleted. (closes issue ASTERISK-20435)
+ Reported by: fhackenberger Patches:
+ asterisk-20435-imap-del-greeting.diff uploaded by Michael L.
+ Young (License #5026) (with suggested modification made by me)
+ ........ Merged revisions 373735 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * res/res_rtp_asterisk.c, channels/chan_local.c, /: Merged
+ revisions 373703,373706 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373703 | kmoore | 2012-09-25 14:34:01 -0500
+ (Tue, 25 Sep 2012) | 11 lines Fix an issue where media would not
+ flow for situations where the legacy STUN code is in use. The
+ STUN packets should *not* be blocked by strict RTP. (closes issue
+ ASTERISK-20415) Reported-by: Michele Cicciotti Patch-by: Josh
+ Colp (trunk r369817) ........ Merged revisions 373702 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r373706 | file | 2012-09-25 15:12:02 -0500 (Tue,
+ 25 Sep 2012) | 22 lines Fix T.38 support when used with
+ chan_local in between. Users of the T.38 API can indicate
+ AST_T38_REQUEST_PARMS on a channel to request that the channel
+ indicate a T.38 negotiation with the parameters present on the
+ channel. The return value of this indication is expected to be
+ AST_T38_REQUEST_PARMS upon success but with chan_local involved
+ this could never occur. This fix changes chan_local to always
+ return AST_T38_REQUEST_PARMS for this situation. If the
+ underlying channel technology on the other side does not support
+ T.38 this would have been determined ahead of time using
+ ast_channel_get_t38_state and an indication would not occur.
+ (closes issue ASTERISK-20229) Reported by: wdoekes Patches:
+ ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review:
+ https://reviewboard.asterisk.org/r/2070/ ........ Merged
+ revisions 373705 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * configs/sip.conf.sample, apps/app_queue.c,
+ channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+ revisions 373665,373675 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373665 | twilson | 2012-09-25 12:35:30 -0500
+ (Tue, 25 Sep 2012) | 21 lines Properly handle UAC/UAS roles for
+ SIP session timers The SIP session timer mechanism contains a
+ mandatory 'refresher' parameter (included in the Session-Expires
+ header) which is used in the session timer offer/answer signaling
+ within a SIP Invite dialog. It looks like asterisk is
+ interpreting the uac resp. uas role only as the initial role of
+ client and server (caller is uac, callee is uas). The standard
+ rfc 4028 however assigns the client role to the ((RE)-Invite)
+ requester, the server role to the ((RE)-Invite) responder. This
+ patch has Asterisk track the actual refresher as "us" or "them"
+ as opposed to relying on just the configured "uas" or "uac"
+ properties. (closes issue AST-922) Reported by: Thomas Airmont
+ Review: https://reviewboard.asterisk.org/r/2118/ ........ Merged
+ revisions 373652 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r373675 | kmoore | 2012-09-25 13:20:04 -0500
+ (Tue, 25 Sep 2012) | 13 lines "show" completion option for
+ "queue" shouldn't appear twice When tab-completing CLI commands
+ starting with "queue", "show" appeared twice in the list due to
+ the way that Asterisk's tab completion functions and the order in
+ which the commands were registered. The registration order has
+ been altered to resolve this issue. (closes issue AST-940)
+ Reported-by: Steve Pitts ........ Merged revisions 373666 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * codecs/ilbc/iLBC_decode.c, codecs/Makefile, /,
+ channels/chan_sip.c, codecs/ilbc/iLBC_encode.c: Merged revisions
+ 373631,373633,373645 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373631 | jrose | 2012-09-25 11:24:34 -0500
+ (Tue, 25 Sep 2012) | 10 lines chan_sip: Set Quality of Service
+ for video rtp instance (closes issue ASTERISK-20201) Reported by:
+ ddkprog Patches: chan_sip.c.diff uploaded by ddkprog (license
+ 6008) ........ Merged revisions 373617 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r373633 | rmudgett | 2012-09-25 11:33:31 -0500
+ (Tue, 25 Sep 2012) | 5 lines Make rebuild GSM, ilbc, or lpc10
+ codecs if the respective sources change. ........ Merged
+ revisions 373618 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r373645 | rmudgett | 2012-09-25 12:19:52 -0500
+ (Tue, 25 Sep 2012) | 14 lines Fix valgrind found memcpy issues in
+ codec_ilbc. Valgrind found codec_ilbc using memcpy instead of
+ memmove for overlapping memory blocks. (issue ASTERISK-19890)
+ (closes issue ASTERISK-20231) Reported by: Walter Doekes Patches:
+ ASTERISK-20231.patch (license #5674) patch uploaded by Walter
+ Doekes ........ Merged revisions 373640 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * configs/res_odbc.conf.sample, /: Merged revisions 373579 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373579 | kmoore | 2012-09-25 08:28:20 -0500
+ (Tue, 25 Sep 2012) | 11 lines Fix documentation for default
+ username in res_odbc This was previously stated to be "root", but
+ is actually the name of the context if unspecified. (closes issue
+ ASTERISK-20258) Reported by: Stefan x ........ Merged revisions
+ 373578 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * res/res_rtp_multicast.c, /: Merged revisions 373551 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373551 | file | 2012-09-25 07:00:23 -0500 (Tue,
+ 25 Sep 2012) | 15 lines Fix an issue where a caller to ast_write
+ on a MulticastRTP channel would determine it failed when in
+ reality it did not. When sending RTP packets via multicast the
+ amount of data sent is stored in a variable and returned from the
+ write function. This is incorrect as any non-zero value returned
+ is considered a failure while a return value of 0 is success. For
+ callers (such as ast_streamfile) that checked the return value
+ they would have considered it a failure when in reality nothing
+ went wrong and it was actually a success. The write function for
+ the multicast RTP engine now returns -1 on failure and 0 on
+ success, as it should. (closes issue ASTERISK-17254) Reported by:
+ wybecom ........ Merged revisions 373550 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 373533 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373533 | file | 2012-09-24 19:11:28 -0500 (Mon,
+ 24 Sep 2012) | 5 lines Add missing checks that I neglected. The
+ SIP technology and SIP info technology should be considered
+ equal. ........ Merged revisions 373532 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions
+ 373501,373505 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373501 | rmudgett | 2012-09-24 17:11:01 -0500
+ (Mon, 24 Sep 2012) | 18 lines Be consistent, send From:
+ "Anonymous" <sip:anonymous@anonymous.invalid> When setting
+ CALLERID(pres)=unavailable in the dialplan, the From header in
+ the SIP message contains "Anonymous"
+ <sip:Anonymous@anonymous.invalid>. For consistency, Asterisk
+ should use a lowercase a in the userpart of the URI. * Make the
+ From header use a lowercase A in the userpart of the anonymous
+ URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola
+ Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383)
+ patch uploaded by Antti Yrjola ........ Merged revisions 373500
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r373505 | mjordan | 2012-09-24 17:17:02 -0500
+ (Mon, 24 Sep 2012) | 19 lines Revert change to res_rtp_asterisk
+ committed in r373236 (1.8) The change committed in r373236
+ attempted to account for endpoints that increased their RTP
+ timestamp in DTMF end of event re-transmissions. This change
+ attempted to make Asterisk continue to work with endpoints that
+ failed to follow the RFC while maintaining the fix that allowed
+ for out of order DTMF to be handled. Unfortunately, there is no
+ free lunch, and this patch broke any system that sent DTMF
+ immediately after an RTP session was established or when an SSRC
+ is updated. As such, that patch is being reverted for the
+ previous behavior. Endpoints that erroneously increase the RTP
+ timestamp in DTMF end of event packets will not work properly
+ with Asterisk. (issue ASTERISK-20424) ........ Merged revisions
+ 373504 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_mixmonitor.c, funcs/func_audiohookinherit.c, /,
+ channels/chan_sip.c: Merged revisions 373466,373468 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373466 | rmudgett | 2012-09-24 15:44:27 -0500
+ (Mon, 24 Sep 2012) | 33 lines Fix potential reentrancy problems
+ in chan_sip. Asterisk v1.8 and later was not as vulnerable to
+ this issue. * Made find_call() lock each private as it processes
+ the found dialogs. (Primary cause of ABE-2876) * Made the other
+ functions that traverse the dialogs container lock each private
+ as it examines them. * Fix race condition in sip_call() if the
+ thread that sent the INVITE is held up long enough for a response
+ to be processed. The p->initid for the INVITE retransmission
+ could be added after it was canceled by the response processing.
+ * Made __sip_destroy() clean up resource pointers after freeing.
+ This is primarily defensive in case someone has a stale private
+ pointer. * Removed redundant memset() in reqprep(). The call to
+ init_req() already does the memset() and is the first reference
+ to req in reqprep(). * Removed useless set of req.method in
+ transmit_invite(). The calls to initreqprep() and reqprep() have
+ to do this because they memset() the req. JIRA ABE-2876
+ .......... Merged -r373423 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 373424 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r373468 | jrose | 2012-09-24 16:05:44 -0500
+ (Mon, 24 Sep 2012) | 13 lines func_audiohookinherit: Document
+ some missed sources. This patch also mentions that
+ AUDIOHOOK_INHERIT can be used to transfer MixMonitor audiohooks.
+ There is also wiki that addresses audiohooks and the use of
+ AUDIOHOOK_INHERIT at the following link:
+ https://wiki.asterisk.org/wiki/display/AST/Audiohooks (closes
+ issue ASTERISK-18220) Reported by: Ishfaq Malik ........ Merged
+ revisions 373467 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-09-24 21:22 +0000 [r373490-373491] Jason Parker <jparker@digium.com>
+
+ * /: reenable automerge
+
+ * /, channels/chan_sip.c: Fix a deadlock caused by a race condition
+ between removing a hint and reloading the dialplan and
+ subscribing to the removed hint. If conditions were right it was
+ possible for both the PBX core and chan_sip to deadlock by both
+ having a lock that the other wants. In the case of the PBX core
+ it had the contexts lock and wanted a SIP dialog lock, while in
+ the case of chan_sip it had the SIP dialog lock and wanted the
+ contexts lock. This fix unlocks the SIP dialog before getting the
+ extension state so that the other thread will not block on trying
+ to lock it. Once the extension state is retrieved the SIP dialog
+ is locked again and life carries on. As the SIP dialog is
+ reference counted it is not possible for it to go away after
+ unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins
+ ........ Merged revisions 373438 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373440 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-24 19:22 +0000 [r373455] Automerge script <automerge@asterisk.org>
+
+ * /: automerge cancel
+
+2012-09-21 19:24 +0000 [r373158-373367] Automerge script <automerge@asterisk.org>
+
+ * channels/iax2-provision.c, /: Merged revisions 373343 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373343 | jrose | 2012-09-21 14:08:58 -0500
+ (Fri, 21 Sep 2012) | 10 lines iax2-provision: Fix improper return
+ on failed cache retrieval (closes issue ASTERISK-20337) reported
+ by: John Covert Patches: iax2-provision.c.patch uploaded by John
+ Covert (license 5512) ........ Merged revisions 373342 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_queue.c, /: Merged revisions 373300 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373300 | jrose | 2012-09-21 10:07:38 -0500
+ (Fri, 21 Sep 2012) | 12 lines app_queue: Make queue reload
+ members and variants of that work Prior to this patch, 'queue
+ reload members' cli command did not work at all. This also
+ affects the manager function 'QueueReload' when supplied with the
+ 'members: yes' field. (closes issue AST-956) Reported by: John
+ Bigelow ........ Merged revisions 373298 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * res/res_rtp_asterisk.c, /, apps/app_meetme.c: Merged revisions
+ 373237,373245 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373237 | mjordan | 2012-09-20 13:42:51 -0500
+ (Thu, 20 Sep 2012) | 18 lines When processing RFC 2833 DTMF,
+ accomodate increasing timestamps in End events While endpoints
+ should not be changing the source timestamp between DTMF event
+ packets, the fact is there exists those endpoints that do exactly
+ that. To work around this, we absorb timestamps within the
+ expected re-transmit period. Note that this period only affects
+ End of Event packets, so it should not prevent the detection of
+ new DTMF digits that happen to arrive right on top of each other.
+ (closes issue ASTERISK-20424) Reported by: Vladimir Mikhelson
+ Tested by: mjordan, Vladimir Mikhelson Review:
+ https://reviewboard.asterisk.org/r/2124 ........ Merged revisions
+ 373236 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r373245 | file | 2012-09-20 14:14:31 -0500 (Thu,
+ 20 Sep 2012) | 15 lines Fix incorrect MeetME conference bridge
+ reference count decrementing and sometimes premature destruction.
+ When using the 'e' or 'E' option to MeetMe the configured
+ conference bridges are loaded and examined to see if any are
+ empty. If no conference bridges are empty the caller is prompted
+ to enter the number of one. This operation left around a pointer
+ to the last created conference bridge still containing
+ participants. When the caller that was not able to find any empty
+ conference bridge hung up this pointer was disposed of and the
+ reference count of the conference bridge decremented. If there
+ was only a single participant in the conference bridge it was
+ ultimately destroyed prematurely. (closes issue AST-994) Reported
+ by: John Bigelow ........ Merged revisions 373242 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/confbridge/conf_config_parser.c, /: Merged revisions 373196
+ via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r373196 | mjordan | 2012-09-19 21:35:13 -0500 (Wed, 19 Sep 2012)
+ | 12 lines Ensure that all ConfBridge sounds can be set using
+ CONFBRIDGE function The CONFBRIDGE function can be used to set
+ the sounds in a ConfBridge bridge profile. Unfortunately, three
+ sounds were missed in the portion of the code that applies the
+ settings passed in from the function: sound_only_one, join, and
+ leave. This patch makes sure that the sounds passed from the
+ function are applied to the bridge profile. (closes issue
+ ASTERISK-20404) Reported by: univ Tested by: mjordan ........
+
+ * /, channels/chan_sip.c: Merged revisions 373179 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373179 | file | 2012-09-19 12:05:47 -0500 (Wed,
+ 19 Sep 2012) | 13 lines Fix a regression where direct media was
+ not permitted for calls using SIP INFO DTMF. A change was
+ committed to fix direct media ACL support. This change wrongly
+ assumed that only a single channel technology structure exists
+ for chan_sip. This is in fact false as a second exists for calls
+ using SIP INFO DTMF. The code which performs direct media ACL
+ checking now checks for both the non-INFO DTMF and INFO DTMF
+ channel technology structures. (closes issue ASTERISK-20409)
+ Reported by: michele cicciotti privatewave ........ Merged
+ revisions 373165 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/manager.c, /: Merged revisions 373132 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373132 | seanbright | 2012-09-18 15:13:21 -0500
+ (Tue, 18 Sep 2012) | 10 lines Don't crash when passing a NULL
+ message to __astman_get_header. Before this commit,
+ __astman_get_header would blindly dereference the passed in
+ 'struct message *' to traverse the header list. There are cases,
+ however, such as '*CLI> sip qualify peer foo' where the message
+ pointer is NULL, so we need to check for that. ........ Merged
+ revisions 373131 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-09-15 00:21 +0000 [r373078-373106] Automerge script <automerge@asterisk.org>
+
+ * channels/sig_ss7.c, /: Merged revisions 373101 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373101 | rmudgett | 2012-09-14 19:20:21 -0500
+ (Fri, 14 Sep 2012) | 20 lines Made companding law for SS7 calls
+ only determined by SS7 signaling type. For SS7, the companding
+ law for a call was chosen inconsistently depending upon ss7type
+ (ITU vs ANSI) and the DAHDI companding default (T1 vs E1). For
+ incoming calls, the companding law was determined by ss7type. For
+ outgoing calls, the companding law was determined by the DAHDI
+ default. With the wrong combination you would get A-law/u-law
+ conflicts. An A-law/u-law conflict sounds like bad static on the
+ line. SS7 ITU signaling with E1 line: ok SS7 ITU signaling with
+ T1 line: noise SS7 ANSI signaling with E1 line: noise SS7 ANSI
+ signaling with T1 line: ok * Fix the companding law used to be
+ determined by the SS7 signaling type only. ........ Merged
+ revisions 373090 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * include/asterisk/astobj2.h, main/ssl.c, main/astobj2.c,
+ main/tcptls.c, /, channels/chan_sip.c: Merged revisions
+ 373059,373062 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373059 | mjordan | 2012-09-14 13:28:40 -0500
+ (Fri, 14 Sep 2012) | 16 lines Constify __ao2_ref_debug in astobj2
+ When REF_DEBUG is enabled in certain files - most notably ccss.c
+ - the 'tag' parameter passed to __ao2_ref_debug will be a const
+ char *. The function currently expects that parameter to not be
+ const. This causes a warning when compiling, as the const
+ qualifier is being discarded. With dev-mode enabled, this
+ prevents compiling Asterisk. This patch makes __ao2_ref_debug's
+ tag and file parameters const. (closes issue ASTERISK-20408)
+ Reported by: mjordan ........ Merged revisions 372959 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r373062 | mjordan | 2012-09-14 14:12:48 -0500
+ (Fri, 14 Sep 2012) | 30 lines Resolve memory leaks in TLS
+ initialization and TLS client connections This patch resolves two
+ sources of memory leaks when using TLS in Asterisk: 1) It removes
+ improper initialization (and multiple re-initializations) of
+ portions of the SSL library. Asterisk calls SSL_library_init and
+ SSL_load_error_strings during SSL initialization; collectively
+ this obviates the need for calling any of the following during
+ initialization or client connection handling: *
+ ERR_load_crypto_strings (handled by SSL_load_error_strings) *
+ OpenSSL_add_all_algorithms (synonym for SSL_library_init) *
+ SSLeay_add_ssl_algorithms (synonym for SSL_library_init) 2)
+ Failure to completely clean up all memory allocated by Asterisk
+ and by the SSL library for TLS clients. This included not freeing
+ the SSL_CTX object in the SIP channel driver, as well as not
+ clearing the error stack when the TLS client exited. Note that
+ these memory leaks were found by Thomas Arimont, and this patch
+ was essentially written by him with some minor tweaks. (closes
+ issue AST-889) Reported by: Thomas Arimont Tested by: Thomas
+ Arimont patches: (bugAST-889.patch) by Thomas Arimont (license
+ 5525) Review: https://reviewboard.asterisk.org/r/2105 ........
+ Merged revisions 373061 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-09-13 19:23 +0000 [r373045] Automerge script <automerge@asterisk.org>
+
+ * include/asterisk/channel.h, main/channel.c, /: Merged revisions
+ 373025 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r373025 | dlee | 2012-09-13 13:44:30 -0500 (Thu,
+ 13 Sep 2012) | 18 lines Fix timeouts for ast_waitfordigit[_full].
+ ast_waitfordigit_full would simply pass its timeout to
+ ast_waitfor_nandfds, expecting it to decrement the timeout by
+ however many milliseconds were waited. This is a problem if it
+ consistently waits less than 1ms. The timeout will never be
+ decremented, and we wait... FOREVER! This patch makes
+ ast_waitfordigit_full manage the timeout itself. It maintains the
+ previously undocumented behavior that negative timeouts wait
+ forever. (closes issue ASTERISK-20375) Reported by: Mark
+ Michelson Tested by: Mark Michelson Review:
+ https://reviewboard.asterisk.org/r/2109/ ........ Merged
+ revisions 373024 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-09-13 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.9.0-digiumphones-rc1 Released.
+
+2012-09-12 15:26 +0000 [r372753-372958] Automerge script <automerge@asterisk.org>
+
+ * /, channels/chan_sip.c: Merged revisions 372933 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372933 | mmichelson | 2012-09-12 09:53:35 -0500
+ (Wed, 12 Sep 2012) | 10 lines Add channel name to a warning to
+ make debugging easier. The "autodestruct with owner in place"
+ message is typically indicative of a channel reference leak.
+ Printing out the name of the channel in the message may be
+ helpful when trying to debug the issue. ........ Merged revisions
+ 372932 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/chan_local.c, /: Merged revisions 372916 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372916 | jrose | 2012-09-11 17:23:20 -0500
+ (Tue, 11 Sep 2012) | 13 lines chan_local: Switch from using a
+ random 4 digit hex identifier to unique id Changes chan_local
+ channels to use an 8 digit hex identifier generated atomically
+ and sequentially in order to eliminate the chance of having
+ multiple channels with the same name during high call volume
+ situations. (issue ASTERISK-20318) Reported by: Dan Cropp Review:
+ https://reviewboard.asterisk.org/r/2104/ ........ Merged
+ revisions 372902 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * include/asterisk/_private.h, main/message.c, main/asterisk.c, /:
+ Merged revisions 372885 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r372885 | mmichelson | 2012-09-11 16:04:36 -0500 (Tue, 11 Sep
+ 2012) | 18 lines Fix inability to shutdown gracefully due to an
+ unending channel reference. message.c makes use of a special
+ message queue channel that exists in thread storage. This channel
+ never goes away due to the fact that the taskprocessor used by
+ message.c does not get shut down, meaning that it never ends the
+ thread that stores the channel. This patch fixes the problem by
+ shutting down the taskprocessor when Asterisk is shut down. In
+ addition, the thread storage has a destructor that will release
+ the channel reference when the taskprocessor is destroyed.
+ (closes issue AST-937) Reported by Jason Parker Patches:
+ AST-937.patch uploaded by Mark Michelson (License #5049) Tested
+ by Jason Parker ........
+
+ * Makefile, /: Merged revisions 372863 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r372863 | dlee | 2012-09-11 12:14:06 -0500 (Tue, 11 Sep 2012) | 4
+ lines Corrects the astsbindir setting when installing the sample
+ asterisk.conf. (closes issue ASTERISK-20406) ........
+
+ * main/features.c, /: Merged revisions 372841 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372841 | mmichelson | 2012-09-11 10:30:37 -0500
+ (Tue, 11 Sep 2012) | 15 lines Fix bad channel application data
+ reference. When channels get bridged due to an AMI bridge action
+ or a DTMF attended transfer, the two channels that get bridged
+ have their application data pointing to the other channel's name.
+ This means that if one channel is hung up but the other moves on,
+ it means that the channel that moves on will have its application
+ data pointing at freed memory. (issue ASTERISK-20335) ........
+ Merged revisions 372840 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/chan_iax2.c, /: Merged revisions 372805 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372805 | kmoore | 2012-09-10 15:56:35 -0500
+ (Mon, 10 Sep 2012) | 13 lines Ensure iax2 debug output is
+ displayed when expected When IAX2 debug was changed from
+ iax_showframe to iax_outputframe, some instances were missed (or
+ added afterward). This was causing debug output to not be
+ displayed when expected. (closes issue ASTERISK-20338)
+ Reported-by: John Covert Patch-by: John Covert ........ Merged
+ revisions 372804 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, apps/app_meetme.c, channels/chan_sip.c: Merged revisions
+ 372764,372767 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372764 | kmoore | 2012-09-10 13:32:51 -0500
+ (Mon, 10 Sep 2012) | 12 lines Warn on CLI when UDPTL init fails
+ This adds a CLI warning when a SDP offer is rejected due to UDPTL
+ initialization failure. Previously, there was no indication of
+ the reason for offer rejection in this case. (closes issue
+ ASTERISK-20357) Reported-by: Francesco Usseglio Gaudi ........
+ Merged revisions 372763 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r372767 | jrose | 2012-09-10 13:41:45 -0500
+ (Mon, 10 Sep 2012) | 8 lines app_meetme: Document that 'p' option
+ will continue in dialplan. (closes issue AST-991) Reported by
+ John Bigelow ........ Merged revisions 372765 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/channel.c, /: Merged revisions 372737 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372737 | jrose | 2012-09-10 12:14:46 -0500
+ (Mon, 10 Sep 2012) | 15 lines Masquerade: Retain parkinglot
+ settings made by CHANNEL function. Prior to this patch, the user
+ would have a parkinglot set on a channel that was parked and when
+ the channel was retrieved, any attempt by that channel to park
+ would simply use the default. This patch makes parkinglot values
+ set in this way be retained through the masquerade. (closes issue
+ AST-990) Reported by: Nick Huskinson Patches:
+ masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose
+ (license 6182) ........ Merged revisions 372736 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-09-09 02:26 +0000 [r372174-372735] Automerge script <automerge@asterisk.org>
+
+ * channels/sip/sdp_crypto.c, /: Merged revisions 372710 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372710 | mjordan | 2012-09-08 20:24:36 -0500
+ (Sat, 08 Sep 2012) | 24 lines Only re-create an SRTP session when
+ needed In r356604, SRTP handling was fixed to accomodate multiple
+ crypto keys in an SDP offer and the ability to re-create an SRTP
+ session when the crypto keys changed. In certain circumstances -
+ most notably when a phone is put on hold after having been
+ bridged for a significant amount of time - the act of re-creating
+ the SRTP session causes problems for certain models of phones.
+ The patch committed in r356604 always re-created the SRTP session
+ regardless of whether or not the cryptographic keys changed.
+ Since this is technically not necessary, this patch modifies the
+ behavior to only re-create the SRTP session if Asterisk detects
+ that the remote key has changed. This allows models of phones
+ that do not handle the SRTP session changing to continue to work,
+ while also providing the behavior needed for those phones that do
+ re-negotiate cryptographic keys. (issue ASTERISK-20194) Reported
+ by: Nicolo Mazzon Tested by: Nicolo Mazzon Review:
+ https://reviewboard.asterisk.org/r/2099 ........ Merged revisions
+ 372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, main/Makefile: Merged revisions 372695 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372695 | dlee | 2012-09-08 00:21:41 -0500 (Sat,
+ 08 Sep 2012) | 10 lines Add OPENSSL_INCLUDE to the CFLAGS for
+ ssl.c and tcptls.c. Without this flag, those files will compile
+ with the system installed OpenSSL headers (if they exist). This
+ is a real bummer if a different path was specified using
+ --with-ssl= (closes issue ASTERISK-20392) ........ Merged
+ revisions 372682 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/astmm.c, /: Merged revisions 372656 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372656 | rmudgett | 2012-09-07 18:06:38 -0500
+ (Fri, 07 Sep 2012) | 8 lines Fix MALLOC_DEBUG version of
+ ast_strndup(). (closes issue ASTERISK-20349) Reported by: Brent
+ Eagles ........ Merged revisions 372655 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * funcs/func_math.c, apps/app_queue.c, apps/app_voicemail.c, /:
+ Merged revisions 372621,372625,372629 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372621 | rmudgett | 2012-09-07 16:24:39 -0500
+ (Fri, 07 Sep 2012) | 18 lines Fix VoicemailUserEntry event
+ headers ServerEmail and MailCommand reported values. The AMI
+ action VoicemailUsersList VoicemailUserEntry event headers
+ ServerEmail and MailCommand did not report the global values if
+ they were not overridden. The VoicemailUserEntry event header
+ ServerEmail was not populated with the global value if the
+ voicemail user did not override it. The VoicemailUserEntry event
+ header MailCommand was never populated with a value. * Removed
+ unused struct ast_vm_user member mailcmd[]. (closes issue
+ AST-973) Reported by: John Bigelow Tested by: rmudgett ........
+ Merged revisions 372620 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r372625 | rmudgett | 2012-09-07 16:49:16 -0500
+ (Fri, 07 Sep 2012) | 10 lines Fix exception path typo in
+ app_queue.c try_calling(). (closes issue ASTERISK-20380) Reported
+ by: Jeremy Pepper Patches: fix-local-channel-locking.patch
+ (license #6350) patch uploaded by Jeremy Pepper ........ Merged
+ revisions 372624 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r372629 | rmudgett | 2012-09-07 17:07:59 -0500
+ (Fri, 07 Sep 2012) | 8 lines Remove annoying unconditional debug
+ message from INC/DEC functions. (closes issue AST-1001) Reported
+ by: Guenther Kelleter ........ Merged revisions 372628 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_minivm.c, /: Merged revisions 372582 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372582 | mjordan | 2012-09-06 21:25:36 -0500
+ (Thu, 06 Sep 2012) | 13 lines Free ast_str objects when temp file
+ fails to be created in MiniVM The previous commit (r372554) was
+ from a patch that was written before r366880, which ensured that
+ ast_str objects allocated in the sendmail routine were free'd in
+ off nominal paths. This commit frees the string objects in the
+ off nominal path introduced in r372554. (issue ASTERISK-17133)
+ Reported by: Tzafrir Cohen ........ Merged revisions 372581 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_minivm.c, /: Merged revisions 372555 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372555 | mjordan | 2012-09-06 21:11:46 -0500
+ (Thu, 06 Sep 2012) | 22 lines Fix file descriptor leak and
+ pointer scope issue in MiniVM when sending mail When MiniVM sends
+ an e-mail and it has the volgain option set, it will spawn sox in
+ a separate process to handle the manipulation of the sound file.
+ In doing so, it creates a temporary file. There are two problems
+ here: 1) The file descriptor returned from mkstemp is leaked 2)
+ The finalfilename character pointer points to a buffer that loses
+ scope once volgain processing is finished. Note that in r316265,
+ Russell fixed some gcc warnings by using the return value of the
+ mkstemp call. A warning was placed in minivm that the file
+ descriptor was going to be leaked. This patch reverts that
+ change, as it handles the leak and 'uses' the file descriptor
+ returned from mkstemp. (closes issue ASTERISK-17133) Reported by:
+ Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir
+ Cohen (license #5035) ........ Merged revisions 372554 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_queue.c, channels/sig_pri.c, /: Merged revisions
+ 372518,372522 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372518 | kmoore | 2012-09-06 16:40:50 -0500
+ (Thu, 06 Sep 2012) | 14 lines Ensure listed queues are not
+ offered for completion When using tab-completion for the list of
+ queues on "queue reset stats" or "queue reload
+ {all|members|parameters|rules}", the tab-completion listing for
+ further queues erroneously listed queues that had already been
+ added to the list. The tab-completion listing now only displays
+ queues that are not already in the list. (closes issue AST-963)
+ Reported-by: John Bigelow ........ Merged revisions 372517 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r372522 | rmudgett | 2012-09-06 17:10:04 -0500
+ (Thu, 06 Sep 2012) | 22 lines Fix loss of MOH on an ISDN channel
+ when parking a call for the second time. Using the AMI redirect
+ action to take an ISDN call out of a parking lot causes the MOH
+ state to get confused. The redirect action does not take the call
+ off of hold. When the call is subsequently parked again, the call
+ no longer hears MOH. * Make chan_dahdi/sig_pri restart MOH on
+ repeated AST_CONTROL_HOLD frames if it is already in a state
+ where it is supposed to be sending MOH. The MOH may have been
+ stopped by other means. (Such as killing the generator.) This
+ simple fix is done rather than making the AMI redirect action
+ post an AST_CONTROL_UNHOLD unconditionally when it redirects a
+ channel and thus potentially breaking something with an
+ unexpected AST_CONTROL_UNHOLD. (closes issue ABE-2873) Patches:
+ jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by
+ rmudgett ........ Merged revisions 372521 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ................
+
+ * configs/res_ldap.conf.sample, /, channels/chan_sip.c: Merged
+ revisions 372499 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r372499 | dsessions | 2012-09-06 13:54:54 -0500 (Thu, 06 Sep
+ 2012) | 16 lines LDAP Realtime Peers Cannot Register Prior to
+ 1.8, it was not necessary for an explicit "type" to be set for an
+ asterisk LDAP realtime peer. Now the routine find_peer actually
+ checks the type field during registration and fails to find the
+ peer if it is not set. The attached patches make the realtime
+ type equal whatever type is being searched for if the type is 0
+ upon return from routine build_peer. (closes issue
+ ASTERISK-17222) Reported by: John Covert Patch by: David Vossel
+ Tested by: Darren Sessions Review:
+ https://reviewboard.asterisk.org/r/2095/ ........
+
+ * UPGRADE-1.8.txt, /: Merged revisions 372472 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372472 | jrose | 2012-09-06 10:54:38 -0500
+ (Thu, 06 Sep 2012) | 15 lines chan_sip: Note change in behavior
+ to how directmediapermit/deny ACL works r366547 introduced a
+ change to the directmedia ACL for chan_sip which modified the
+ behavior significantly. Prior to the patch, this option would
+ bridge peers with directmedia if a peer's IP address matched its
+ own directmedia ACL. After that patch, the peer would check the
+ bridged peer's ACL instead. This change has been present since
+ 1.8.14.0. That patched failed to document the change in
+ Upgrade.txt, so this patch adds mention of that change to
+ UPGRADE.txt (UPGRADE-1.8.txt in newer branches) (issue AST-876)
+ ........ Merged revisions 372471 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_queue.c, /: Merged revisions 372445 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372445 | kmoore | 2012-09-06 09:29:35 -0500
+ (Thu, 06 Sep 2012) | 14 lines Ensure "rules" is tab-completable
+ for "queue show" Previously, tabbing at the end of "queue show"
+ produced a list of available queues about which information could
+ be shown, but did not include an alternative command, "rules", to
+ access information about queue rules. The "rules" item should now
+ be shown in the list of tab-completable items. (closes issue
+ AST-958) Reported-by: John Bigelow ........ Merged revisions
+ 372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * pbx/pbx_dundi.c, /: Merged revisions 372418 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372418 | mjordan | 2012-09-05 21:49:41 -0500
+ (Wed, 05 Sep 2012) | 25 lines Fix DUNDi message routing bug when
+ neighboring peer is unreachable Consider a scenario where DUNDi
+ peer PBX1 has two peers that are its neighbors, PBX2 and PBX3,
+ and where PBX2 and PBX3 are also neighbors. If the connection is
+ temporarily broken between PBX1 and PBX3, PBX1 should not include
+ PBX3 in the list of peers it sends to PBX2 in a DPDISCOVER
+ message, as it cannot send messages to PBX3. If it does, PBX2
+ will assume that PBX3 already received the message and fail to
+ forward the message on to PBX3 itself. This patch fixes this by
+ only including peers in a DPDISCOVER message that are reachable
+ by the sending node. This includes all peers with an empty
+ address (00:00:00:00:00:00) and that are have been reached by a
+ qualify message. This patch also prevents attempting to qualify a
+ dynamic peer with an empty address until that peer registers.
+ (closes issue ASTERISK-19309) Reported by: Peter Racz patches:
+ dundi_routing.patch uploaded by Peter Racz (license 6290) The
+ patch uploaded by Peter was modified slightly for this commit.
+ ........ Merged revisions 372417 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_followme.c, /: Merged revisions 372391 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372391 | mjordan | 2012-09-05 19:56:47 -0500
+ (Wed, 05 Sep 2012) | 24 lines Allow configured numbers for
+ FollowMe to be greater than 90 characters When parsing a 'number'
+ defined in followme.conf, FollowMe previously parsed the number
+ in the configuration file into a buffer with a length of 90
+ characters. This can artificially limit some parallel dial
+ scenarios. This patch allows for numbers of any length to be
+ defined in the configuration file. Note that Clod Patry
+ originally wrote a patch to fix this problem and received a Ship
+ It! on the JIRA issue. The patch originally expanded the buffer
+ to 256 characters. Instead, the patch being committed duplicates
+ the string in the config file on the stack before parsing it for
+ consumption by the application. (closes issue ASTERISK-16879)
+ Reported by: Clod Patry Tested by: mjordan patches:
+ followme_no_limit.diff uploaded by Clod Patry (license #5138)
+ Slightly modified for this commit. ........ Merged revisions
+ 372390 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/dsp.c, /: Merged revisions 372372 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r372372 | rmudgett | 2012-09-05 14:42:17 -0500 (Wed, 05 Sep 2012)
+ | 1 line Fix compile error. ........
+
+ * main/dsp.c, main/pbx.c, main/manager.c, /: Merged revisions
+ 372338,372341,372358 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372338 | kmoore | 2012-09-05 13:30:49 -0500
+ (Wed, 05 Sep 2012) | 13 lines Ensure counts generated in
+ manager_show_dialplan_helper are correct When
+ manager_show_dialplan_helper was written, the counter increment
+ for the total number of contexts was placed with the extensions
+ increment instead of in the enclosing loop. This function should
+ now generate correct context counts. (closes issue AST-970)
+ Reported-by: John Bigelow ........ Merged revisions 372337 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r372341 | alecdavis | 2012-09-05 13:43:12 -0500
+ (Wed, 05 Sep 2012) | 7 lines dsp.c: in ast_mf_detect_init
+ incorrectly sets goertzel samples to 160, should be MF_GSIZE
+ Related https://reviewboard.asterisk.org/r/2097/ ........ Merged
+ revisions 372339 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r372358 | kmoore | 2012-09-05 14:22:08 -0500
+ (Wed, 05 Sep 2012) | 13 lines Correct documentation for
+ ModuleLoad AMI action The documentation incorrectly listed 'rtp'
+ as a reloadable subsystem and left out many other reloadable
+ subsystems. It is now also documented that subsystems may only be
+ reloaded, not loaded or unloaded. (closes issue AST-977)
+ Reported-by: John Bigelow ........ Merged revisions 372354 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 372288 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372288 | mjordan | 2012-09-05 08:42:54 -0500
+ (Wed, 05 Sep 2012) | 27 lines Fix memory leaks in app_voicemail
+ when using IMAP storage or realtime config This patch fixes two
+ memory leaks: 1. When find_user is called with NULL as its first
+ parameter, the voicemail user returned is allocated on the heap.
+ The inboxcount2 function uses find_user in such a fashion when
+ counting new messages, and fails to free the resulting voicemail
+ user object. 2. When populate_defaults is called on a voicemail
+ user, it wipes whatever flags have been set on the object by
+ copying over the global flags object. If the VM_ALLOCED flag was
+ ste on the voicemail user prior to doing so, that flag is
+ removed. This leaks the voicemail user when free_user is later
+ called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek
+ patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277)
+ Patch slightly modified for this commit. Review:
+ https://reviewboard.asterisk.org/r/2096 ........ Merged revisions
+ 372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/dsp.c, /: Merged revisions 372240 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372240 | alecdavis | 2012-09-05 02:37:42 -0500
+ (Wed, 05 Sep 2012) | 21 lines dsp.c: Fix multiple issues when
+ no-interdigit delay is present, and fast DTMF 50ms/50ms Revert
+ DTMF hit/miss detector to original -r349249 method with some
+ changes, remove unnecessary; 1. reseting of hits=0, when no
+ signal, only need to set it once. 2. incrementing of hits, when
+ the hit is the same as the current hit. 3. setting of lasthit,
+ when it's the same as before. Change HITS_TO_BEGIN to 2,
+ MISSES_TO_END to 3 & 3 spelling mistakes (closes issue
+ ASTERISK-19610) alecdavis (license 585) Reported by:
+ Jean-Philippe Lord Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/2085/ ........ Merged
+ revisions 372239 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/dsp.c, /: Merged revisions 372213 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372213 | alecdavis | 2012-09-05 01:47:54 -0500
+ (Wed, 05 Sep 2012) | 13 lines dsp.c: optimize goerztzel sample
+ loops, in dtmf_detect, mf_detect and tone_detect use a temporary
+ short int when repeatedly used to call goertzel_sample. alecdavis
+ (license 585) Reported by: alecdavis Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/2093/ ........ Merged
+ revisions 372212 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * res/res_rtp_asterisk.c, /: Merged revisions 372198 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372198 | elguero | 2012-09-04 23:47:00 -0500
+ (Tue, 04 Sep 2012) | 19 lines Fix Incrementing Sequence Number
+ For Retransmitted DTMF End Packets In Asterisk 1.4+, a fix was
+ put in place to increment the sequence number for retransmitted
+ DTMF end packets. With the introduction of the RTP engine API in
+ 1.8, the sequence number was no longer being incremented. This
+ patch fixes this regression as well as cleans up a few lines that
+ were not doing anything. (closes issue ASTERISK-20295) Reported
+ by: Nitesh Bansal Tested by: Michael L. Young Patches:
+ 01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license
+ 6418) asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2083/ ........ Merged
+ revisions 372185 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * cel/cel_pgsql.c, /: Merged revisions 372165 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372165 | mjordan | 2012-09-04 21:19:25 -0500
+ (Tue, 04 Sep 2012) | 18 lines Fix memory leak when CEL is
+ successfully written to PostgreSQL database PQClear is not called
+ when the result object of a call to PQExec has a status of
+ PGRES_COMMAND_OK. Interestingly enough, the off nominal case was
+ handled properly, so this memory leak only occurred when CEL
+ records were successfully written. This patch properly clears the
+ result in the nominal code path. (closes issue ASTERISK-19991)
+ Reported by: Etienne Lessard Tested by: Etienne Lessard patches:
+ mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license
+ #6394) ........ Merged revisions 372158 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-08-30 21:24 +0000 [r371850-372115] Automerge script <automerge@asterisk.org>
+
+ * apps/app_queue.c, /: Merged revisions 372090 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372090 | mmichelson | 2012-08-30 15:53:09 -0500
+ (Thu, 30 Aug 2012) | 17 lines Prevent crash on shutdown due to
+ refcount error on queues container. When app_queue is unloaded,
+ the queues container has its refcount decremented, potentially to
+ 0. Then the taskprocessor responsible for handling device state
+ changes is unreferenced. If the taskprocessor happens to be just
+ about to run its task, then it will create and destroy an
+ iterator on the queues container. This can cause the refcount on
+ the queues container to increase to 1 and then back to 0. Going
+ back to 0 a second time results in double frees. This failure was
+ seen periodically in the testsuite when Asterisk would shut down.
+ ........ Merged revisions 372089 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_queue.c, /: Merged revisions 372049 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r372049 | mmichelson | 2012-08-30 13:33:37 -0500
+ (Thu, 30 Aug 2012) | 16 lines Help prevent ringing queue members
+ from being rung when ringinuse set to no. Queue member status
+ would not always get updated properly when the member was called,
+ thus resulting in the member getting multiple calls. With this
+ change, we update the member's status at the time of calling, and
+ we also check to make sure the member is still available to take
+ the call before placing an outbound call. (closes issue
+ ASTERISK-16115) reported by nik600 Patches:
+ app_queue.c-svn-r370418.patch uploaded by Italo Rossi (license
+ #6409) ........ Merged revisions 372048 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/chan_iax2.c, main/manager.c, /,
+ README-SERIOUSLY.bestpractices.txt: Merged revisions
+ 371999,372020 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371999 | mjordan | 2012-08-30 11:06:47 -0500
+ (Thu, 30 Aug 2012) | 36 lines AST-2012-012: Resolve AMI User
+ Unauthorized Shell Access through ExternalIVR The AMI Originate
+ action can allow a remote user to specify information that can be
+ used to execute shell commands on the system hosting Asterisk.
+ This can result in an unwanted escalation of permissions, as the
+ Originate action, which requires the "originate" class
+ authorization, can be used to perform actions that would
+ typically require the "system" class authorization. Previous
+ attempts to prevent this permission escalation (AST-2011-006,
+ AST-2012-004) have sought to do so by inspecting the names of
+ applications and functions passed in with the Originate action
+ and, if those applications/functions matched a predefined set of
+ values, rejecting the command if the user lacked the "system"
+ class authorization. As reported by IBM X-Force Research, the
+ "ExternalIVR" application is not listed in the predefined set of
+ values. The solution for this particular vulnerability is to
+ include the "ExternalIVR" application in the set of defined
+ applications/functions that require "system" class authorization.
+ Unfortunately, the approach of inspecting fields in the Originate
+ action against known applications/functions has a significant
+ flaw. The predefined set of values can be bypassed by creative
+ use of the Originate action or by certain dialplan
+ configurations, which is beyond the ability of Asterisk to
+ analyze at run-time. Attempting to work around these scenarios
+ would result in severely restricting the applications or
+ functions and prevent their usage for legitimate means. As such,
+ any additional security vulnerabilities, where an
+ application/function that would normally require the "system"
+ class authorization can be executed by users with the "originate"
+ class authorization, will not be addressed. Instead, the
+ README-SERIOUSLY.bestpractices.txt file has been updated to
+ reflect that the AMI Originate action can result in commands
+ requiring the "system" class authorization to be executed. Proper
+ system configuration can limit the impact of such scenarios.
+ (closes issue ASTERISK-20132) Reported by: Zubair Ashraf of IBM
+ X-Force Research ........ Merged revisions 371998 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r372020 | mjordan | 2012-08-30 11:22:54 -0500
+ (Thu, 30 Aug 2012) | 17 lines AST-2012-013: Resolve ACL rules
+ being ignored during calls by some IAX2 peers When an IAX2 call
+ is made using the credentials of a peer defined in a dynamic
+ Asterisk Realtime Architecture (ARA) backend, the ACL rules for
+ that peer are not applied to the call attempt. This allows for a
+ remote attacker who is aware of a peer's credentials to bypass
+ the ACL rules set for that peer. This patch ensures that the ACLs
+ are applied for all peers, regardless of their storage mechanism.
+ (closes issue ASTERISK-20186) Reported by: Alan Frisch Tested by:
+ mjordan, Alan Frisch ........ Merged revisions 372015 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * doc/CODING-GUIDELINES (added), /: Merged revisions 371962 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371962 | mjordan | 2012-08-30 07:48:07 -0500
+ (Thu, 30 Aug 2012) | 17 lines Restore CODING-GUIDELINES to doc
+ folder In r294740, the CODING-GUIDELINES was removed from the doc
+ folder in favor of the content on the Asterisk wiki. Some folks
+ still look in the doc folder initially for coding guideline
+ suggestions; as such, this patch adds a CODING-GUIDELINES file
+ back into the doc folder. The content of the file merely points
+ to the correct page on the Asterisk wiki where the coding
+ guidelines currently live. (closes issue ASTERISK-20279) Reported
+ by: Andrew Latham Patches: CODING-GUIDELINES.diff uploaded by
+ Andrew Latham (license 5985) ........ Merged revisions 371961
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, apps/app_meetme.c: Merged revisions 371920 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371920 | jrose | 2012-08-29 15:58:21 -0500
+ (Wed, 29 Aug 2012) | 5 lines app_meetme: Adding test events for
+ following activity in MeetMe. ........ Merged revisions 371919
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/channel.c, /: Merged revisions 371890 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371890 | rmudgett | 2012-08-29 14:40:20 -0500
+ (Wed, 29 Aug 2012) | 13 lines Initialize file descriptors for
+ dummy channels to -1. Dummy channels usually aren't read from,
+ but functions like SHELL and CURL use autoservice on the channel.
+ (closes issue ASTERISK-20283) Reported by: Gareth Palmer Patches:
+ svn-371580.patch (license #5169) patch uploaded by Gareth Palmer
+ (modified) ........ Merged revisions 371888 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_dial.c, /: Merged revisions 371861 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371861 | rmudgett | 2012-08-29 13:24:54 -0500
+ (Wed, 29 Aug 2012) | 15 lines Fix hangup cause passthrough
+ regression. The v1.8 -r369258 change to fix the F and F(x) action
+ logic introduced a regression in passing the hangup cause from
+ the called channel to the caller channel. (closes issue
+ ASTERISK-20287) Reported by: Konstantin Suvorov Patches:
+ app_dial_hangupcause.patch (license #6421) patch uploaded by
+ Konstantin Suvorov (modified) Tested by: rmudgett ........ Merged
+ revisions 371860 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 371825 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371825 | jrose | 2012-08-29 12:07:35 -0500
+ (Wed, 29 Aug 2012) | 8 lines chan_sip: Send 408 on retransmit
+ timeout instead of 603 (closes issue ASTERISK-20124) Reported by:
+ Walter Doekes ........ Merged revisions 371824 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-08-27 22:25 +0000 [r371689-371816] Automerge script <automerge@asterisk.org>
+
+ * configs/agents.conf.sample, main/manager.c, /: Merged revisions
+ 371783,371789 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371783 | mmichelson | 2012-08-27 16:29:29 -0500
+ (Mon, 27 Aug 2012) | 9 lines Fix incorrect documentation of the
+ MailboxStatus manager command. The "Waiting" field was
+ misdocumented as reporting the number of messages waiting. In
+ reality, it simply indicated the presence or absence of waiting
+ messages. ........ Merged revisions 371782 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r371789 | mmichelson | 2012-08-27 16:49:51 -0500
+ (Mon, 27 Aug 2012) | 13 lines Fix misleading documentation in
+ agents.conf.sample regarding ackcall usage. The documentation
+ made it sound as if the DTMF acknowledgment was needed at the
+ time the agent logs in, rather than when the agent is called.
+ This is likely a relic from the days when there were multiple
+ ways of logging in agents. (closes issue AST-962) reported by
+ Steve Pitts ........ Merged revisions 371787 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * configs/queues.conf.sample, /: Merged revisions 371748 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371748 | mmichelson | 2012-08-27 12:36:43 -0500
+ (Mon, 27 Aug 2012) | 10 lines Fix incorrectly documented option
+ in queues.conf sharedlastcall defaults to "no" not "yes" (closes
+ issue AST-979) reported by Steve Pitts ........ Merged revisions
+ 371747 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/lock.c, /: Merged revisions 371719 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371719 | dlee | 2012-08-27 11:43:09 -0500 (Mon,
+ 27 Aug 2012) | 15 lines Fixes ast_rwlock_timed[rd|wr]lock for BSD
+ and variants. The original implementations simply wrap pthread
+ functions, which take absolute time as an argument. The spinlock
+ version for systems without those functions treated the argument
+ as a delta. This patch fixes the spinlock version to be
+ consistent with the pthread version. (closes issue
+ ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch
+ uploaded by Egor Gorlin (license 6416) ........ Merged revisions
+ 371718 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/utils.c, /: Merged revisions 371691 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371691 | kmoore | 2012-08-27 08:57:10 -0500
+ (Mon, 27 Aug 2012) | 14 lines Implement workaround for
+ BETTER_BACKTRACES crash When compiling with BETTER_BACKTRACES
+ enabled, Asterisk will sometimes crash when "core show locks" is
+ run. This happens regularly in the testsuite since several tests
+ run "core show locks" to help with debugging. This seems to be a
+ fault with libraries on certain operating systems (notably CentOS
+ 6.2/6.3) running on virtual machines and utilizing gcc 4.4.6.
+ (closes issue ASTERISK-20090) ........ Merged revisions 371690
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/dsp.c, /: Merged revisions 371663 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371663 | alecdavis | 2012-08-26 18:06:14 -0500
+ (Sun, 26 Aug 2012) | 5 lines mf_detect: incorrectly used
+ DTMF_GSIZE instead of MF_GSIZE ........ Merged revisions 371662
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-08-21 21:28 +0000 [r371617] Automerge script <automerge@asterisk.org>
+
+ * main/file.c, main/utils.c, apps/app_queue.c, pbx/pbx_config.c,
+ res/res_jabber.c, apps/app_stack.c, channels/chan_oss.c,
+ res/res_config_sqlite.c, cdr/cdr_tds.c, main/xmldoc.c,
+ apps/app_dial.c, channels/chan_dahdi.c, /, channels/chan_sip.c,
+ funcs/func_odbc.c: Merged revisions 371591 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371591 | mmichelson | 2012-08-21 15:40:18 -0500
+ (Tue, 21 Aug 2012) | 22 lines Fix misuses of asprintf throughout
+ the code. This fixes three main issues * Change asprintf() uses
+ to ast_asprintf() so that it pairs properly with ast_free() and
+ no longer causes MALLOC_DEBUG to freak out. * When ast_asprintf()
+ fails, set the pointer NULL if it will be referenced later. * Fix
+ some memory leaks that were spotted while taking care of the
+ first two points. (Closes issue ASTERISK-20135) reported by
+ Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071
+ ........ Merged revisions 371590 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-08-20 16:25 +0000 [r371570] Automerge script <automerge@asterisk.org>
+
+ * main/udptl.c, /: Merged revisions 371545 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371545 | kmoore | 2012-08-20 10:27:15 -0500
+ (Mon, 20 Aug 2012) | 15 lines Ignore recovered zero-length
+ secondary UDPTL packets In some cases, recovering lost packets
+ using the secondary packet recovery mechanism with UDPTL/T.38 can
+ result in the recovery of zero-length packets. These must be
+ ignored or the frame generated from them can cause segfaults and
+ allocation failures. (closes issue ASTERISK-19762) (closes issue
+ ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob
+ Gagnon (rgagnon) ........ Merged revisions 371544 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-08-18 03:19 +0000 [r371254-371534] Automerge script <automerge@asterisk.org>
+
+ * main/http.c, /: Merged revisions 371529 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r371529 | mjordan | 2012-08-17 21:34:10 -0500 (Fri, 17 Aug 2012)
+ | 7 lines Remove old debug code from http configuration loading
+ (closes issue ASTERISK-20254) Reported by: Andrew Latham Patches:
+ http.diff uploaded by Andrew Latham (license #5985) ........
+
+ * main/xmldoc.c, /: Merged revisions 371491 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371491 | mjordan | 2012-08-17 15:21:30 -0500
+ (Fri, 17 Aug 2012) | 17 lines Fix memory leak in XML
+ documentation When formatting documentation fields, the XML
+ documentation parser calls xmldoc_get_formatted. This function
+ allocates a string buffer at the beginning of its routine.
+ Unfortunately, on certain code paths, it also calls
+ xmldoc_string_cleanup, which assumes that it will create the
+ string buffer. The previously allocated string buffer is then
+ leaked by the xmldoc_string_cleanup routine. Now: we don't do
+ that. (closes issue AST-932) Reported by: Alexander Homig
+ ........ Merged revisions 371469 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/loader.c, /: Merged revisions 371437 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371437 | kmoore | 2012-08-17 10:51:06 -0500
+ (Fri, 17 Aug 2012) | 11 lines Add instrumentation to subsystem
+ reloads When Asterisk is built with TEST_FRAMEWORK defined,
+ Asterisk will now generate TestEvent AMI events on subsystem
+ reloads such as cdr, dnsmgr, extconfig, etc. (issue PQ-1126)
+ ........ Merged revisions 371436 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/config.c, main/loader.c, /: Merged revisions 371394,371398
+ via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371394 | kmoore | 2012-08-16 17:42:53 -0500
+ (Thu, 16 Aug 2012) | 11 lines Add module reload instrumentation
+ for TEST_FRAMEWORK This adds AMI events for module reloads when
+ Asterisk is built with TEST_FRAMEWORK enabled and corrects
+ generation of the module load AMI event. (issue PQ-1126) ........
+ Merged revisions 371393 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r371398 | twilson | 2012-08-16 17:50:12 -0500
+ (Thu, 16 Aug 2012) | 13 lines Handle integer over/under-flow in
+ ast_parse_args The strtol family of functions will return
+ *_MIN/*_MAX on overflow. To detect when an overflow has happened,
+ errno must be set to 0 before calling the function, then checked
+ afterward. (closes issue ASTERISK-20120) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2073/ ........ Merged
+ revisions 371392 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 371358 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371358 | jrose | 2012-08-16 14:05:21 -0500
+ (Thu, 16 Aug 2012) | 11 lines chan_sip: Use pvt outgoing_call
+ variable to set Remote-Party-ID Header Previously the pvt
+ SIP_OUTGOING flag was used instead, which will frequently flip
+ during reinvites. (closes issue AST-897) Reported by: Thomas
+ Arimont ........ Merged revisions 371357 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 371338 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371338 | jrose | 2012-08-16 11:16:04 -0500
+ (Thu, 16 Aug 2012) | 14 lines chan_sip: Trigger reinvite if the
+ SDP answer is included in the SIP ACK Under certain conditions, a
+ SIP transaction involving directmedia wouldn't trigger a
+ re-invite because the SDP answer was included in an ACK instead
+ of in a message that we would have triggered the invite with.
+ This patch just queues a source change control frame if the
+ dialog is using directmedia when we find sdp for an ACK. (closes
+ issue AST-913) Reported by: Thomas Arimont ........ Merged
+ revisions 371337 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_queue.c, /: Merged revisions 371313 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371313 | mmichelson | 2012-08-15 18:19:09 -0500
+ (Wed, 15 Aug 2012) | 20 lines Fix bug where final queue member
+ would not be removed from memory. If a static queue had realtime
+ members, then there could be a potential for those realtime
+ members not to be properly deleted from memory. If the queue's
+ members were loaded from realtime and then all the members were
+ deleted from the backend, then the queue would still think these
+ members existed. The reason was that there was a short- circuit
+ in code such that if there were no members found in the backend,
+ then the queue would not be updated to reflect this. Note that
+ this only affected static queues with realtime members. Realtime
+ queues with realtime members were unaffected by this issue.
+ (closes issue ASTERISK-19793) reported by Marcus Haas ........
+ Merged revisions 371306 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 371271 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371271 | kmoore | 2012-08-15 15:15:08 -0500
+ (Wed, 15 Aug 2012) | 12 lines Avoid unconditional NULLing of
+ mwipvt on relatedpeer on SIP dialog destruction The other
+ instance of this bug was fixed by jcolp/file in r121496. If we
+ are destroying a dialog only set the MWI dialog pointer on the
+ related peer to NULL if it is the dialog currently being
+ destroyed. (closes issue ASTERISK-20119) Patch-by: Misha
+ Vodsedalek ........ Merged revisions 371270 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /: Merged revisions 371250-371251 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r371250 | elguero | 2012-08-14 20:35:57 -0500 (Tue, 14 Aug 2012)
+ | 17 lines Fix Segfault When Registering SIP Over WebSockets The
+ helper function, get_address_family_filter, in chan_sip for dns
+ resolution by address family was not recognizing the websockets
+ transport and resulting in a null pointer being sent to functions
+ in netsock2, in an attempt to determine if we are bound to ANY
+ address ([::]) or not. This patch fixes this issue by handling
+ the transport types SIP_TRANSPORT_WS and SIP_TRANSPORT_WSS which
+ results in a sock address being set properly for use in
+ determining the address family. (closes issue ASTERISK-20221)
+ Reported by: Sven Beisiegel Tested by: Sven Beisiegel, James
+ Mortensen Patches: asterisk-20221-ws-family-filter.diff uploaded
+ by Michael L. Young (license 5026) ........ r371251 | elguero |
+ 2012-08-14 20:43:23 -0500 (Tue, 14 Aug 2012) | 4 lines Reverting
+ this change that was meant for branch 11. (issue ASTERISK-20221)
+ ........
+
+2012-08-13 20:25 +0000 [r371226] Automerge script <automerge@asterisk.org>
+
+ * main/loader.c, /, apps/app_meetme.c, channels/chan_sip.c: Merged
+ revisions 371199,371203 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371199 | mmichelson | 2012-08-13 14:51:19 -0500
+ (Mon, 13 Aug 2012) | 5 lines Fix problem where incorrect pointer
+ was checked for nullity. ........ Merged revisions 371198 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r371203 | kmoore | 2012-08-13 15:04:15 -0500
+ (Mon, 13 Aug 2012) | 13 lines Add test instrumentation This adds
+ test instrumentation for loading and unloading of modules and for
+ certain actions in MeetMe to be used in the testsuite or any
+ other consumer of AMI events. These will only be generated when
+ Asterisk is built with TEST_FRAMEWORK enabled. (issue PQ-1131)
+ (issue PQ-1133) ........ Merged revisions 371201 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-08-10 22:23 +0000 [r370922-371168] Automerge script <automerge@asterisk.org>
+
+ * apps/app_queue.c, /: Merged revisions 371142 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371142 | mmichelson | 2012-08-10 16:23:52 -0500
+ (Fri, 10 Aug 2012) | 18 lines Fix a couple of documentation
+ problems in app_queue.c * The RemoveQueueMember app made mention
+ of options that could be passed in, but no options are supported.
+ I have removed the listing of options from the documentation. *
+ The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible
+ value that could be set. (closes issue AST-949) reported by Steve
+ Pitts (closes issue AST-954) reported by Steve Pitts ........
+ Merged revisions 371141 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * addons/chan_ooh323.c, /: Merged revisions 371090 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371090 | may | 2012-08-10 11:46:38 -0500 (Fri,
+ 10 Aug 2012) | 12 lines remove ALREADYGONE flag on ooh323 call
+ data by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone
+ there really. This indication arrive from asterisk core not h.323
+ stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov
+ Patches: ASTERISK-19308.patch ........ Merged revisions 371089
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * addons/ooh323c/src/ooGkClient.c, /: Merged revisions 371061 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371061 | may | 2012-08-10 10:13:10 -0500 (Fri,
+ 10 Aug 2012) | 10 lines Send re-register packets by GRQ
+ (gatekeeper request) interval (close issue ASTERISK-20094)
+ Patches: ASTERISK-20094-2.patch ........ Merged revisions 371060
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * configure.ac, channels/sig_pri.c, channels/sig_ss7.c,
+ addons/ooh323c/src/ooGkClient.c, channels/chan_dahdi.c, /,
+ configure, include/asterisk/autoconfig.h.in: Merged revisions
+ 371013,371022 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r371013 | rmudgett | 2012-08-09 14:11:01 -0500
+ (Thu, 09 Aug 2012) | 5 lines Use better libss7 detection test and
+ move libpri compile test. ........ Merged revisions 371012 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r371022 | may | 2012-08-09 14:20:09 -0500 (Thu,
+ 09 Aug 2012) | 10 lines Fix to resend GRQ/RRQ if RRJ
+ (registration reject) is received (close issue ASTERISK-20094)
+ Patches: ASTERISK-20094.patch ........ Merged revisions 371011
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * addons/ooh323c/src/ooh323ep.c, /, apps/app_meetme.c: Merged
+ revisions 370986,370989 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370986 | kmoore | 2012-08-09 12:39:52 -0500
+ (Thu, 09 Aug 2012) | 11 lines Correct documentation for the
+ MeetMe x flag The documentation for the x flag for MeetMe
+ incorrectly described its function as closing down the conference
+ when the last marked user left. It actually causes the users with
+ that flag to leave the conference when the last marked user
+ exits. The functionality of this flag is not changing. ........
+ Merged revisions 370985 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r370989 | may | 2012-08-09 13:05:34 -0500 (Thu,
+ 09 Aug 2012) | 5 lines change opening h323 logfile with append
+ mode instead of overwrite ........ Merged revisions 370988 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_chanspy.c, /: Merged revisions 370954 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370954 | elguero | 2012-08-08 17:42:05 -0500
+ (Wed, 08 Aug 2012) | 26 lines Fix Not Unreferencing A Spied
+ Channel When a channel hangs up while being spied upon and the
+ option to exit the ChanSpy application when the spied on channel
+ hangs up is set, ast_autochan_destroy is not being called and
+ therefore a reference to the spied upon channel is not removed.
+ The symptom being reported was that when using func_group in the
+ dialplan and calling "group show channels" at the cli, the spied
+ upon channel was still being shown while "core show channels"
+ showed that the channel was not up. This patch calls
+ ast_autochan_destroy when a spied upon channel hangs up and the
+ option to exit the ChanSpy application is set, removing the
+ reference to the channel allowing the count for the group that
+ the spied channel was part of to be decremented. (closes issue
+ ASTERISK-17515) Reported by: Arkadiusz Malka Tested by: Alexandr
+ Gordeev, Michael L. Young Patches:
+ asterisk-17515-destroy-autochan.diff uploaded by Michael L. Young
+ (license 5026) ........ Merged revisions 370952 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/channel.c, /: Merged revisions 370924 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370924 | kmoore | 2012-08-08 15:29:16 -0500
+ (Wed, 08 Aug 2012) | 9 lines Do not define a cause that doesn't
+ actually exist AST_CAUSE_NOTDEFINED is a placeholder for usage
+ when there is no cause information. As such, it should not be
+ defined and translatable as a cause. ........ Merged revisions
+ 370923 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/sig_analog.h, channels/chan_dahdi.c,
+ channels/sig_analog.c, /: Merged revisions 370901 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370901 | rmudgett | 2012-08-08 15:04:44 -0500
+ (Wed, 08 Aug 2012) | 5 lines Fix the analog dial *0 flash-hook of
+ bridged peer feature. ........ Merged revisions 370900 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-08-07 20:23 +0000 [r370880] Automerge script <automerge@asterisk.org>
+
+ * main/channel.c, /: Merged revisions 370858 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370858 | kmoore | 2012-08-07 14:21:54 -0500
+ (Tue, 07 Aug 2012) | 5 lines Add missing AST_CAUSE_* -> text
+ translations ........ Merged revisions 370856 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-08-06 15:24 +0000 [r370817] Automerge script <automerge@asterisk.org>
+
+ * /, channels/chan_sip.c: Merged revisions 370798 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370798 | mmichelson | 2012-08-06 10:02:04 -0500
+ (Mon, 06 Aug 2012) | 7 lines Improve debug message for temporary
+ outbound proxies. Thanks to Paul Belanger for pointing this out.
+ ........ Merged revisions 370797 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-08-03 22:26 +0000 [r370793] Automerge script <automerge@asterisk.org>
+
+ * channels/sip/config_parser.c, channels/sip/include/sip.h, /,
+ channels/chan_sip.c: Merged revisions 370772 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370772 | mmichelson | 2012-08-03 16:50:29 -0500
+ (Fri, 03 Aug 2012) | 41 lines Multiple revisions 370769-370771
+ ........ r370769 | mmichelson | 2012-08-03 16:35:00 -0500 (Fri,
+ 03 Aug 2012) | 24 lines Fix error in the "IPorHost" section of a
+ SIP dialstring. This is based on the review request posted by
+ Walter Doekes (referenced lower in the commit message) The main
+ fix here is to treat the IPorHost portion of the dial string as a
+ temporary outbound proxy. This ensures requests get sent to the
+ proper location. Due to the age of the request, some parts were
+ no longer relevant. For instance, the request moved outbound
+ proxy parsing code into a single method. This is done in a
+ previous commit, so it was not necessary to do again. Also, the
+ review request fixed some errors with regards to request routing
+ for CANCEL and ACK requests. This has also been fixed in more
+ recent commits. (closes issue ASTERISK-19677) reported by Walter
+ Doekes Review https://reviewboard.asterisk.org/r/1859 ........
+ r370770 | mmichelson | 2012-08-03 16:39:35 -0500 (Fri, 03 Aug
+ 2012) | 3 lines Remove unused variable. ........ r370771 |
+ mmichelson | 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) | 5
+ lines Seriously? Another compilation error fixed. Somebody beat
+ me. ........ Merged revisions 370769-370771 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-08-01 03:22 +0000 [r370633-370717] Automerge script <automerge@asterisk.org>
+
+ * utils/extconf.c, /: Merged revisions 370698 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370698 | kmoore | 2012-07-31 21:26:09 -0500
+ (Tue, 31 Jul 2012) | 8 lines Revert alloca changes for utils
+ These changes were a tad overzealous in the utils directory.
+ Unfortunately, these don't compile with a "make". ........ Merged
+ revisions 370697 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 370672 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370672 | mjordan | 2012-07-31 16:19:41 -0500
+ (Tue, 31 Jul 2012) | 24 lines Schedule pokes of registered SIP
+ peers within a given timespan after SIP reload With a large
+ number of SIP peers registered, performing a SIP reload causes a
+ flood of SIP OPTIONS request packets. These are immediately sent
+ out, and, as responses come back, can cause peers to be flagged
+ as 'lagged' due to handling of the many response messages. This
+ fix prevents this "packet storm" and schedules the pokes for a
+ random time. That time varies between 1 ms and the peer's qualify
+ time, or, if the qualify time is unknown, the global qualifyfreq
+ setting. The committed patch has some very small modifications to
+ the patch schmidts wrote for the review. (closes issue
+ ASTERISK-19154) Reported by: Nicolo Mazzon patches:
+ issue19154.patch license #6034 uploaded by schmidts Review:
+ https://reviewboard.asterisk.org/r/1652 ........ Merged revisions
+ 370666 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * funcs/func_cut.c, tests/test_linkedlists.c,
+ channels/chan_gtalk.c, cdr/cdr_pgsql.c, main/config.c,
+ channels/chan_jingle.c, pbx/pbx_spool.c,
+ apps/app_directed_pickup.c, funcs/func_channel.c,
+ apps/app_minivm.c, main/features.c, res/res_agi.c, main/http.c,
+ main/logger.c, pbx/pbx_ael.c, apps/app_macro.c, main/event.c,
+ apps/app_sms.c, main/astmm.c, include/asterisk/strings.h,
+ main/db.c, main/dsp.c, apps/app_voicemail.c, addons/app_mysql.c,
+ channels/chan_sip.c, main/threadstorage.c, main/say.c,
+ apps/app_dictate.c, apps/app_festival.c, funcs/func_strings.c,
+ pbx/pbx_lua.c, main/utils.c, funcs/func_logic.c,
+ apps/app_getcpeid.c, channels/chan_iax2.c, res/res_jabber.c,
+ funcs/func_global.c, main/channel.c, res/ael/pval.c,
+ main/tcptls.c, apps/app_osplookup.c, main/manager.c,
+ main/strcompat.c, main/callerid.c, main/file.c, main/app.c,
+ channels/chan_alsa.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
+ utils/extconf.c, addons/chan_mobile.c, apps/app_mixmonitor.c,
+ main/asterisk.c, apps/app_while.c, addons/res_config_mysql.c,
+ res/res_config_pgsql.c, main/pbx.c, include/asterisk/utils.h, /,
+ apps/app_meetme.c: Merged revisions 370643 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370643 | kmoore | 2012-07-31 14:57:09 -0500
+ (Tue, 31 Jul 2012) | 12 lines Clean up and ensure proper usage of
+ alloca() This replaces all calls to alloca() with ast_alloca()
+ which calls gcc's __builtin_alloca() to avoid BSD semantics and
+ removes all NULL checks on memory allocated via ast_alloca() and
+ ast_strdupa(). (closes issue ASTERISK-20125) Review:
+ https://reviewboard.asterisk.org/r/2032/ ........ Merged
+ revisions 370642 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * configs/sip.conf.sample, channels/sip/include/sip.h, /,
+ channels/chan_sip.c: Merged revisions 370619 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370619 | mmichelson | 2012-07-31 10:31:57 -0500
+ (Tue, 31 Jul 2012) | 34 lines Help mitigate potential reinvite
+ glare scenarios. When Asterisk servers are set up back-to-back,
+ and direct media is to be used betweeen endpoints, it is fairly
+ common for the two Asterisk servers to send direct media
+ reinvites to each other simultaneously. This results in 491s and
+ ACKs being exchanged between the servers. While the media
+ eventually gets set up properly, the problem is that there can be
+ a noticeable delay for the streams to stabilize. This patch adds
+ a new directmedia option called "outgoing". With this set, an
+ immediate direct media reinvite will only be sent if the call
+ direction is outgoing. For incoming dialogs, an immediate direct
+ media reinvite will not be sent, but further "reactionary" direct
+ media reinvites may be sent. For those who are having some deja
+ vu, that's because this patch was originally committed to trunk
+ since there is a new configuration option added. After seeing a
+ bug report about audio being slow to set up on SIP calls, it
+ became apparent that this patch would be the best solution for
+ resolving the issue. The patch is unintrusive and will have no
+ effect unless the option is explicitly enabled. (closes issue
+ AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857)
+ reported by Matt Jordan ........ Merged revisions 370618 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-09-13 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.8.0-digiumphones Released.
+
+2012-09-11 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.8.0-digiumphones-rc2 Released.
+
+ * AST-2012-013: Resolve ACL rules being ignored during calls by some
+ IAX2 peers
+
+ * AST-2012-012: Resolve AMI User Unauthorized Shell Access through
+ ExternalIVR
+
+ * r371861: Fix hangup cause passthrough regression.
+
+ The v1.8 -r369258 change to fix the F and F(x) action logic
+ introduced a regression in passing the hangup cause from the called
+ channel to the caller channel.
+
+ (closes issue ASTERISK-20287)
+ Reported by: Konstantin Suvorov
+ Patches:
+ app_dial_hangupcause.patch (license #6421) patch uploaded by
+ Konstantin Suvorov (modified)
+ Tested by: rmudgett
+
+ * r372710: Only re-create an SRTP session when needed; respond with
+ correct crypto policy
+
+ In r356604, SRTP handling was fixed to accomodate multiple crypto
+ keys in an SDP offer and the ability to re-create an SRTP session
+ when the crypto keys changed. In certain circumstances - most
+ notably when a phone is put on hold after having been bridged for a
+ significant amount of time - the act of re-creating the SRTP session
+ causes problems for certain models of phones. The patch committed in
+ r356604 always re-created the SRTP session regardless of whether or
+ not the cryptographic keys changed. Since this is technically
+ not necessary, this patch modifies the behavior to only re-create the
+ SRTP session if Asterisk detects that the remote key has changed.
+ This allows models of phones that do not handle the SRTP session
+ changing to continue to work, while also providing the behavior
+ needed for those phones that do re-negotiate cryptographic keys.
+
+ (issue ASTERISK-20194)
+ Reported by: Nicolo Mazzon
+ Tested by: Nicolo Mazzon
+
+ Review: https://reviewboard.asterisk.org/r/2099
+
+ * r372841: Fix bad channel application data reference.
+
+ When channels get bridged due to an AMI bridge action
+ or a DTMF attended transfer, the two channels that
+ get bridged have their application data pointing to
+ the other channel's name. This means that if one channel
+ is hung up but the other moves on, it means that the
+ channel that moves on will have its application data
+ pointing at freed memory.
+
+ (issue ASTERISK-20335)
+
+2012-07-31 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.8.0-digiumphones-rc1 Released.
+
+2012-07-30 17:24 +0000 [r370555-370584] Automerge script <automerge@asterisk.org>
+
+ * channels/chan_misdn.c, /: Merged revisions 370564 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370564 | rmudgett | 2012-07-30 11:49:12 -0500
+ (Mon, 30 Jul 2012) | 5 lines Release B channel allocation on
+ error path in chan_misdn. ........ Merged revisions 370563 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, apps/app_meetme.c: Merged revisions 370547 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r370547 | jrose | 2012-07-30 09:50:34 -0500 (Mon, 30 Jul 2012) |
+ 5 lines app_meetme: Change app_meetme support level to extended
+ from deprecated (closes issue ASTERISK-20134) Reported by: Leif
+ Madsen ........
+
+2012-07-25 21:22 +0000 [r370509] Automerge script <automerge@asterisk.org>
+
+ * res/res_agi.c, /: Merged revisions 370495 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370495 | jrose | 2012-07-25 16:12:50 -0500
+ (Wed, 25 Jul 2012) | 14 lines res_agi: Add message indicating
+ need for \n character in verbose message The while loop
+ responsible for reading AGI messages from a fastAGI service can
+ end up looping indefinitely when an AGI script fails to indicate
+ the end of a message with a \n character. This patch adds an
+ indication that we are expecting a \n character to end the
+ message to make it more clear to users that this is necessary if
+ they are receiving this warning over and over. (issue
+ ASTERISK-20061) Reported by: Eike Kuiper ........ Merged
+ revisions 370494 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-25 03:45 +0000 [r370473] Terry Wilson <twilson@digium.com>
+
+ * main/pbx.c, /: Revert a change that broke compilation 1) There is
+ no such function as ast_ref() 2) The patch was originally
+ credited as the one uploaded by Guenther Kelleter (license 6372)
+ via issue AST-921, but the patch committed was not the patch
+ referenced on the issue. 3) Guenther Kelleter's patch was
+ actually correct. It moved the ast_free above the
+ presencechange_cleanup label. I am not committing his change as
+ it is not technically necesary--calling ast_free(NULL) is
+ perfectly safe and I worry that moving the ast_free outside of
+ the label could lead to future bugs if someone ever adds another
+ failure conditional and expects 'goto presencechange_cleanup;' to
+ clean up after everything.
+
+2012-07-24 21:08 +0000 [r370465] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c, /: Don't attempt free of NULL ptr in pbx.c
+ handle_presencechange (closes issue AST-921) Reported by:
+ Guenther Kelleter Patches: nullptr.patch uploaded by Guenther
+ Kelleter (license 6372)
+
+2012-07-24 17:24 +0000 [r370381-370452] Automerge script <automerge@asterisk.org>
+
+ * channels/chan_oss.c, main/frame.c, /: Merged revisions
+ 370430,370432 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370430 | kpfleming | 2012-07-24 11:54:01 -0500
+ (Tue, 24 Jul 2012) | 5 lines Rewrite a comment that didn't
+ adequately explain the code it was documenting. ........ Merged
+ revisions 370429 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r370432 | tzafrir | 2012-07-24 12:08:40 -0500
+ (Tue, 24 Jul 2012) | 4 lines chan_oss: fix "sample rate" error
+ message Merged revisions 370428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, funcs/func_shell.c: Merged revisions 370384 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370384 | kpfleming | 2012-07-23 16:09:53 -0500
+ (Mon, 23 Jul 2012) | 5 lines Improve documentation for the
+ SHELL() dialplan function. ........ Merged revisions 370383 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/channel.c, /: Merged revisions 370361 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370361 | kpfleming | 2012-07-23 09:51:21 -0500
+ (Mon, 23 Jul 2012) | 13 lines Free any datastores attached to
+ dummy channels. Revision 370205 added the use of a datastore
+ attached to a dummy channel to resolve a memory leak, but
+ ast_dummy_channel_destructor() in this branch did not free
+ datastores, resulting in a continued (but slightly smaller)
+ memory leak. This patch backports the change to free said
+ datastores from the Asterisk trunk. (related to issue AST-916)
+ ........ Merged revisions 370360 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-19 22:24 +0000 [r370297] Automerge script <automerge@asterisk.org>
+
+ * main/cel.c, res/res_rtp_asterisk.c, /: Merged revisions
+ 370271,370274,370277 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370271 | mjordan | 2012-07-19 16:37:09 -0500
+ (Thu, 19 Jul 2012) | 49 lines Handle extremely out of order RFC
+ 2833 DTMF The current implementation of RFC 2833 DTMF handling in
+ res_rtp_asterisk will, if a packet arrives out of order, drop the
+ packet. This is to prevent duplicate ton generation in the
+ Asterisk core. Since the RTP layer does not buffer data itself,
+ this is the only option the RTP layer currently has for handling
+ packets that arrive out of order. For the most part, this doesn't
+ matter. For a particular digit, so long as a BEGIN packet arrives
+ before the first END packet, the digit will be produced. If
+ subsequent BEGIN packets arrive interleaved with the ENDs, they
+ will be dropped; likewise, if the BEGIN or END packets themselves
+ are out of order, those packets are dropped but sufficient
+ information is conveyed to the Asterisk core to produce the
+ appropriate digit. For certain sequences of DTMF packets - most
+ notably when, for a particular digit, an END packet arrives
+ before any BEGIN packet for that digit - this is a real problem.
+ When an END arrives before any BEGINs, the END packet is dropped
+ - but at the same time, it causes subsequent BEGIN packets for
+ that digit to be ignored. When the next in order END packet
+ arrives, it too is dropped - Asterisk believes that there was no
+ initial BEGIN. The solution this patch provides is to trust the
+ END packet to convey the information needed for the Asterisk core
+ to produce the DTMF digit. If we receive an END packet, and it: *
+ Has a timestamp greater then the last timestamp received from an
+ END packet * Does not have the same sequence number as the last
+ received sequence number (and is thus not an END packet
+ retransmission) Then we send the END frame up to the Asterisk
+ core. It contains enough DTMF information for Asterisk to produce
+ the digit. On the other hand, if we receive a BEGIN or
+ continuation packet that occurs with a timestamp equal to or less
+ then the last END timestamp, then we've received something out of
+ order - but we already have received enough information to
+ produce the digit. These packets are dropped. Much thanks goes to
+ Olle Johansson (oej) for providing the idea for this solution.
+ Review: https://reviewboard.asterisk.org/r/2033/ (issue
+ ASTERISK-18404) Reporter: Stephane Chazelas Tested by: Matt
+ Jordan ........ Merged revisions 370252 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r370274 | mjordan | 2012-07-19 17:01:32 -0500
+ (Thu, 19 Jul 2012) | 17 lines Fix compilation error when
+ MALLOC_DEBUG is enabled To fix a memory leak in CEL, a channel
+ datastore was introduced whose destruction function pointer was
+ pointed to the ast_free macro. Without MALLOC_DEBUG enabled this
+ compiles as fine, as ast_free is defined as free. With
+ MALLOC_DEBUG enabled, however, ast_free takes on a definition
+ from a different place then utils.h, and became undefined. This
+ patch resolves this by using a reference to ast_free_ptr. When
+ MALLOC_DEBUG is enabled, this calls ast_free; when MALLOC_DEBUG
+ is not enabled, this is defined to be ast_free, which is defined
+ to be free. (issue AST-916) Reported by: Thomas Arimont ........
+ Merged revisions 370273 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r370277 | rmudgett | 2012-07-19 17:11:48 -0500
+ (Thu, 19 Jul 2012) | 7 lines Fix compiler warnings. gcc (GCC)
+ 4.2.4 has problems casting away constness. ........ Merged
+ revisions 370275 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-18 19:23 +0000 [r370202-370224] Automerge script <automerge@asterisk.org>
+
+ * main/cel.c, /: Merged revisions 370206 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370206 | kpfleming | 2012-07-18 14:14:09 -0500
+ (Wed, 18 Jul 2012) | 19 lines Resolve severe memory leak in CEL
+ logging modules. A customer reported a significant memory leak
+ using Asterisk 1.8. They have tracked it down to
+ ast_cel_fabricate_channel_from_event() in main/cel.c, which is
+ called by both in-tree CEL logging modules (cel_custom.c and
+ cel_sqlite3_custom.c) for each and every CEL event that they log.
+ The cause was an incorrect assumption about how data attached to
+ an ast_channel would be handled when the channel is destroyed;
+ the data is now stored in a datastore attached to the channel,
+ which is destroyed along with the channel at the proper time.
+ (closes issue AST-916) Review:
+ https://reviewboard.asterisk.org/r/2053/ ........ Merged
+ revisions 370205 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * funcs/func_lock.c, apps/app_macro.c, channels/chan_iax2.c,
+ apps/app_mixmonitor.c, apps/app_stack.c, funcs/func_global.c,
+ res/res_odbc.c, main/channel.c, addons/app_mysql.c, main/pbx.c,
+ funcs/func_curl.c, /, main/ccss.c, funcs/func_odbc.c: Merged
+ revisions 370184 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370184 | kpfleming | 2012-07-18 12:13:07 -0500
+ (Wed, 18 Jul 2012) | 10 lines Ensure that all ast_datastore_info
+ structures are 'const'. While addressing a bug, I came across a
+ instance of 'struct ast_datastore_info' that was not declared
+ 'const'. Since the API already expects them to be 'const', this
+ patch changes the declarations of all existing instances that
+ were not already declared that way. ........ Merged revisions
+ 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-16 20:24 +0000 [r370101-370151] Automerge script <automerge@asterisk.org>
+
+ * /, channels/chan_sip.c: Merged revisions 370132 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370132 | wdoekes | 2012-07-16 14:52:45 -0500
+ (Mon, 16 Jul 2012) | 11 lines Code cleanup and bugfix in chan_sip
+ outboundproxy parsing. The bug was clearing the global
+ outboundproxy when a peer-specific outboundproxy was bad. The
+ cleanup reduces duplicate code. Review:
+ https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark
+ Michelson ........ Merged revisions 370131 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * UPGRADE.txt, CHANGES, UPGRADE-1.8.txt, /: Merged revisions 370082
+ via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370082 | kmoore | 2012-07-16 08:51:57 -0500
+ (Mon, 16 Jul 2012) | 8 lines Add comments about the BUILD_NATIVE
+ change This is a significant change and mention of it should have
+ gone into UPGRADE.txt and CHANGES. ........ Merged revisions
+ 370081 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-12 20:24 +0000 [r369958-370036] Automerge script <automerge@asterisk.org>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/chan_sip.c: Merged revisions 370015,370025 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370015 | kmoore | 2012-07-12 15:05:45 -0500
+ (Thu, 12 Jul 2012) | 11 lines Include Expires header for SIP
+ PUBLISH requests RFC3903 requres SIP PUBLISH requests to have
+ Expires headers, so add them. Review:
+ https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth
+ ........ Merged revisions 370014 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r370025 | rmudgett | 2012-07-12 15:20:02 -0500
+ (Thu, 12 Jul 2012) | 8 lines Add missing ast_hangup() calls on
+ some analog exception paths. Make starting analog_ss_thread() or
+ __analog_ss_thread() failure paths hangup the channel. ........
+ Merged revisions 370017 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369994 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369994 | kmoore | 2012-07-12 13:55:17 -0500
+ (Thu, 12 Jul 2012) | 12 lines Prevent double uri_escaping in
+ chan_sip when pedantic is enabled If pedantic mode is enabled,
+ outbound invites will have double-escaped contacts. This avoids
+ setting an already-escaped string into a field where it is
+ expected to be unescaped. (closes issue ASTERISK-20023)
+ Reported-by: Walter Doekes ........ Merged revisions 369993 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * funcs/func_math.c, /: Merged revisions 369971 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369971 | elguero | 2012-07-12 09:25:45 -0500
+ (Thu, 12 Jul 2012) | 14 lines Correct Documentation For DEC
+ Function The documentation for DEC in func_math.c was incorrect.
+ Looks like a copy and paste error. (Closes issue ASTERISK-20095)
+ Reported by: Billy Chia Tested by: Michael L. Young Patches:
+ func_math.patch uploaded by Billy Chia (license 6381) ........
+ Merged revisions 369970 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * funcs/func_realtime.c, /: Merged revisions 369938 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369938 | tilghman | 2012-07-11 12:12:28 -0500
+ (Wed, 11 Jul 2012) | 11 lines Allow the REALTIME() function to
+ report errors back to the caller. Also, do more error checking on
+ the arguments specified to the REALTIME() function and clarify
+ the documentation. While I was editing the file, a few coding
+ guidelines fixups, as well. Review:
+ https://reviewboard.asterisk.org/r/2031/ ........ Merged
+ revisions 369937 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-30 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.7.0-digiumphones Released.
+
+2012-07-11 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.7.0-digiumphones-rc1 Released.
+
+2012-07-10 14:22 +0000 [r369889] Automerge script <automerge@asterisk.org>
+
+ * apps/app_stack.c, main/pbx.c, /: Merged revisions 369871 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369871 | kmoore | 2012-07-10 08:35:30 -0500
+ (Tue, 10 Jul 2012) | 12 lines Improve Goto and GotoIf related
+ documentation Correct documentation on labeliftrue and
+ labeliffalse parameters of GotoIf() and update several other
+ locations that use the same syntax. (closes issue ASTERISK-20007)
+ Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged
+ revisions 369869 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-09 19:51 +0000 [r369846] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Add support for exposing the received
+ contact URI and also for setting the request URI in messages.
+ (closes issue AST-911)
+
+2012-07-09 17:22 +0000 [r369810-369836] Automerge script <automerge@asterisk.org>
+
+ * configs/sip_notify.conf.sample, /: Merged revisions 369819 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369819 | qwell | 2012-07-09 12:06:40 -0500
+ (Mon, 09 Jul 2012) | 9 lines Add Digium phones context to
+ sip_notify sample config. This makes it so that they can be
+ reconfigured remotely. (closes issue ASTERISK-19910) ........
+ Merged revisions 369818 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369793 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369793 | jrose | 2012-07-09 09:43:49 -0500
+ (Mon, 09 Jul 2012) | 9 lines chan_sip: Fix small behavioral
+ change accidentally introduced in r369750 When removing the
+ warning for AST_CONTROL_FLASH from sip_indicate, I also
+ inadvertently changed the return value, which would likely make
+ the indication not be sent in audio. This fixes that while still
+ removing the warning message. ........ Merged revisions 369792
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-06 21:21 +0000 [r369643-369763] Automerge script <automerge@asterisk.org>
+
+ * /, channels/chan_sip.c: Merged revisions 369751 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369751 | jrose | 2012-07-06 16:02:37 -0500
+ (Fri, 06 Jul 2012) | 12 lines chan_sip: Add case for FLASH
+ control frames so that we don't display a warning. chan_sip
+ channels can receive flash control frames when connected to
+ analog phones and possibly for other reasons. There really isn't
+ a reason to warn when these frames are received, we can safely
+ ignore them. Patches: dahdi_sip_flash.diff uploaded by Jonathan
+ Rose (license 6182) ........ Merged revisions 369750 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/tcptls.c, /: Merged revisions 369732 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369732 | mmichelson | 2012-07-06 13:47:05 -0500
+ (Fri, 06 Jul 2012) | 21 lines Remove a superfluous and dangerous
+ freeing of an SSL_CTX. The problem here is that multiple server
+ sessions share a SSL_CTX. When one session ended, the SSL_CTX
+ would be freed and set NULL, leaving the other sessions unable to
+ function. The code being removed is superfluous because the
+ SSL_CTX structures for servers will be properly freed when
+ ast_ssl_teardown is called. (closes issue ASTERISK-20074)
+ Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded
+ by Mark Michelson (license #5049) Testers: Trevor Helmsley
+ ........ Merged revisions 369731 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/bridging.c, /: Merged revisions 369709 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369709 | mmichelson | 2012-07-06 10:23:28 -0500
+ (Fri, 06 Jul 2012) | 14 lines Fix bridging thread leak. The
+ bridge thread was exiting but was never being reaped using
+ pthread_join(). This has been fixed now by calling pthread_join()
+ in ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported
+ by Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012
+ ........ Merged revisions 369708 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 369653 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369653 | kmoore | 2012-07-05 14:12:33 -0500
+ (Thu, 05 Jul 2012) | 20 lines Resolve heap corruption issue with
+ voicemail The heard and deleted arrays in the voicemail state
+ structure were not handled properly following the memory leak fix
+ in r354890 and a fix for an invalid free in r356797. This could
+ result in accessing and writing into freed memory. The allocation
+ for these arrays has been reworked to avoid the possibility of
+ invalid frees, access of freed memory, and crashes that were
+ occurring as a result of this. Locking around accesses and
+ modifications of the voicemail state structure members
+ dh_arraysize, heard, and deleted has been added to prevent
+ simultaneous modification and access when IMAP storage is in use.
+ If IMAP storage is not in use, this locking is not compiled in.
+ Review: https://reviewboard.asterisk.org/r/1994/ (closes issue
+ ASTERISK-19923) ........ Merged revisions 369652 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369627 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369627 | mjordan | 2012-07-05 12:02:53 -0500
+ (Thu, 05 Jul 2012) | 18 lines Do not send a BYE when a
+ provisional response arrives during a re-INVITE Commits r369557
+ and r369579 were done to improve handling of re-INVITEs when the
+ UA that was supposed to receive the re-INVITE fails to respond. A
+ limitation of those patches occurred when a UA sent a provisional
+ response to the re-INVITE. This triggered a sending of a BYE in
+ check_pending. This patch tweaks the handling of the re-INVITE
+ such that a BYE is not sent in response to those messages. (issue
+ ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies
+ patches: (reinvite_tweak.diff license #5012 by Steve Davies)
+ ........ Merged revisions 369626 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-03 17:23 +0000 [r369578-369598] Automerge script <automerge@asterisk.org>
+
+ * /, channels/chan_sip.c: Merged revisions 369580 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369580 | twilson | 2012-07-03 12:02:18 -0500
+ (Tue, 03 Jul 2012) | 11 lines More improvements to re-INVITEs
+ timing out after a provisional response There is no need to call
+ check_pendings() on a final response to an INVITE when destroying
+ the scheduler entry as it will be done later during normal
+ processing. (issue ASTERISK-19992) ........ Merged revisions
+ 369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+ revisions 369558 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369558 | twilson | 2012-07-03 09:34:22 -0500
+ (Tue, 03 Jul 2012) | 14 lines Better handle re-INVITEs with
+ provisional but no final repsonses A previous attempt at fixing
+ this issue had negative side effects related to attended
+ transfers which this patch should resolve. Many thanks to Steve
+ Davies for all of the good suggestions and testing. (closes issue
+ ASTERISK-19992) Reported by: Steve Davies Tested by: Steve
+ Davies, Terry Wilson Review:
+ https://reviewboard.asterisk.org/r/2009/ ........ Merged
+ revisions 369557 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-06-29 21:19 +0000 [r369488-369516] Automerge script <automerge@asterisk.org>
+
+ * main/rtp_engine.c, /: Merged revisions 369511 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r369511 | mmichelson | 2012-06-29 15:28:10 -0500 (Fri, 29 Jun
+ 2012) | 3 lines Fix apparent copy and paste error where incorrect
+ "glue" is used. ........
+
+ * /, channels/chan_sip.c: Merged revisions 369491 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369491 | file | 2012-06-29 11:54:11 -0500 (Fri,
+ 29 Jun 2012) | 5 lines With some configurations a transport is
+ not actually specified so assume UDP in these cases. ........
+ Merged revisions 369490 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369472 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369472 | file | 2012-06-29 10:30:47 -0500 (Fri,
+ 29 Jun 2012) | 10 lines Make the address family filter specific
+ to the transport. (closes issue ASTERISK-16618) Reported by: Leif
+ Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........
+ Merged revisions 369471 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-06-27 21:22 +0000 [r369453] Automerge script <automerge@asterisk.org>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+ revisions 369437 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369437 | twilson | 2012-06-27 16:10:01 -0500
+ (Wed, 27 Jun 2012) | 16 lines Clean up after a reinvite that
+ never gets a final response The basic problem is that if a
+ re-INVITE is sent by Asterisk and it receives a provisional
+ response, but no final response, then the dialog is never torn
+ down. In addition to leaking memory, this also leaks file
+ descriptors and will eventually lead to Asterisk no longer being
+ able to process calls. This patch just keeps track of whether
+ there is an outstanding re-INVITE, and if there is goes ahead and
+ cleans up everything as though there was no outstanding reinvite.
+ (closes issue ASTERISK-19992) ........ Merged revisions 369436
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-06-26 14:21 +0000 [r369322-369406] Automerge script <automerge@asterisk.org>
+
+ * main/adsi.c, /: Merged revisions 369391 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369391 | mjordan | 2012-06-26 08:22:42 -0500
+ (Tue, 26 Jun 2012) | 15 lines Fix crash in unloading of res_adsi
+ module When res_adsi is unloaded, it removes the ADSI functions
+ that it previously installed by passing a NULL adsi_funcs pointer
+ to ast_adsi_install_funcs. This function was not checking whether
+ or not the adsi_funcs pointer passed in was NULL before
+ dereferencing it to check whether or not the version of the
+ functions matches what the core was expecting it. This patch
+ makes it so that the version is only checked if a potentially
+ valid adsi_funcs pointer was passed in. Passing in NULL removes
+ the installed functions, bypassing the version check. ........
+ Merged revisions 369390 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/cdr.c, /: Merged revisions 369369 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369369 | mjordan | 2012-06-25 14:36:02 -0500
+ (Mon, 25 Jun 2012) | 29 lines Fix incorrect duration reporting in
+ CDRs created in batch mode Certain places in core/cdr.c would, if
+ the duration value were 0, calculate the duration as being the
+ delta between the current time and the time at which the CDR
+ record was started. While this does not typically cause a problem
+ in non-batch mode, this can cause an issue in batch mode where
+ CDR records are gathered and written long after those calls have
+ ended. In particular, this affects calls that were never
+ answered, as those are expected to have a duration of 0. Often,
+ this would result in CDR logs with a significant number of calls
+ with lengthy durations, but dispositions of "BUSY". Note that
+ this does not affect cdr_csv, as that backend does not use
+ ast_cdr_getvar and instead directly reports the duration value.
+ The affected core backends include cdr_apative_odbc and
+ cdr_custom; other extended or deprecated CDR backends may
+ potentially still directly manipulate the duration values. (issue
+ ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883)
+ Reported by: Thomas Arimont Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1996/ ........ Merged
+ revisions 369351 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+ revisions 369353 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369353 | mmichelson | 2012-06-25 14:16:52 -0500
+ (Mon, 25 Jun 2012) | 14 lines Re-fix how local tag is generated
+ when sending a 481 to an INVITE. Match our local tag to whatever
+ to-tag was sent in the initial INVITE. Because the size of the
+ to-tag may not fit in the buffer in the sip_pvt, it has been
+ changed to a string field. (closes issue ASTERISK-19892) reported
+ by Walter Doekes Review: https://reviewboard.asterisk.org/r/1977
+ ........ Merged revisions 369352 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/features.c, res/res_adsi.c, main/adsi.c (added),
+ res/res_adsi.exports.in (removed), include/asterisk/adsi.h, /,
+ main/Makefile: Merged revisions 369325,369328 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369325 | mmichelson | 2012-06-25 10:52:42 -0500
+ (Mon, 25 Jun 2012) | 20 lines Multiple revisions 369323-369324
+ ........ r369323 | mmichelson | 2012-06-25 10:35:43 -0500 (Mon,
+ 25 Jun 2012) | 9 lines Eliminate embedding of res_adsi.so module.
+ The way this is done is to stop using the optional API. Instead,
+ res_adsi.so, when loaded fills in a table of function pointers.
+ Review: https://reviewboard.asterisk.org/r/1991 ........ r369324
+ | mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2
+ lines Forgot to svn add this file in my last commit. ........
+ Merged revisions 369323-369324 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r369328 | rmudgett | 2012-06-25 10:59:28 -0500
+ (Mon, 25 Jun 2012) | 15 lines Fix Bridge application occasionally
+ returning to the wrong location. * Fix do_bridge_masquerade()
+ getting the resume location from the zombie channel. The code
+ must not touch a clone channel after it has masqueraded it. The
+ clone channel has become a zombie and is starting to hangup.
+ (closes issue ASTERISK-19985) Reported by: jamicque Patches:
+ jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: jamicque ........ Merged revisions 369327
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369303 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369303 | mmichelson | 2012-06-25 09:23:16 -0500
+ (Mon, 25 Jun 2012) | 14 lines Be more consistent with the return
+ code for requests received from invalid domain. When Asterisk
+ receives an INVITE from an external domain when
+ allowexternaldomains=no send a 403 instead of a 404. This is
+ consistent with Asterisk's behavior when receiving a REGISTER in
+ this situation. (Closes issue ASTERISK-19601) Reported by Matthew
+ Jordan Patches: ASTERISK-19601-no401.patch uploaded by Mark
+ Michelson (License #5049) ........ Merged revisions 369302 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-06-23 00:20 +0000 [r369213-369294] Automerge script <automerge@asterisk.org>
+
+ * main/features.c, /: Merged revisions 369283 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369283 | rmudgett | 2012-06-22 19:12:27 -0500
+ (Fri, 22 Jun 2012) | 22 lines Fix Bridge application and AMI
+ Bridge action error handling. * Fix AMI Bridge action
+ disconnecting the AMI link on error. * Fix AMI Bridge action and
+ Bridge application not checking if their masquerades were
+ successful. * Fix Bridge application running the h-exten when it
+ should not. * Made do_bridge_masquerade() return if the
+ masquerade was successful so the Bridge application and AMI
+ Bridge action could deal with it correctly. * Made
+ bridge_call_thread_launch() hangup the passed in channels if the
+ bridge_call_thread fails to start. Those channels would have been
+ orphaned. * Made builtin_atxfer() check the success of the
+ transfer masquerade setup. ........ Merged revisions 369282 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_queue.c, apps/app_dial.c, /: Merged revisions
+ 369259,369263 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369259 | rmudgett | 2012-06-22 16:37:05 -0500
+ (Fri, 22 Jun 2012) | 5 lines Check if PBX was started and fix F
+ and F(x) action logic in Dial application. ........ Merged
+ revisions 369258 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r369263 | rmudgett | 2012-06-22 17:09:29 -0500
+ (Fri, 22 Jun 2012) | 5 lines Explicitly check caller hangup in
+ app Queue rather than a polluted res2 value. ........ Merged
+ revisions 369262 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c, main/ccss.c: Merged revisions
+ 369236,369239 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369236 | rmudgett | 2012-06-22 15:49:33 -0500
+ (Fri, 22 Jun 2012) | 5 lines Change incorrect chan_sip zombie
+ hangup debug message. They are all zombies now. ........ Merged
+ revisions 369235 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r369239 | rmudgett | 2012-06-22 16:04:25 -0500
+ (Fri, 22 Jun 2012) | 5 lines Check if PBX was started for generic
+ CCSS recall. ........ Merged revisions 369238 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369215 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369215 | twilson | 2012-06-22 14:34:59 -0500
+ (Fri, 22 Jun 2012) | 9 lines Don't crash on a guest directmedia
+ call A sip_pvt may not have relatedpeer set if a call doesn't
+ match up with a peer. If there is no relatedpeer, there is no
+ direct media ACL to apply, so just return that it is allowed.
+ ........ Merged revisions 369214 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369206 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369206 | kmoore | 2012-06-22 12:23:26 -0500
+ (Fri, 22 Jun 2012) | 11 lines Don't parse media stream state for
+ SIP video streams The sendonly/recvonly/sendrecv/inactive media
+ stream attributes were parsed for video, but nothing was ever
+ done with them. With this code removed, an UNSUPPORTED message is
+ produced when these attributes are used in conjunction with a
+ video stream which is the better behavior since they were never
+ really supported in the first place. ........ Merged revisions
+ 369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-06-20 18:22 +0000 [r369056-369164] Automerge script <automerge@asterisk.org>
+
+ * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c, /:
+ Merged revisions 369147 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369147 | may | 2012-06-20 12:36:27 -0500 (Wed,
+ 20 Jun 2012) | 10 lines fix locking issue on empty callList
+ (issue ASTERISK-19298) Reported by: Dmitry Melekhov Patches:
+ ASTERISK-18322-2.patch ........ Merged revisions 369146 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * include/asterisk/netsock2.h, main/netsock2.c, /: Merged revisions
+ 369109 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369109 | elguero | 2012-06-19 21:04:58 -0500
+ (Tue, 19 Jun 2012) | 23 lines Fix NULL pointer segfault in
+ ast_sockaddr_parse() While working with ast_parse_arg() to
+ perform a validity check, a segfault occurred. The segfault
+ occurred due to passing a NULL pointer to ast_sockaddr_parse()
+ from ast_parse_arg(). According to the documentation in config.h,
+ "result pointer to the result. NULL is valid here, and can be
+ used to perform only the validity checks." This patch fixes the
+ segfault by checking for a NULL pointer. This patch also adds
+ documentation to netsock2.h about why it is necessary to check
+ for a NULL pointer. (Closes issue ASTERISK-20006) Reported by:
+ Michael L. Young Tested by: Michael L. Young Patches:
+ asterisk-20006-netsock-null-ptr.diff uploaded by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/1990/
+ ........ Merged revisions 369108 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * addons/chan_ooh323.c, /: Merged revisions 369091 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10 ........
+ r369091 | may | 2012-06-19 18:32:06 -0500 (Tue, 19 Jun 2012) | 9
+ lines check rtptimeouts in ooh323 channels as per config file
+ (rtp voice, video, udptl except rtcp) (closes issue
+ ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches:
+ 19179-ooh323-ast10.patch ........
+
+ * /, channels/chan_sip.c: Merged revisions 369067 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369067 | mmichelson | 2012-06-19 10:37:37 -0500
+ (Tue, 19 Jun 2012) | 17 lines Fix request routing issue when
+ outboundproxy is used. Asterisk was incorrectly setting the
+ destination of CANCELs and ACKs for error responses to the URI of
+ the initial INVITE. This resulted in further requests, such as
+ INVITEs with authentication credentials, to be routed
+ incorrectly. Instead, when these CANCEL or ACKs are to be sent,
+ we should simply keep the destination the same as what it
+ previously was. There is no need to alter it any. (closes issue
+ ASTERISK-20008) Reported by Marcus Hunger Patches:
+ ASTERISK-20008.patch uploaded by Mark Michelson (license #5049)
+ ........ Merged revisions 369066 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/features.c, /: Merged revisions 369044 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369044 | rmudgett | 2012-06-18 13:11:30 -0500
+ (Mon, 18 Jun 2012) | 12 lines Fix monitoring calls put in a
+ parking lot. * Fix a regression that was introduced by -r366167
+ which effectively disabled monitoring parked calls. (closes issue
+ ASTERISK-20012) Reported by: sdolloff Tested by: rmudgett
+ ........ Merged revisions 369043 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-06-15 16:30 +0000 [r369026] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c, /: Fix voicemail API tests by using the
+ correct argument order for create/destroy. ........ Merged
+ revisions 369024 from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+2012-06-15 16:25 +0000 [r369023] Automerge script <automerge@asterisk.org>
+
+ * main/translate.c, channels/vcodecs.c,
+ channels/sip/security_events.c, main/jitterbuf.c,
+ main/autochan.c, pbx/dundi-parser.c, main/aoc.c, main/cel.c,
+ main/enum.c, channels/iax2-parser.c, main/fskmodem.c,
+ main/config.c, channels/misdn_config.c, main/netsock.c,
+ build_tools/find_missing_support_level (added), main/loader.c,
+ main/ulaw.c, main/dial.c, channels/sig_analog.c, main/srv.c,
+ main/heap.c, main/privacy.c, channels/misdn/ie.c, res/ais/evt.c,
+ main/syslog.c, res/snmp/agent.c, main/event.c, main/astmm.c,
+ channels/sip/config_parser.c, channels/vgrabbers.c, main/db.c,
+ main/udptl.c, main/lock.c, channels/sip/sdp_crypto.c,
+ main/stun.c, main/frame.c, channels/sip/srtp.c,
+ main/threadstorage.c, channels/console_video.c,
+ channels/iax2-provision.c, main/xml.c, main/astfd.c,
+ main/taskprocessor.c, utils/astdb2bdb.c,
+ apps/confbridge/conf_config_parser.c, main/channel.c, main/cdr.c,
+ res/ael/pval.c, channels/chan_misdn.c, main/framehook.c,
+ main/tdd.c, main/strcompat.c, channels/console_gui.c,
+ channels/sip/dialplan_functions.c, main/fixedjitterbuf.c,
+ main/callerid.c, main/file.c, main/app.c,
+ main/stdtime/localtime.c, main/dns.c, main/message.c,
+ main/datastore.c, main/sched.c, main/timing.c, main/netsock2.c,
+ main/fskmodem_float.c, /, main/slinfactory.c, main/acl.c,
+ channels/sip/reqresp_parser.c, channels/sig_pri.c,
+ channels/misdn/isdn_lib.c, main/term.c, main/io.c,
+ main/hashtab.c, main/format_cap.c, main/abstract_jb.c,
+ main/fskmodem_int.c, main/logger.c, main/audiohook.c,
+ main/bridging.c, main/dsp.c, main/global_datastores.c,
+ main/autoservice.c, main/dnsmgr.c, main/security_events.c,
+ main/say.c, main/utils.c, channels/misdn/isdn_msg_parser.c,
+ utils/astdb2sqlite3.c, main/devicestate.c, main/ssl.c,
+ main/format_pref.c, main/astobj2.c, main/indications.c,
+ main/chanvars.c, main/cli.c, main/tcptls.c, main/data.c,
+ main/plc.c, main/test.c, channels/console_board.c,
+ channels/misdn/portinfo.c, main/image.c, main/alaw.c,
+ channels/sig_ss7.c, main/asterisk.c, main/xmldoc.c,
+ main/format.c, main/strings.c, main/pbx.c, main/rtp_engine.c,
+ main/ccss.c, res/ais/clm.c: Merged revisions 369005 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369005 | kpfleming | 2012-06-15 11:07:08 -0500
+ (Fri, 15 Jun 2012) | 22 lines Multiple revisions 369001-369002
+ ........ r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15
+ Jun 2012) | 11 lines Add support-level indications to many more
+ source files. Since we now have tools that scan through the
+ source tree looking for files with specific support levels, we
+ need to ensure that every file that is a component of a 'core' or
+ 'extended' module (or the main Asterisk binary) is explicitly
+ marked with its support level. This patch adds support-level
+ indications to many more source files in tree, but avoids adding
+ them to third-party libraries that are included in the tree and
+ to source files that don't end up involved in Asterisk itself.
+ ........ r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15
+ Jun 2012) | 3 lines Add a script to enable finding source files
+ without support-levels defined. ........ Merged revisions
+ 369001-369002 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-06-15 15:32 +0000 [r368963-368999] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.exports.in, /: Remove some symbol exports that
+ got missed in the removal of global symbols. (issue AST-807)
+ (issue AST-901) (issue AST-908) ........ Merged revisions 368998
+ from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+ * apps/app_voicemail.c, /: These functions that were moved need to
+ be static. Also wrap test functions in a #ifdef. (issue AST-807)
+ (issue AST-901) (issue AST-908) ........ Merged revisions 368964
+ from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+ * tests/test_voicemail_api.c, main/app.c,
+ include/asterisk/app_voicemail.h, apps/app_voicemail.c,
+ include/asterisk/app.h, /: Remove global symbol requirement from
+ app_voicemail. This uses the existing "function installation"
+ stuff that already existed for other functions, like getting
+ message counts. (closes issue AST-807) (issue AST-901) (issue
+ AST-908) Review: https://reviewboard.asterisk.org/r/1965/
+ ........ Merged revisions 368962 from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+2012-06-14 18:20 +0000 [r368872-368960] Automerge script <automerge@asterisk.org>
+
+ * /, channels/chan_skinny.c: Merged revisions 368947 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10 ........
+ r368947 | mjordan | 2012-06-14 12:31:33 -0500 (Thu, 14 Jun 2012)
+ | 21 lines AST-2012-009: Fix crash in chan_skinny due to Key Pad
+ Button Message handling AST-2012-008 (r367844) fixed a denial of
+ service attack exploitable in the Skinny channel driver that
+ occurred when certain messages are sent after a previously
+ registered station sends an Off Hook message. Unresolved in that
+ patch is an issue in the Asterisk 10 releases, wherein, if a
+ Station Key Pad Button Message is processed after an Off Hook
+ message, the channel driver will inappropriately dereference a
+ NULL pointer. This patch fixes those places where the message
+ handling or the channel callback functions would attempt to
+ dereference the line's pointer to the device. (issue
+ ASTERISK-19905) Reported by: Christoph Hebeisen Tested by:
+ mjordan, Christoph Hebeisen Patches: AST-2012-009-10.diff
+ uploaded by mjordan (license 6283) ........
+
+ * /, main/Makefile: Merged revisions 368928 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r368928 | mmichelson | 2012-06-14 10:25:23 -0500
+ (Thu, 14 Jun 2012) | 10 lines Revert Makefile change to remove
+ embedding res_adsi.so The change has resulted in a linking error
+ for certain versions of GCC. This is much worse than the original
+ issue, so for now, temporarily revert the change. A more thorough
+ change will be sought out. ........ Merged revisions 368927 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * res/res_adsi.c, res/res_smdi.c, /, funcs/func_volume.c: Merged
+ revisions 368895,368899 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r368895 | mjordan | 2012-06-13 15:27:28 -0500
+ (Wed, 13 Jun 2012) | 21 lines Mark res_smdi/res_adsi as 'core'
+ supported modules Recently, various issues surrounding weak
+ attributes have caused problems with modules that rely on that
+ feature to be enabled in menuselect. This includes app_voicemail
+ and chan_dahdi, as they both rely upon res_smdi and res_adsi,
+ which, in certain circumstances, may not be enabled by default in
+ menuselect. Because res_smdi/res_adsi are dependencies for
+ chan_dahdi/app_voicemail, this patch marks both as 'core'
+ supported modules. This will allow both app_voicemail and
+ chan_dahdi to be enabled as well, regardless of whether or not
+ that system supports weak attributes. (issue AST-900) Reported
+ by: Thomas Arimont (issue AST-885) Reported by: Denis Alberto
+ Martinez ........ Merged revisions 368894 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r368899 | mmichelson | 2012-06-13 16:13:30 -0500
+ (Wed, 13 Jun 2012) | 19 lines Fix a deadlock that occurs when
+ func_volume is used on a local channel. This was discovered by
+ trying to perform a call forward to an extension that makes use
+ of func_volume. When the local channel is optimized away, the
+ datastore on the local;2 channel would have its audiohook
+ destroyed rather than detaching the audiohook from the channel
+ and then destroying it. With this patch, func_volume's datastore
+ destructor takes the proper route of detaching the audiohook and
+ then destroying it. (closes issue ASTERISK-19611) reported by
+ Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark
+ Michelson (license #5049) ........ Merged revisions 368898 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, main/Makefile: Merged revisions 368885 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r368885 | mmichelson | 2012-06-13 14:36:39 -0500
+ (Wed, 13 Jun 2012) | 11 lines Remove forced linking of res_adsi.o
+ In GCC 4.5+ the result is that Asterisk has a phantom module
+ loaded at startup, claiming to be res_adsi. (closes issue
+ ASTERISK-19920) reported by Leif Madsen ........ Merged revisions
+ 368873 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * Makefile, /: Merged revisions 368831,368853 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r368831 | mjordan | 2012-06-12 13:30:06 -0500
+ (Tue, 12 Jun 2012) | 24 lines Do not perform install on existing
+ directories If a directory already exists, performing a 'make
+ install' will remove the permissions associated with the current
+ directory and replace them with the permissions of the user
+ executing the install. This patch changes this behavior to only
+ perform an install on the directory if the directory does not
+ exist. Thus, if a user later changes the permissions on that
+ directory, those permissions will be preserved in subsequent
+ installs. Review: https://reviewboard.asterisk.org/r/1986 Review:
+ https://reviewboard.asterisk.org/r/1864 (closes issue
+ ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger,
+ Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded
+ by mjordan) ........ Merged revisions 368830 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r368853 | mjordan | 2012-06-13 09:30:34 -0500
+ (Wed, 13 Jun 2012) | 11 lines Do not install empty directories;
+ add ASTLIBDIR r368830 modified the installation script to only
+ create a directory if that directory does not exist. If some
+ directory variable was empty, it would attempt to create the
+ empty location. It also failed to create the ASTLIBDIR directory.
+ This patch fixes it such that the correct directories are made
+ and only created if a value specifying them actually exists.
+ ........ Merged revisions 368852 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-06-12 16:22 +0000 [r368810-368826] Jason Parker <jparker@digium.com>
+
+ * /: Let's fix the 1.8-merged prop, to give automerge the best
+ chance at succeeding.
+
+ * funcs/func_strings.c, channels/sip/reqresp_parser.c,
+ include/asterisk/channel.h, apps/app_queue.c,
+ channels/chan_iax2.c, main/loader.c, main/channel.c,
+ channels/chan_dahdi.c, channels/sig_analog.c,
+ res/res_config_odbc.c, channels/sip/dialplan_functions.c,
+ apps/app_directory.c, pbx/pbx_config.c, main/md5.c,
+ res/res_odbc.c, res/res_speech.c, apps/app_voicemail.c,
+ main/udptl.c, channels/sip/sdp_crypto.c, channels/chan_sip.c, /,
+ res/res_fax.c, main/say.c: Multiple revisions
+ 368721,368739,368760,368808 ........ r368721 | kmoore |
+ 2012-06-11 09:11:14 -0500 (Mon, 11 Jun 2012) | 8 lines Fix
+ compilation in dev-mode Backport a compilation fix in md5.c from
+ trunk that only showed up in dev-mode under certain compiler
+ versions. ........ Merged revisions 368719 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368739 | kmoore | 2012-06-11 10:15:07 -0500 (Mon, 11 Jun 2012) |
+ 10 lines Fix coverity UNUSED_VALUE findings in core support level
+ files Most of these were just saving returned values without
+ using them and in some cases the variable being saved to could be
+ removed as well. (issue ASTERISK-19672) ........ Merged revisions
+ 368738 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r368760 | rmudgett | 2012-06-11 12:08:50 -0500 (Mon, 11
+ Jun 2012) | 17 lines Fix deadlock potential with
+ ast_set_hangupsource() calls. Calling ast_set_hangupsource() with
+ the channel lock held can result in a deadlock because the
+ function also locks the bridged channel. (issue ASTERISK-19537)
+ (closes issue AST-891) Reported by: Guenther Kelleter Tested by:
+ Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec
+ Davis ........ Merged revisions 368759 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368808 | mmichelson | 2012-06-12 10:37:38 -0500 (Tue, 12 Jun
+ 2012) | 15 lines Set the Caller ID "tag" on peers even if remote
+ party information is present. On incoming calls, we were setting
+ the cid_tag on the dialog only if there was no remote party
+ information (Remote-Party-ID or P-Asserted-Identity) present. The
+ Caller ID tag is an invented parameter, though, and should be set
+ no matter the circumstance. (closes issue ASTERISK-19859)
+ Reported by Thomas Arimont (closes issue AST-884) Reported by
+ Trey Blancher ........ Merged revisions 368807 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368721,368739,368760,368808 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /: Let's try using an automerge-propname, since we have multiple
+ heads.
+
+ * /: enable automerge
+
+2012-07-10 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.6.0-digiumphones Released.
+
+2012-07-06 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.6.0-digiumphones-rc2 Released.
+
+ * AST-2012-009: Skinny Channel Driver Remote Crash Vulnerability
+
+ * AST-2012-010: Possible Resource Leak on Uncompleted Re-INVITE
+ Transactions
+
+ * AST-2012-011: Remote Crash Vulnerability in VoiceMail Application
+
+ * Fix crash on a guest directmedia call
+
+ A sip_pvt may not have relatedpeer set if a call doesn't match up
+ with a peer. If there is no relatedpeer, there is no direct media
+ ACL to apply, so just return that is is allowed.
+
+ (closes issue ASTERISK-20040)
+
+ * Fix request routing issue when outboundproxy is used
+
+ Asterisk was incorrectly setting the destination of CANCELs and ACKs
+ for error responses to the URI of the initial INVITE. This resulted
+ in further requests, such as INVITEs with authentication
+ credentials, to be routed incorrectly. Instead when these CANCEL or
+ ACKs are to be esnt, we should simply keep the destination the same
+ as what it previously was. There is no need to alter it any.
+
+ (closes issue ASTERISK-20008)
+
+ * Fix monitoring calls put in a parking lot
+
+ Fix a regression that was introduced by r366167 which effectively
+ disabled monitoring parked calls.
+
+ (closes issue ASTERISK-20012)
+
+2012-06-12 Asterisk Development Team <asteriskteam@digium.com>
+
+ * 10.6.0-digiumphones-rc1 Released.
+
+2012-06-12 14:03 +0000 [r368791-368792] Matthew Jordan <mjordan@digium.com>
+
+ * /: Update merge property info
+
+ * channels/chan_sip.c: Fix deadlock in SIP transfers that involve a
+ REFER request In r367163, "send to voicemail" functionality was
+ added to the SIP channel driver. This required updating the party
+ redirecting information for the channel based on the headers
+ provided in the REFER request. When the redirecting party
+ information is updated on the channel, a call to
+ ast_indicate_data occurs. Because handle_request_refer still had
+ the sip_pvt locked, a deadlock could occur between the pbx_thread
+ and the do_monitor thread servicing the REFER request. This patch
+ preserves the proper locking order between the channel and the
+ sip_pvt by ensuring that the sip_pvt is unlocked prior to
+ updating the party redirecting information on the channel.
+ (closes issue AST-903) Reported by: Matt Jordan patches:
+ jira_ast_903_trunk.patch by rmudgett (license 5621)
+
+2012-06-11 22:49 +0000 [r368781-368783] Jason Parker <jparker@digium.com>
+
+ * /: Fix merge prop.
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/chan_sip.c: Multiple revisions 368629,368645 ........
+ r368629 | mmichelson | 2012-06-06 14:18:20 -0500 (Wed, 06 Jun
+ 2012) | 31 lines Fix a specific scenario where ACKs are not
+ matched. If a dialog-starting INVITE contains a to-tag, then
+ Asterisk will respond with a 481. In this case, the resulting
+ incoming ACK would not be matched, so Asterisk would continue
+ retransmitting the 481 until the transaction times out. There
+ were two issues. Asterisk, upon creating a sip_pvt would generate
+ a local tag. However, when the time came to transmit the 481,
+ since there was a to-tag in the INVITE, Asterisk would place this
+ original to-tag in the 481 response. When the ACK came in,
+ Asterisk would attempt to match the to-tag in the ACK to the
+ generated local tag. Unfortunately, Asterisk never actually
+ transmitted a response with the generated local tag, so the
+ to-tag in the ACK would not match. The other problem was that
+ when the 481 was sent, nothing was set on the sip_pvt to indicate
+ what CSeq is expected in the ACK. To fix the first problem, we
+ zero out the to-tag seen in the incoming INVITE. This way,
+ Asterisk, when time to send a response, will send its generated
+ local tag instead. To fix the second problem, we set the
+ sip_pvt's pendinginvite to the CSeq of the INVITE when we send a
+ 481. (closes issue ASTERISK-19892) Reported by Mark Michelson
+ ........ Merged revisions 368625 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368645 | rmudgett | 2012-06-06 16:32:09 -0500 (Wed, 06 Jun 2012)
+ | 17 lines Fix POTS flash hook to orignate a second call
+ deadlock. A deadlock can occur when a POTS phone tries to flash
+ hook to originate a second call for 3-way or transfer. If another
+ process is scanning the channels container when the POTS line
+ flash hooks then a deadlock will occur. * Release the channel and
+ private locks when creating a new channel as a result of a flash
+ hook. (closes issue ASTERISK-19842) Reported by: rmudgett Tested
+ by: rmudgett ........ Merged revisions 368644 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368629,368645 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/app_record.c, include/asterisk/channel.h,
+ res/res_calendar_caldav.c, pbx/dundi-parser.c,
+ apps/app_followme.c, main/cel.c, apps/app_queue.c, main/enum.c,
+ channels/iax2-parser.c, res/res_calendar_ews.c, main/config.c,
+ main/editline/tokenizer.c, channels/chan_dahdi.c,
+ channels/sig_analog.c, main/editline/readline.c, main/event.c,
+ channels/sip/config_parser.c, res/ael/ael.flex,
+ apps/app_chanspy.c, apps/app_stack.c, res/res_odbc.c,
+ res/res_calendar.c, channels/chan_sip.c, channels/chan_agent.c,
+ funcs/func_math.c, channels/iax2-provision.c, UPGRADE.txt,
+ addons/ooh323c/src/h323/H323-MESSAGES.h, channels/chan_iax2.c,
+ res/res_monitor.c, main/channel.c, addons/ooh323c/src/ooh323.c,
+ main/cdr.c, res/ael/pval.c, main/manager.c, main/app.c,
+ pbx/pbx_dundi.c, addons/ooh323c/src/h323/H323-MESSAGESEnc.c,
+ addons/ooh323c/src/ooq931.c, main/netsock2.c,
+ res/res_rtp_asterisk.c, apps/app_meetme.c, /,
+ channels/sip/reqresp_parser.c, main/acl.c, res/res_musiconhold.c,
+ include/asterisk/tcptls.h, channels/sig_pri.c, res/res_srtp.c,
+ res/res_config_odbc.c, funcs/func_odbc.c, funcs/func_cdr.c,
+ funcs/func_channel.c, apps/app_minivm.c, main/features.c,
+ apps/app_confbridge.c, codecs/codec_dahdi.c, pbx/pbx_config.c,
+ apps/app_voicemail.c, apps/app_dial.c, funcs/func_speex.c,
+ res/res_calendar_exchange.c, funcs/func_dialgroup.c,
+ apps/app_page.c, include/asterisk/cel.h, main/say.c,
+ funcs/func_lock.c, apps/app_disa.c, main/devicestate.c, CHANGES,
+ res/res_jabber.c, main/editline/term.c, main/cli.c,
+ main/tcptls.c, main/data.c, channels/chan_skinny.c,
+ funcs/func_aes.c, tests/test_config.c, funcs/func_devstate.c,
+ channels/sip/include/sip.h, channels/sig_ss7.c, main/asterisk.c,
+ main/xmldoc.c, res/res_calendar_icalendar.c, main/pbx.c,
+ channels/chan_local.c, addons/format_mp3.c: Multiple revisions
+ 365155,365160,365299,365320,365399,365475,365478,365575,365632,365701,365898,365990,366049,366053,366106,366168,366241,366297,366390,366412,366591,366598,366741,366792,366881,366884,366948,367003,367028,367267,367299,367369,367417,367470,367562,367679,367731,367782,367844,367907,367978,367981,368042,368093,368267,368310,368407,368470,368499,368524,368536,368568,368587
+ ........ r365155 | may | 2012-05-03 09:27:00 -0500 (Thu, 03 May
+ 2012) | 11 lines Fix coverity static analysis warning, allocate
+ full ie structure instead of without data buffer (close issue
+ ASTERISK-19674) Reported by: Matt Jordan Patches:
+ ASTERISK-19674.patch (License #5415) ........ Merged revisions
+ 365143 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r365160 | may | 2012-05-03 10:01:14 -0500 (Thu, 03 May
+ 2012) | 11 lines Fix warning of Coverity Static analysis, change
+ H225ProtocolIdentifier from value to pointer per functions that
+ use this. (close issue ASTERISK-19670) Reported by: Matt Jordan
+ Patches: ASTERISK-19670.patch (License #5415) ........ Merged
+ revisions 365159 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365299 | mmichelson | 2012-05-04 10:51:04 -0500 (Fri, 04 May
+ 2012) | 12 lines Fix core FINDING 2, FINDING 3, and FINDING 4
+ from Coverity's CONSTANT_EXPRESSION_RESULT report. These three
+ all are in RTP code that attempts to print the number of sequence
+ number cycles in an RTCP RR report. The code was masking out the
+ upper 16 bits and then shifting the number right by 16 bits. This
+ led to an all zero result in all cases. The fix is to do the
+ shift without the bit masking. (issue ASTERISK-19649) ........
+ Merged revisions 365298 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365320 | rmudgett | 2012-05-04 11:28:06 -0500 (Fri, 04 May 2012)
+ | 30 lines Fix local channel chains optimizing themselves out of
+ a call. * Made chan_local.c:check_bridge() check the return value
+ of ast_channel_masquerade(). In long chains of local channels,
+ the masquerade occasionally fails to get setup because there is
+ another masquerade already setup on an adjacent local channel in
+ the chain. * Made the outgoing local channel (the ;2 channel)
+ flush one voice or video frame per optimization attempt. * Made
+ sure that the outgoing local channel also does not have any
+ frames in its queue before the masquerade. * Made do the
+ masquerade immediately to minimize the chance that the outgoing
+ channel queue does not get any new frames added and thus
+ unconditionally flushed. * Made block indication -1 (Stop tones)
+ event when the local channel is going to optimize itself out.
+ When the call is answered, a chain of local channels pass down a
+ -1 indication for each bridge. This blizzard of -1 events really
+ slows down the optimization process. (closes issue
+ ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec
+ Davis Review: https://reviewboard.asterisk.org/r/1894/ ........
+ Merged revisions 365313 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365399 | kmoore | 2012-05-04 17:15:05 -0500 (Fri, 04 May 2012) |
+ 13 lines Fix many issues from the NULL_RETURNS Coverity report
+ Most of the changes here are trivial NULL checks. There are a
+ couple optimizations to remove the need to check for NULL and
+ outboundproxy parsing in chan_sip.c was rewritten to avoid use of
+ strtok. Additionally, a bug was found and fixed with the parsing
+ of outboundproxy when "outboundproxy=," was set. (Closes issue
+ ASTERISK-19654) ........ Merged revisions 365398 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365475 | mjordan | 2012-05-07 13:39:10 -0500 (Mon, 07 May 2012)
+ | 20 lines Support VoiceMail d() option when extension does not
+ exist in channel's context The VoiceMail d([c]) option is
+ documented to accept digits for a new extension in context <c>,
+ if played during the greeting. This option works fine if the
+ extension being redirected to has an extension with the same
+ initial digit in the channel's current context. If that digit did
+ not happen to exist in some extension, a dialplan match would
+ fail and the user would not be redirected. This patch fixes it
+ such that if the <c> option is used, the extensions are matched
+ in that context as opposed to the caller's original context.
+ (closes issue ASTERISK-18243) Reported by: mjordan Tested by:
+ mjordan Review: https://reviewboard.asterisk.org/r/1892 ........
+ Merged revisions 365474 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365478 | rmudgett | 2012-05-07 13:43:08 -0500 (Mon, 07 May 2012)
+ | 5 lines Fix type punned compiler warning in test_config.c
+ ........ Merged revisions 365476 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365575 | mmichelson | 2012-05-08 10:51:13 -0500 (Tue, 08 May
+ 2012) | 22 lines Send more accurate identification information in
+ dialog-info SIP NOTIFYs. This uses the calling channel's caller
+ ID and connected line information to populate the remote and
+ local identities in the dialog-info NOTIFY when an extension is
+ ringing. There is a bit of an oddity here, and that is that we
+ seed the remote target with the To header of the outbound call
+ rather than the from header. This is because it was reported that
+ seeding with the from header caused hints to be broken with
+ certain SNOM devices. A comment has been added to the code to
+ explain this. (closes issue ASTERISK-16735) reported by Maciej
+ Krajewski patches: local_remote_hint2.diff uploaded by Mark
+ Michelson (license #5049) 16735_tweak1.diff uploaded by Mark
+ Michelson (license #5049) Tested by Niccolo Belli ........ Merged
+ revisions 365574 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365632 | rmudgett | 2012-05-08 13:08:01 -0500 (Tue, 08 May 2012)
+ | 13 lines * Fix accept/decline DTMF buffer overwrite in
+ FollowMe. * Made use MAX_YN_STRING define to make all
+ accept/decline DTMF buffers the same size. Just using 20 isn't
+ good enough when someone didn't get the memo. * Fix stupid use of
+ a global variable in FollowMe. (ynlongest) * Fix bit field
+ declarations in FollowMe. ........ Merged revisions 365631 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365701 | rmudgett | 2012-05-08 15:25:08 -0500 (Tue, 08 May 2012)
+ | 12 lines * Fix FollowMe memory leak on error paths in
+ app_exec(). * Fix FollowMe leaving recorded caller name file on
+ error paths in app_exec(). * Use correct buffer dimension define
+ in struct call_followme.moh[] and struct fm_args.namerecloc[].
+ This fixes unexpected namerecloc filename length restriction.
+ ........ Merged revisions 365692 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365898 | mmichelson | 2012-05-09 11:15:28 -0500 (Wed, 09 May
+ 2012) | 29 lines Prevent sip_pvt refleak when an ast_channel
+ outlasts its corresponding sip_pvt. chan_sip was coded under the
+ assumption that a SIP dialog with an owner channel will always be
+ destroyed after the owner channel has been hung up. However,
+ there are situations where the SIP dialog can time out and auto
+ destruct before the corresponding channel has hung up. A typical
+ example of this would be if the 'h' extension in the dialplan
+ takes a long time to complete. In such cases,
+ __sip_autodestruct() would complain about the dialog being auto
+ destroyed with an owner channel still in place. The problem is
+ that even once the owner channel was hung up, the sip_pvt would
+ still be linked in its ao2_container because nothing would ever
+ unlink it. The fix for this is that if __sip_autodestruct() is
+ called for a sip_pvt that still has an owner channel in place,
+ the destruction is rescheduled for 10 seconds in the future. This
+ will continue until the owner channel is finally hung up. (closes
+ issue ASTERISK-19425) reported by David Cunningham Patches:
+ ASTERISK-19425.patch uploaded by Mark Michelson (License #5049)
+ (closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by
+ Dean Vesvuio ........ Merged revisions 365896 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r365990 | jrose | 2012-05-09 14:12:32 -0500 (Wed, 09 May 2012) |
+ 18 lines Block on frameout if the hardware has enough samples to
+ complete a frame. Fixes some problems with skipping audio in
+ elaborate scenarios involving multiple codecs by making
+ codec_dahdi operate in a more synchronous fashion similar to
+ codec_g729. This change also fixes the use of file conversion
+ tools from Asterisk's CLI. This change may cause the thread
+ responsible for transcoding audio to block briefly (Shaun Ruffell
+ describes this as 'several milliseconds') while waiting for the
+ hardware transcoder. (closes issue ASTERISK-19643) reported by:
+ Shaun Ruffell Patches:
+ 0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch
+ uploaded by Shaun Ruffell (license 5417) ........ Merged
+ revisions 365989 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r366049 | jrose | 2012-05-10 10:43:06 -0500 (Thu, 10 May 2012) |
+ 9 lines Coverity Report: Fix issues for error type UNINIT in Core
+ supported modules (issue ASTERISK-19652) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/1909/ ........ Merged
+ revisions 366048 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r366053 | mmichelson | 2012-05-10 11:13:06 -0500 (Thu, 10 May
+ 2012) | 9 lines Close the proper tcptls_session when session
+ creation fails. (issue AST-998) Reported by: Thomas Arimont
+ Tested by: Thomas Arimont ........ Merged revisions 366052 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r366106 | jrose | 2012-05-10 11:55:22 -0500 (Thu, 10 May 2012) |
+ 9 lines Coverity Report: Fix issues for error type CHECKED_RETURN
+ for core (issue ASTERISK-19658) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1905/ ........ Merged
+ revisions 366094 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r366168 | kmoore | 2012-05-10 15:54:08 -0500 (Thu, 10 May 2012) |
+ 13 lines Resolve FORWARD_NULL static analysis warnings This
+ resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7,
+ 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90,
+ 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and
+ 29 were already resolved. Those skipped were either
+ extended/deprecated or in areas of code that shouldn't be
+ disturbed. (Closes issue ASTERISK-19650) ........ Merged
+ revisions 366167 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r366241 | rmudgett | 2012-05-10 18:42:43 -0500 (Thu, 10 May 2012)
+ | 7 lines * Made ast_change_name() hold the channels container
+ lock while changing the channel name. * Eliminate redundant list
+ not empty check in clone_variables(). ........ Merged revisions
+ 366240 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r366297 | russell | 2012-05-11 18:59:35 -0500 (Fri, 11
+ May 2012) | 19 lines format_mp3: Fix a possible crash mp3_read().
+ This patch fixes a potential crash in mp3_read() by not assuming
+ that dbuf has enough data to finish filling up the output buffer.
+ The patch also makes sure that the dbuf state gets reset after we
+ know we read everything out of it already. In passing, this patch
+ includes some other cleanups of this module, including stripping
+ trailing whitespace, formatting fixes based on coding guidelines,
+ and removing a number of unused members from the private state
+ struct. (closes issue ASTERISK-19761) Reported by: Chris
+ Maciejewsk Tested by: Chris Maciejewsk ........ Merged revisions
+ 366296 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r366390 | mmichelson | 2012-05-14 14:16:36 -0500 (Mon,
+ 14 May 2012) | 25 lines Fix broken reinvite glare scenario. To
+ make a long story short, reinvite glares were broken because
+ Asterisk would invert the To and From headers when ACKing a 491
+ response. The reason was because the initreq of the dialog was
+ being changed to the incoming glared reinvite instead of being
+ set to the outgoing glared reinvite. This change has three parts
+ * In handle_incoming, we never will reject an ACK because it has
+ a to-tag present, even if we think the request may be out of
+ dialog. * In handle_request_invite, we do not change the initreq
+ when receiving a reinvite to which we will respond with a 491. *
+ In handle_request_invite, several superflous settings up
+ pendinginvite have been removed since this is dones automatically
+ by transmit_response_reliable Review:
+ https://reviewboard.asterisk.org/r/1911 ........ Merged revisions
+ 366389 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r366412 | mmichelson | 2012-05-14 15:06:58 -0500 (Mon,
+ 14 May 2012) | 19 lines Fix two more coverity constant expression
+ result findings. These correspond to findings 0 and 1 in the core
+ findings of ASTERISK-19649. After contacting Mark Spencer, he was
+ unsure of what the intent behind these lines of code were, so
+ they are being axed. For Asterisk 1.8 and 10, the output of
+ debugging DUNDi frames will not be changed, but for trunk the
+ "Retry" portion will be omitted since it does not properly
+ distinguish retransmissions from initial frames. (closes issue
+ ASTERISK-19649) Reported by Matthew Jordan ........ Merged
+ revisions 366409 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r366591 | jrose | 2012-05-15 15:44:59 -0500 (Tue, 15 May 2012) |
+ 15 lines chan_sip: Check the right channel's host address for
+ directmediapermit/deny Prior to this patch, when checking the
+ addresses for directmediapermit and denydirectmediadeny, Asterisk
+ would check the host address of the channel permit/deny was
+ specified, which defers from the expectations of both our users
+ and the development team. Instead, directmediapermit/deny now
+ checks against the address of the channel that the peer with the
+ ACL is connected to. (issue AST-876) Review:
+ https://reviewboard.asterisk.org/r/1899/ ........ Merged
+ revisions 366547 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r366598 | mmichelson | 2012-05-15 18:39:06 -0500 (Tue, 15 May
+ 2012) | 8 lines Correct misuse of ast_strip_quoted() when getting
+ a Diversion header's reason parameter. The use here was assuming
+ that the pointer would be updated, but the updated string is
+ actually returned by ast_strip_quoted() instead. ........ Merged
+ revisions 366597 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r366741 | mjordan | 2012-05-17 07:57:30 -0500 (Thu, 17 May 2012)
+ | 23 lines Fix checking bounds of array index after using it;
+ improper sizeof This patch fixes two problems pointed out by a
+ static analysis tool. * In chan_dahdi, when an event is handled
+ the index of the sub channel is first obtained. In very off
+ nominal cases, the method that determines the index can return a
+ negative value. In the event handling code, whether or not the
+ index returned is valid was being checked after that value was
+ used to index into an array. This patch makes it so the value is
+ checked before any indexing is done. * In res_calendar_ews,
+ sizeof was being passed a pointer instead of the struct to
+ determine the amount of memory to allocate. (issue
+ ASTERISK-19651) Reported by: Matt Jordan (closes issue
+ ASTERISK-19671) Reported by: Matt Jordan ........ Merged
+ revisions 366740 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r366792 | jrose | 2012-05-17 09:41:13 -0500 (Thu, 17 May 2012) |
+ 10 lines chan_sip: Fix missed locking of opposing pvt for
+ directmedia acl from r366547 It also required deadlock avoidance
+ since two sip_pvts structs needed to be locked simultaneously.
+ Trunk handles it differently, so this is a 1.8 and 10 patch only.
+ ........ (issue AST-876) Merged revisions 366791 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r366881 | mjordan | 2012-05-18 09:01:56 -0500 (Fri, 18 May 2012)
+ | 65 lines Fix a variety of memory leaks This patch addresses a
+ number of memory leaks in a variety of modules that were found by
+ a static analysis tool. A brief summary of the changes: *
+ app_minivm: free ast_str objects on off nominal paths * app_page:
+ free the ast_dial object if the requested channel technology
+ cannot be appended to the dialing structure * app_queue: if a
+ penalty rule failed to match any existing rule list names, the
+ created rule would not be inserted and its memory would be leaked
+ * app_read: dispose of the created silence detector in the
+ presence of off nominal circumstances * app_voicemail: dispose of
+ an allocated unique ID field for MWI event un-subscribe requests
+ in off nominal paths; dispose of configuration objects when using
+ the secret.conf option * chan_dahdi: dispose of the allocated
+ frame produced by ast_dsp_process * chan_iax2: properly unref
+ peer in CLI command "iax2 unregister" * chan_sip: dispose of the
+ allocated frame produced by sip_rtp_read's call of
+ ast_dsp_process; free memory in parse unit tests *
+ func_dialgroup: properly deref ao2 object grhead in nominal path
+ of dialgroup_read * func_odbc: free resultset in off nominal
+ paths of odbc_read * cli: free match_list in off nominal paths of
+ CLI match completion * config: free comment_buffer/list_buffer
+ when configuration file load is unchanged; free the same buffers
+ any time they were created and config files were processed *
+ data: free XML nodes in various places * enum: free context
+ buffer in off nominal paths * features: free ast_call_feature in
+ off nominal paths of applicationmap config processing * netsock2:
+ users of ast_sockaddr_resolve pass in an ast_sockaddr struct that
+ is allocated by the method. Failures in ast_sockaddr_resolve
+ could result in the users of the method not knowing whether or
+ not the buffer was allocated. The method will now not allocate
+ the ast_sockaddr struct if it will return failure. * pbx: cleanup
+ hash table traversals in off nominal paths; free ignore pattern
+ buffer if it already exists for the specified context * xmldoc:
+ cleanup various nodes when we no longer need them *
+ main/editline: various cleanup of pointers not being freed before
+ being assigned to other memory, cleanup along off nominal paths *
+ menuselect/mxml: cleanup of value buffer for an attribute when
+ that attribute did not specify a value * res_calendar*: responses
+ are allocated via the various *_request method returns and should
+ not be allocated in the various write_event methods; ensure
+ attendee buffer is freed if no data exists in the parsed node;
+ ensure that calendar objects are de-ref'd appropriately *
+ res_jabber: free buffer in off nominal path * res_musiconhold:
+ close the DIR* object in off nominal paths * res_rtp_asterisk: if
+ we run out of ports, close the rtp socket object and free the rtp
+ object * res_srtp: if we fail to create the session in libsrtp,
+ destroy the temporary ast_srtp object (issue ASTERISK-19665)
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1922 ........ Merged revisions
+ 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r366884 | kmoore | 2012-05-18 09:18:47 -0500 (Fri, 18
+ May 2012) | 9 lines Reorder and renumber tests appropriately It
+ appears that a patch did not apply properly when adding tests 12
+ and 13 and test 11 was duplicated. These tests have been
+ reordered and renumbered such that they make sense. ........
+ Merged revisions 366882 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r366948 | mjordan | 2012-05-18 10:45:42 -0500 (Fri, 18 May 2012)
+ | 20 lines Fix more memory leaks This patch adds to what was
+ fixed in r366880. Specifically, it addresses the following: *
+ chan_sip: dispose of an allocated frame in off nominal code paths
+ in sip_rtp_read * func_odbc: when disposing of an allocated
+ resultset, ensure that any rows that were appended to that
+ resultset are also disposed of * cli: free the created return
+ string buffer in another off nominal code path (issue
+ ASTERISK-19665) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1922/ ........ Merged
+ revisions 366944 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r367003 | mmichelson | 2012-05-18 12:00:14 -0500 (Fri, 18 May
+ 2012) | 19 lines Fix memory leak of SSL_CTX structures in TLS
+ core. SSL_CTX structures were allocated but never freed. This was
+ a bigger issue for clients than servers since new SSL_CTX
+ structures could be allocated for each connection. Servers, on
+ the other hand, typically set up a single SSL_CTX for their
+ lifetime. This is solved in two ways: 1. In __ssl_setup(), if a
+ tcptls_cfg has an ssl_ctx on it, it is freed so that a new one
+ can take its place. 2. A companion to ast_ssl_setup() called
+ ast_ssl_teardown() has been added so that servers can properly
+ free their SSL_CTXs. (issue ASTERISK-19278) ........ Merged
+ revisions 367002 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r367028 | mmichelson | 2012-05-18 12:50:18 -0500 (Fri, 18 May
+ 2012) | 18 lines Address MISSING_BREAK static analysis reports
+ some more. This addresses core findings 4 and 6. Moises Silva
+ helped me by stating that a break could be safely added to the
+ case where it is added in chan_dahdi.c In say.c, I have added a
+ comment indicating that static analysis complains but that it is
+ currently unknown if this is correct. This fixes all core
+ findings of this type. (closes issue ASTERISK-19662) reported by
+ Matthew Jordan ........ Merged revisions 367027 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r367267 | twilson | 2012-05-22 11:17:46 -0500 (Tue, 22 May 2012)
+ | 14 lines Resolve crash in subscribing for MWI notifications
+ ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the
+ variable should definitely not be used after that. To solve this
+ in the two cases that affect subscribing for MWI notifications,
+ we instead save the ref locally, and unref them in the error
+ conditions. (closes issue ASTERISK-19827) Reported by: B. R
+ Review: https://reviewboard.asterisk.org/r/1940/ ........ Merged
+ revisions 367266 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r367299 | twilson | 2012-05-22 12:21:51 -0500 (Tue, 22 May 2012)
+ | 21 lines Fix race condition for CEL LINKEDID_END event This
+ patch fixes to situations that could cause the CEL LINKEDID_END
+ event to be missed. 1) During a core stop gracefully, modules are
+ unloaded when ast_active_channels == 0. The LINKDEDID_END event
+ fires during the channel destructor. This means that
+ occasionally, the cel_* module will be unloaded before the
+ channel is destroyed. It seemed generally useful to wait until
+ the refcount of all channels == 0 before unloading, so I added a
+ channel counter and used it in the shutdown code. 2) During a
+ masquerade, ast_channel_change_linkedid is called. It calls
+ ast_cel_check_retire_linkedid which unrefs the linkedid in the
+ linkedids container in cel.c. It didn't ref the new linkedid. Now
+ it does. Review: https://reviewboard.asterisk.org/r/1900/
+ ........ Merged revisions 367292 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r367369 | mjordan | 2012-05-23 08:25:04 -0500 (Wed, 23 May 2012)
+ | 26 lines Re-add LastMsgsSent value for SIP peers Previously,
+ MWI logic utilized a counter called 'lastmsgssent' to know
+ whether or not MWI NOTIFY requests had been sent to a specific
+ peer. When MWI notifications were changed to use the internal
+ event framework, this value was no longer needed for its original
+ purpose. Hence, it was no longer updated with the new/old message
+ counts for a peer. The value was previously removed for Asterisk
+ 10; however, since it was still present in Asterisk 1.8 and still
+ useful for reporting purposes, it was decided to re-add the
+ value. This patch re-adds the 'LastMsgsSent' field in the
+ response to an AMI/CLI 'sip show peer [peer]' command, and makes
+ it so that the value of lastmsgssent is updated appropriately.
+ The value should now display the new/old message counts for a
+ particular peer. (closes issue ASTERISK-17866) Reported by: Steve
+ Davies patches by: ast-17866-rb1272.patch (License #5041 by
+ irroot) Modified slightly for this commit Review:
+ https://reviewboard.asterisk.org/r/1939 ........ Merged revisions
+ 367362 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r367417 | mmichelson | 2012-05-23 15:29:03 -0500 (Wed,
+ 23 May 2012) | 7 lines Only call SSL_CTX_free if DO_SSL is
+ defined. Thanks to Paul Belanger for pointing out this error.
+ ........ Merged revisions 367416 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r367470 | rmudgett | 2012-05-23 18:16:49 -0500 (Wed, 23 May 2012)
+ | 9 lines Fix WaitExten(x,m(musicclass)) string termination. The
+ AST_CONTROL_HOLD MOH class from the WaitExten application can now
+ be queued onto a channel, passed over local channels with the /m
+ option, and passed over IAX channels. ........ Merged revisions
+ 367469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r367562 | mjordan | 2012-05-24 08:32:33 -0500 (Thu, 24
+ May 2012) | 24 lines Fix crash in ConfBridge when user
+ announcement is played for more than 2 users A patch introduced
+ in r354938 made it so that ConfBridge would not attempt to play
+ sound files if those files did not exist. Unfortunately,
+ ConfBridge uses the same underlying function, play_sound_helper,
+ to playback both sound files and numbers to callers. When a
+ number is being played back, the name of the sound file is
+ expected to be NULL. This NULL value was passed into a function
+ that tested for the existance of a sound file and is not tolerant
+ to NULL file names, causing a crash. This patch fixes the
+ behavior, such that if a sound file does not exist we do not
+ attempt to play it, but we only attempt that check if the a sound
+ file was specified in the first place. If a sound file was not
+ specified, we use the 'play number' logic in the helper function.
+ (closes issue ASTERISK-19899) Reported by: Florian Gilcher Tested
+ by: Florian Gilcher patches: asterisk-19899.diff uploaded by
+ mjordan (license 6283) ........ r367679 | rmudgett | 2012-05-24
+ 17:29:23 -0500 (Thu, 24 May 2012) | 34 lines Fix Dial I option
+ ignored if dial forked and one fork redirects. The Dial and Queue
+ I option is intended to block connected line updates and
+ redirecting updates. However, it is a feature that when a call is
+ locally redirected, the I option is disabled if the redirected
+ call runs as a local channel so the administrator can have an
+ opportunity to setup new connected line information.
+ Unfortunately, the Dial and Queue I option is disabled for *all*
+ forked calls if one of those calls is redirected. * Make the Dial
+ and Queue I option apply to each outgoing call leg independently.
+ Now if one outgoing call leg is locally redirected, the other
+ outgoing calls are not affected. * Made Dial not pass any
+ redirecting updates when forking calls. Redirecting updates do
+ not make sense for this scenario. * Made Queue not pass any
+ redirecting updates when using the ringall strategy. Redirecting
+ updates do not make sense for this scenario. * Fixed deadlock
+ potential with chan_local when Dial and Queue send redirecting
+ updates for a local redirect. * Converted the Queue stillgoing
+ flag to a boolean bitfield. (closes issue ASTERISK-19511)
+ Reported by: rmudgett Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1920/ ........ Merged
+ revisions 367678 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r367731 | elguero | 2012-05-24 21:29:26 -0500 (Thu, 24 May 2012)
+ | 20 lines Fix pvt_sip for inbound call to use peer's
+ allowtransfer setting The pvt_sip allowtransfer was not being set
+ to that of the peer's setting. Therefore, the global
+ allowtransfer setting was being used instead which would lead to
+ calls not being transfered if the global setting was set to 'no'
+ despite the setting on the peer being 'yes' and vice versa, calls
+ would be allowed to transfer even if the peer's setting was 'no'
+ but the global setting was 'yes'. (Closes issue ASTERISK-19856)
+ Reported by: Jacek Tested by: Michael L. Young, Jacek Patches:
+ issue-asterisk-19856-branch10-v3.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/1923/ ........ Merged
+ revisions 367730 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r367782 | rmudgett | 2012-05-25 11:30:55 -0500 (Fri, 25 May 2012)
+ | 18 lines AST-2012-007: Fix IAX receiving HOLD without suggested
+ MOH class crash. * Made schedule_delivery() set the received
+ frame f->data.ptr to NULL if the datalen is zero. * Fix
+ queue_signalling() memcpy() size error. * Made queue_signalling()
+ not use C++ keyword variable names. (closes issue ASTERISK-19597)
+ Reported by: mgrobecker Patches: jira_asterisk_19597_v1.8.patch
+ (license #5621) patch uploaded by rmudgett Tested by: rmudgett,
+ Michael L. Young ........ Merged revisions 367781 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r367844 | mjordan | 2012-05-29 13:33:20 -0500 (Tue, 29 May 2012)
+ | 21 lines AST-2012-008: Fix remote crash vulnerability in
+ chan_skinny When a skinny session is unregistered, the
+ corresponding device pointer is set to NULL in the channel
+ private data. If the client was not in the on-hook state at the
+ time the connection was closed, the device pointer can later be
+ dereferened if a message or channel event attempts to use a
+ line's pointer to said device. The patches prevent this from
+ occurring by checking the line's pointer in message handlers and
+ channel callbacks that can fire after an unregistration attempt.
+ (closes issue ASTERISK-19905) Reported by: Christoph Hebeisen
+ Tested by: mjordan, Damien Wedhorn Patches: AST-2012-008-1.8.diff
+ uploaded by mjordan (license 6283) AST-2012-008-10.diff uploaded
+ by mjordan (licesen 6283) ........ r367907 | rmudgett |
+ 2012-05-29 17:28:55 -0500 (Tue, 29 May 2012) | 17 lines Coverity
+ Report: Fix issues for error type REVERSE_INULL (deprecated
+ modules) * Fix only issue pointed out by
+ deprecated_REVERSE_INULL.txt for app_meetme.c in find_user(). *
+ Change use of %i to %d in sscanf() in find_user(). The use of %i
+ gives unexpected parsing because it can accept hex, octal, and
+ decimal integer formats. * Changed other uses of %i in
+ app_meetme() to use %d for consistency. (issue ASTERISK-19648)
+ Reported by: Matt Jordan ........ Merged revisions 367906 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r367978 | rmudgett | 2012-05-30 12:39:24 -0500 (Wed, 30 May 2012)
+ | 19 lines Fix deadlock when executing CLI "pri show channels"
+ and "ss7 show channels" commands. * Fix sig_pri_lock_owner() to
+ avoid deadlock properly. * Code pri_grab() better. * Fix
+ sig_ss7_lock_owner() to avoid deadlock properly. * Code
+ ss7_grab() better. (closes issue ASTERISK-19854) Reported by:
+ Jaxon Patches: jira_asterisk_19854_v1.8.6.patch (license #5621)
+ patch uploaded by rmudgett (Modified to do the same thing to
+ sig_ss7) Tested by: Jaxon ........ Merged revisions 367976 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r367981 | rmudgett | 2012-05-30 13:07:28 -0500 (Wed, 30 May 2012)
+ | 7 lines Use the DEADLOCK_AVOIDANCE() macro instead. (issue
+ ASTERISK-19854) ........ Merged revisions 367980 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368042 | rmudgett | 2012-05-31 13:20:15 -0500 (Thu, 31 May 2012)
+ | 10 lines Coverity Report: Fix issues for error type
+ REVERSE_INULL (core modules) * Fixes findings:
+ 0-2,5,7-15,24-26,28-31 (issue ASTERISK-19648) Reported by: Matt
+ Jordan ........ Merged revisions 368039 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368093 | elguero | 2012-05-31 22:28:09 -0500 (Thu, 31 May 2012)
+ | 17 lines Add documentation to function CHANNEL for options
+ echocan_mode and buffers The ability to set "echocan_mode" and
+ "buffers" through the dialplan was added to chan_dahdi some time
+ ago. This patch adds some documentation to func_channel. (Closes
+ issue ASTERISK-19911) Reported by: Dale Noll Tested by: Michael
+ L. Young Patches: asterisk-19911-branch18.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/1949/ ........ Merged
+ revisions 368092 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368267 | kpfleming | 2012-06-01 15:22:44 -0500 (Fri, 01 Jun
+ 2012) | 20 lines Improve SDP parsing warning messages *
+ 'Unsupported media type' is only reported when that is in fact
+ the case, not when a supported media type is included in an 'm'
+ line that has an invalid format. * All warning messages related
+ to parsing 'm' lines now include the 'm' line contents. * (minor
+ bugfix) newline added to port-number-zero warning messages. *
+ Warning messages improved to use RFC-specified terminology for
+ various items. * Warnings for offers that include more than one
+ port for a single media type now include the media type. Review:
+ https://reviewboard.asterisk.org/r/1811/ ........ Merged
+ revisions 368218 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368310 | rmudgett | 2012-06-01 18:24:25 -0500 (Fri, 01 Jun 2012)
+ | 15 lines Fix deadlock when Gosub used with alternate dialplan
+ switches. Attempting to remove a channel from autoservice with
+ the channel lock held will result in deadlock. * Restructured
+ gosub_exec() to not call ast_parseable_goto() and
+ ast_exists_extension() with the channel lock held. (closes issue
+ ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett
+ ........ Merged revisions 368308 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368407 | rmudgett | 2012-06-04 14:08:52 -0500 (Mon, 04 Jun 2012)
+ | 23 lines Fix potential deadlock between masquerade and
+ chan_local. * Restructure ast_do_masquerade() to not hold channel
+ locks while it calls ast_indicate(). * Simplify many calls to
+ ast_do_masquerade() since it will never return a failure now. If
+ it does fail internally because a channel driver callback
+ operation failed, the only thing ast_do_masquerade() can do is
+ generate a warning message about strange things may happen and
+ press on. * Fixed the call to ast_bridged_channel() in
+ ast_do_masquerade(). This change fixes half of the deadlock
+ reported in ASTERISK-19801 between masquerades and chan_iax.
+ (closes issue ASTERISK-19537) Reported by: rmudgett Tested by:
+ rmudgett Review: https://reviewboard.asterisk.org/r/1915/
+ ........ Merged revisions 368405 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368470 | rmudgett | 2012-06-04 16:11:42 -0500 (Mon, 04 Jun 2012)
+ | 10 lines Document BLINDTRANSFER behavior change. (issue
+ ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call
+ ........ Merged revisions 368469 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368499 | mmichelson | 2012-06-04 17:02:26 -0500 (Mon, 04 Jun
+ 2012) | 16 lines Relay proper SIP responses on calling side.
+ Revision 351130 broke corect HANGUPCAUSE setting for the 404 case
+ in chan_sip. Other cases were also potentially broken. This patch
+ fixes the relaying of causes to be what they used to be. (closes
+ issue ASTERISK-19914) Reported by Pavel Troller Tested by Walter
+ Doekes (via a reviewboard test to be committed later) Patches:
+ chan_sip.diff uploaded by Pavel Troller (license #6302) ........
+ Merged revisions 368498 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368524 | kmoore | 2012-06-05 10:19:58 -0500 (Tue, 05 Jun 2012) |
+ 11 lines Ensure that pages and emails are sent using
+ RFC822-compliant date format When localization was added to
+ app_voicemail, these headers were altered when they should have
+ remained in en_US format for RFC compliance. This reverts the
+ changes to those two lines. (closes issue ASTERISK-19876)
+ ........ Merged revisions 368520 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368536 | kmoore | 2012-06-05 10:27:01 -0500 (Tue, 05 Jun 2012) |
+ 8 lines Resolve some build warnings My newly upgraded compiler
+ caught these usages of uninitialized values. They weren't
+ actually used. ........ Merged revisions 368533 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368568 | rmudgett | 2012-06-05 20:10:10 -0500 (Tue, 05 Jun 2012)
+ | 15 lines Fix parked call performing a DTMF blind transfer after
+ being retrieved. When a parked call was retrieved from the
+ parking lot, it could not do a blind transfer because it caused
+ the involved calls to be hung up unconditionally. * Made the
+ ParkedCall application return the ast_bridge_call() return value.
+ (closes issue ABE-2862) Reported by: Vlad Povorozniuc ........
+ Merged revisions 368567 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368587 | kmoore | 2012-06-06 11:09:10 -0500 (Wed, 06 Jun 2012) |
+ 12 lines Ensure overlapping hold flags do not conflict When
+ changing between different modes of hold, the flags were not
+ being cleared out properly causing a failure to change hold
+ states. (closes issue ASTERISK-19919) Patch-by: Morten Tryfoss
+ Reported-by: Morten Tryfoss ........ Merged revisions 368586 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions
+ 365155,365160,365299,365320,365399,365475,365478,365575,365632,365701,365898,365990,366049,366053,366106,366168,366241,366297,366390,366412,366591,366598,366741,366792,366881,366884,366948,367003,367028,367267,367299,367369,367417,367470,367562,367679,367731,367782,367844,367907,367978,367981,368042,368093,368267,368310,368407,368470,368499,368524,368536,368568,368587
+ from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-06 17:35 +0000 [r368609] Matthew Jordan <mjordan@digium.com>
+
+ * /, build_tools/make_version: Add feature modifier to versions
+ produced from branches Certain branches, such as Certified
+ Asterisk, may have a modifier added to them that specifies the
+ features available in that branch. For branches, this modifier is
+ expected to be reflected in the location of the branch in
+ subversion. For example, a subversion of URL of
+ /certified/branches/1.8.11 would have a feature modifier of
+ 'certified'. This is slightly different then how features are
+ determined for tags, where the feature is part of the actual tag
+ name, e.g., "10.5.0-digiumphones". In keeping with the
+ nomenclature used for tags, the feature specifier for branches is
+ translated and placed after the revision numbers. For the example
+ given previously, this would result in a branch version of
+ "Asterisk SVN-branch-1.8.11-cert-rXXXXXX".
+
+2012-05-21 19:16 +0000 [r367162] Mark Michelson <mmichelson@digium.com>
+
+ * main/callerid.c, include/asterisk/callerid.h,
+ channels/chan_sip.c: Add "send to voicemail" Digium phone
+ functionality to Asterisk. This change accommodates two methods
+ by which calls can be directed to a user's voicemail. * Incoming
+ calls can be redirected to any user's voicemail. * Established
+ calls can be blind transferred to any user's voicemail. Digium
+ phones indicate the desire to direct a call to voicemail by using
+ a Diversion header with a reason parameter of "send_to_vm". This
+ patch adds the "send_to_vm" reason as a valid redirecting reason.
+ In addition, chan_sip.c has been modified to update redirecting
+ information on the transferred channel by reading a Diversion
+ header on a REFER request. (closes issue AST-871) Reported by
+ Malcolm Davenport Review: https://reviewboard.asterisk.org/r/1925
+
+2012-05-04 21:28 +0000 [r365396] Jason Parker <jparker@digium.com>
+
+ * apps/app_mixmonitor.c, apps/app_voicemail.c, /: Add support for
+ folders in MixMonitor 'm' option. Backport manager actions. The
+ manager actions are needed, so MixMonitor can be executed on
+ existing channels. (issue DPMA-68) ........ Merged revisions
+ 365395 from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+2012-05-03 20:54 +0000 [r365297] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_mixmonitor.c: Populate file extensions for mixmonitor
+ recordings properly.
+
+2012-05-03 20:06 +0000 [r365264] Jason Parker <jparker@digium.com>
+
+ * main/jitterbuf.c, configs/queues.conf.sample,
+ configs/usbradio.conf.sample (removed),
+ res/res_calendar_caldav.c, apps/rpt_flow.pdf (removed),
+ apps/app_queue.c, main/cel.c, res/res_config_sqlite.c,
+ res/res_calendar_ews.c, main/config.c, formats/format_siren7.c,
+ channels/chan_dahdi.c, formats/format_vox.c, funcs/func_volume.c,
+ configure, formats/format_h263.c, main/event.c,
+ apps/app_chanspy.c, formats/format_g719.c, channels/chan_sip.c,
+ funcs/func_env.c, channels/chan_agent.c, funcs/func_strings.c,
+ channels/console_video.c, Makefile.rules, main/astfd.c,
+ formats/format_wav_gsm.c, bridges/bridge_multiplexed.c,
+ channels/chan_iax2.c, funcs/func_global.c,
+ apps/confbridge/conf_config_parser.c, res/res_config_curl.c,
+ build_tools/cflags.xml, main/cdr.c, funcs/func_curl.c,
+ main/manager.c, main/tdd.c, channels/console_gui.c,
+ formats/format_pcm.c, main/app.c, main/stdtime/localtime.c,
+ utils/extconf.c, makeopts.in, main/message.c,
+ formats/format_gsm.c, res/res_clioriginate.c,
+ include/asterisk/time.h, res/res_rtp_asterisk.c,
+ res/res_config_pgsql.c, apps/app_meetme.c, /,
+ formats/format_wav.c, configure.ac, res/res_musiconhold.c,
+ channels/chan_gtalk.c, tests/test_linkedlists.c, apps/app_ices.c,
+ channels/sig_pri.c, res/res_srtp.c, formats/format_ilbc.c,
+ channels/sig_pri.h, Makefile, apps/app_forkcdr.c,
+ res/res_config_odbc.c, bridges/bridge_builtin_features.c,
+ codecs/gsm/src/k6opt.s, build_tools/menuselect-deps.in,
+ funcs/func_channel.c, apps/app_directed_pickup.c,
+ main/features.c, res/res_agi.c, main/http.c, main/logger.c,
+ apps/app_confbridge.c, apps/app_sms.c, main/audiohook.c,
+ formats/format_h264.c, apps/app_voicemail.c,
+ codecs/lpc10/Makefile, apps/app_dial.c, formats/format_sln.c,
+ codecs/gsm/Makefile, funcs/func_sysinfo.c,
+ formats/format_ogg_vorbis.c, CHANGES, main/astobj2.c,
+ main/format_pref.c, apps/app_speech_utils.c,
+ tests/test_security_events.c, main/tcptls.c,
+ addons/ooh323cDriver.c, formats/format_g723.c,
+ apps/app_externalivr.c, tests/test_config.c, tests/test_poll.c,
+ addons/chan_mobile.c, formats/format_siren14.c,
+ funcs/func_devstate.c, main/asterisk.c, main/xmldoc.c,
+ channels/chan_mgcp.c, formats/format_g729.c,
+ channels/chan_unistim.c, configs/chan_dahdi.conf.sample,
+ main/pbx.c, res/res_calendar_icalendar.c, channels/chan_local.c,
+ funcs/func_version.c, configs/rpt.conf.sample (removed): Multiple
+ revisions
+ 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083
+ ........ r361208 | jrose | 2012-04-04 14:30:09 -0500 (Wed, 04 Apr
+ 2012) | 10 lines Make 'help devstate change' display properly
+ (get rid of excess comma) (closes issue ASTERISK-19444) Reported
+ by: Makoto Dei Patches: devstate-change-usage-truncate.patch
+ uploaded by Makoto Dei (license 5027) ........ Merged revisions
+ 361201 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r361211 | jrose | 2012-04-04 15:00:23 -0500 (Wed, 04 Apr
+ 2012) | 12 lines Fix some stuff involving calls to memcpy and
+ memset The important parts of the patch were already applied
+ through other updates. (closes issue ASTERISK-19445) Reported by:
+ Makoto Dei Patches: memset-memcpy-length.patch uploaded by Makoto
+ Dei (license 5027) ........ Merged revisions 361210 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361270 | jrose | 2012-04-05 11:53:35 -0500 (Thu, 05 Apr 2012) |
+ 10 lines Fix MusicOnHold in MeetMe so that it always uses the
+ class if it's been defined There were a few instances of
+ restarting music on hold in meetme that would cause Asterisk to
+ revert to the default class of music on hold for no adequate
+ reason. Review: https://reviewboard.asterisk.org/r/1844/ ........
+ Merged revisions 361269 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361330 | kmoore | 2012-04-06 08:31:51 -0500 (Fri, 06 Apr 2012) |
+ 11 lines Remove unnecessary error message in app_dial.c The error
+ message for failure to stop autoservice after a gosub or macro
+ call during a dial was removed for macro while Asterisk 1.4 was
+ still being actively developed. The corresponding gosub error
+ message was never removed. (closes issue ASTERISK-19551) ........
+ Merged revisions 361329 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361333 | mjordan | 2012-04-06 09:01:33 -0500 (Fri, 06 Apr 2012)
+ | 11 lines Fix a typo in the warning messages for an ignored
+ media stream Added a '\n' to the warning messages when we ignore
+ a media stream due to the port number being '0'. (closes issue
+ ASTERISK-19646) Reported by: Badalian Vyacheslav ........ Merged
+ revisions 361332 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361381 | russell | 2012-04-06 10:49:19 -0500 (Fri, 06 Apr 2012)
+ | 5 lines Remove a few more files related to chan_usbradio and
+ app_rpt. ........ Merged revisions 361380 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361422 | pabelanger | 2012-04-06 11:31:18 -0500 (Fri, 06 Apr
+ 2012) | 14 lines Multiple revisions 361403,361412 ........
+ r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri, 06 Apr
+ 2012) | 2 lines Fix typo in svn:keywords ........ r361412 |
+ pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2
+ lines Fix typo in svn:keywords ........ Merged revisions
+ 361403,361412 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361472 | kmoore | 2012-04-06 13:13:04 -0500 (Fri, 06 Apr 2012) |
+ 5 lines Add missing newlines to CLI logging ........ Merged
+ revisions 361471 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361522 | rmudgett | 2012-04-06 14:47:29 -0500 (Fri, 06 Apr 2012)
+ | 8 lines Don't add an empty MESSAGE_DATA(key) header if it
+ doesn't already exist. Doing Set(MESSAGE_DATA(key)=) would add an
+ empty key header if the key header did not already exist. If it
+ already existed it would delete it. * Made msg_set_var_full()
+ exit early if the named variable did not already exist and the
+ value to set is empty. ........ r361560 | mjordan | 2012-04-06
+ 15:32:13 -0500 (Fri, 06 Apr 2012) | 13 lines Fix memory leak when
+ using MeetMeAdmin 'e' option with user specified A memory
+ leak/reference counting leak occurs if the MeetMeAdmin 'e'
+ command (eject last user that joined) is used in conjunction with
+ a specified user. Regardless of the command being executed, if a
+ user is specified for the command, MeetMeAdmin will look up that
+ user. Because the 'e' option kicks the last user that joined, as
+ opposed to the one specified, the reference to the user specified
+ by the command would be leaked when the user variable was
+ assigned to the last user that joined. ........ Merged revisions
+ 361558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r361607 | mjordan | 2012-04-06 17:00:11 -0500 (Fri, 06
+ Apr 2012) | 12 lines Fix memory leak in res_calendar_ews when
+ event email address node is empty If the XML calendar data
+ returned by a Microsoft Exchange Web Service specifies an XML
+ Event E-Mail Address ("EmailAddress"), and no e-mail address is
+ provided, a condition existed where an ast_calendar_attendee
+ struct would be allocated but not appended to the list of
+ attendees. Because of that, the memory associated with the
+ attendee would never be freed. This patch frees the memory if no
+ e-mail address is provided. ........ Merged revisions 361606 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361658 | mjordan | 2012-04-09 14:42:53 -0500 (Mon, 09 Apr 2012)
+ | 15 lines Change SHARED function to use a safe traversal when
+ modifying a variable When the SHARED function modifies a
+ variable, it removes it from its list of variables and reinserts
+ the new value at the head of the list of variables. Doing this
+ inside a standard list traversal can be dangerous, as the
+ standard list traversal does not account for the list being
+ changed. While the code in question should not cause a use after
+ free violation due to its breaking out of the loop after freeing
+ the variable, it could lead to a maintenance issue if the loop
+ was modified. This also fixes a violation reported by a static
+ analysis tool, which also makes this code easier to maintain in
+ the future. ........ Merged revisions 361657 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361706 | mjordan | 2012-04-09 15:54:55 -0500 (Mon, 09 Apr 2012)
+ | 17 lines Prevent invalid access of free'd memory if DAHDI
+ channel during an MWI event In the MWI processing loop, when a
+ valid event occurs the temporary caller ID information is
+ deallocated. If a new DAHDI channel is successfully created, the
+ event is passed up to the analog_ss_thread without error and the
+ loop exits. If, however, the DAHDI channel is not created, then
+ the caller ID struct has been free'd, and the gains reset to
+ their previous level. This will almost certainly cause an invalid
+ access to the free'd memory, either in subsequent calls to
+ callerid_free or calls to callerid_feed. This patch makes it so
+ that we only free the caller ID structure if a DAHDI channel is
+ successfully created, and we bump the gains back up if we fail to
+ make a DAHDI channel. ........ Merged revisions 361705 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361754 | mjordan | 2012-04-09 16:44:30 -0500 (Mon, 09 Apr 2012)
+ | 12 lines Allow func_curl to exit gracefully if list allocation
+ fails during write If the global_curl_info data structure could
+ not be allocated, the datastore associated with the operation
+ would be free'd, but the function would not return. This would
+ later dereference the datastore, almost certainly causing
+ Asterisk to crash. With this patch, if the data structure is not
+ allocated the method will return an error code, and not attempt
+ any further operation. ........ Merged revisions 361753 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361804 | mjordan | 2012-04-10 14:57:30 -0500 (Tue, 10 Apr 2012)
+ | 10 lines Fix crash caused by unloading or reloading of
+ res_http_post When unlinking itself from the registered HTTP
+ URIs, res_http_post could inadvertently free all URIs registered
+ with the HTTP server. This patch modifies the unregister method
+ to only free the URI that is actually being unregistered, as
+ opposed to all of them. ........ Merged revisions 361803 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361855 | rmudgett | 2012-04-10 16:47:42 -0500 (Tue, 10 Apr 2012)
+ | 19 lines Prevent invalid access of free'd memory if DAHDI
+ channel during an MWI event In the MWI processing loop, when a
+ valid event occurs the temporary caller ID information is
+ deallocated. If a new DAHDI channel is successfully created, the
+ event is passed up to the analog_ss_thread without error and the
+ loop exits. If, however, the DAHDI channel is not created, then
+ the caller ID struct has been free'd, and the gains reset to
+ their previous level. This will almost certainly cause an invalid
+ access to the free'd memory, either in subsequent calls to
+ callerid_free or calls to callerid_feed. * Rework the -r361705
+ patch to better manage the cs and mtd allocated resources. *
+ Fixed use of mwimonitoractive flag to be correct if the
+ mwi_thread() fails to start. ........ Merged revisions 361854
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361907 | jrose | 2012-04-11 11:07:50 -0500 (Wed, 11 Apr 2012) |
+ 10 lines Change default value of 'ignorebusy' on Queue members so
+ that behavior is more like 1.8 Prior to this patch, in order to
+ restore that behavior, a function would have to be used on the
+ QueueMember to make the ringinuse option do anything, which is
+ pretty unreasonable. (closes issue ASTERISK-19536) reported by:
+ Philippe Lindheimer Review:
+ https://reviewboard.asterisk.org/r/1860/ ........ r361956 |
+ kmoore | 2012-04-12 10:01:13 -0500 (Thu, 12 Apr 2012) | 13 lines
+ Simplify build system architecture optimization This change to
+ the build system rips out any usage of PROC along with
+ architecture-specific optimizations in favor of using
+ -march=native where it is supported. This fixes broken builds on
+ 64bit Intel systems and results in better optimized code on
+ systems running GCC 4.2+. Review:
+ https://reviewboard.asterisk.org/r/1852/ (closes issue
+ ASTERISK-19462) ........ Merged revisions 361955 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361981 | kmoore | 2012-04-12 11:22:28 -0500 (Thu, 12 Apr 2012) |
+ 12 lines Make trunkfreq take effect when set Previously, setting
+ trunkfreq had no effect on initial load or on reload and only
+ ever used the default value. This causes trunkfreq to be used
+ appropriately on initial load and reload. (closes issue
+ ASTERISK-19521) Patch-by: Jaco Kroon ........ Merged revisions
+ 361972 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r362080 | jrose | 2012-04-13 10:30:22 -0500 (Fri, 13 Apr
+ 2012) | 10 lines Send relative path named recordings to the
+ meetme directory instead of sounds Prior to this patch, no effort
+ was made to parse the path name to determine a proper destination
+ for recordings of MeetMe's r option. This fixes that. Review:
+ https://reviewboard.asterisk.org/r/1846/ ........ Merged
+ revisions 362079 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362084 | jrose | 2012-04-13 11:04:22 -0500 (Fri, 13 Apr 2012) |
+ 15 lines Make ForkCDR e option not set end time of the newly
+ forked CDR log Prior to this patch, ForkCDR's e option would
+ immediately set the end time of the forked CDR to that of the CDR
+ that is being terminated. This resulted in the new CDR's end time
+ being roughly the same as it's beginning time (which is in turn
+ roughly the same as the original's end time). (closes issue
+ ASTERISK-19164) Reported by: Steve Davies Patches:
+ cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)
+ ........ Merged revisions 362082 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362152 | mjordan | 2012-04-16 14:39:32 -0500 (Mon, 16 Apr 2012)
+ | 19 lines Check for IO stream failures in various format's
+ truncate/seek operations For the formats that support seek and/or
+ truncate operations, many of the C library calls used to
+ determine or set the current position indicator in the file
+ stream were not being checked. In some situations, if an error
+ occurred, a negative value would be returned from the library
+ call. This could then be interpreted inappropriately as
+ positional data. This patch checks the return values from these
+ library calls before using them in subsequent operations. (issue
+ ASTERISK-19655) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1863/ ........ Merged
+ revisions 362151 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362202 | mjordan | 2012-04-16 16:40:29 -0500 (Mon, 16 Apr 2012)
+ | 18 lines Fix handling of negative return code when storing
+ voicemails in ODBC storage When storing a voicemail message using
+ an ODBC connection to a database, the voicemail message is first
+ stored on disk. The sound file associated with the message is
+ read into memory before being transmitted to the database. When
+ this occurs, a failure in the C library's lseek function would
+ cause a negative value to be passed to the mmap as the size of
+ the memory map to create. This would almost certainly cause the
+ creation of the memory map to fail, resulting in the message
+ being lost. (issue ASTERISK-19655) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/1863 ........ Merged
+ revisions 362201 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362205 | mjordan | 2012-04-16 16:57:19 -0500 (Mon, 16 Apr 2012)
+ | 25 lines Fix negative return handling in channel drivers In
+ chan_agent, while handling a channel indicate, the agent channel
+ driver must obtain a lock on both the agent channel, as well as
+ the channel the agent channel is using. To do so, it attempts to
+ lock the other channel first, then unlock the agent channel which
+ is locked prior to entry into the indicate handler. If this
+ unlock fails with a negative return value, which can occur if the
+ object passed to agent_indicate is an invalid ao2 object or is
+ NULL, the return value is passed directly to strerror, which can
+ only accept positive integer values. In chan_dahdi, the return
+ value of dahdi_get_index is used to directly index into the
+ sub-channel array. If dahd_get_index returns a negative value, it
+ would use that value to index into the array, which could cause
+ an invalid memory access. If dahdi_get_index returns a negative
+ number, we now default to SUB_REAL. (issue ASTERISK-19655)
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1863/ ........ Merged
+ revisions 362204 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362264 | elguero | 2012-04-17 09:53:04 -0500 (Tue, 17 Apr 2012)
+ | 23 lines Turn off warning message when bind address is set to
+ any. When a bind address is set to an ANY address
+ (udpbindport=::), a warning message is displayed stating that
+ "Address remapping activated in sip.conf but we're using IPv6,
+ which doesn't need it. Please remove 'localnet' and/or
+ 'externaddr' settings." But if one is running dual stack, we
+ shouldn't be told to turn those settings off. This patch checks
+ if the bind address is an ANY address or not. The warning message
+ will now only be displayed if the bind address is NOT an ANY
+ address and IPv6 is being used. Also, updated the copyright year.
+ (closes issue ASTERISK-19456) Reported by: Michael L. Young
+ Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff
+ uploaded by Michael L. Young (license 5026) ........ Merged
+ revisions 362253 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362305 | mjordan | 2012-04-17 13:27:44 -0500 (Tue, 17 Apr 2012)
+ | 15 lines Fix error that caused seek format operations to set
+ max file size to '1' or '0' A very inappropriate placement of a
+ ')' (introduced in r362151) caused the maximum size of a file to
+ be set as the result of a comparison operation, as opposed to the
+ result of the ftello operation. This resulted in seeking being
+ restricted to the beginning of the file, or 1 byte into the file.
+ Thanks to the Asterisk Test Suite for properly freaking out about
+ this on at least one test. (issue ASTERISK-19655) Reported by:
+ Matt Jordan ........ Merged revisions 362304 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362356 | mjordan | 2012-04-17 15:56:05 -0500 (Tue, 17 Apr 2012)
+ | 17 lines Fix places where a negative return from ftello could
+ be used as invalid input In a variety of locations in both
+ reading and writing a file, the result from the C library
+ function ftello is used as input to other functions. For the
+ parameters and functions in question, a negative value is invalid
+ input. This patch checks the return value from the ftello
+ function to determine if we were able to determine the current
+ position in the file stream and, if not, fail gracefully. (issue
+ ASTERISK-19655) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1863/ ........ Merged
+ revisions 362355 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362357 | jrose | 2012-04-17 15:57:36 -0500 (Tue, 17 Apr 2012) |
+ 12 lines Make use of va_args more appropriate to form in various
+ res_config modules plus utils. A number of va_copy operations
+ weren't matched with a corresponding va_end in res_config_odbc.
+ Also, there was a potential for va_end to be invoked twice on the
+ same va_arg in utils, which would mean invoking va_end on an
+ undefined variable... which is bad. va_end is removed from
+ various functions in config_pgsql and config_curl since they
+ aren't making their own copy. The invokers of those functions are
+ responsible for calling va_end on them. (issue ASTERISK-19451)
+ Reported by: Walter Doekes Review:
+ https://reviewboard.asterisk.org/r/1848/ ........ Merged
+ revisions 362354 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362360 | mjordan | 2012-04-17 16:07:29 -0500 (Tue, 17 Apr 2012)
+ | 24 lines Fix places in main where a negative return value could
+ impact execution This patch addresses a number of modules in main
+ that did not handle the negative return value from function calls
+ adequately, or were not sufficiently clear that the conditions
+ leading to improper handling of the return values could not
+ occur. This includes: * asterisk.c: A negative return value from
+ the read function would be used directly as an index into a
+ buffer. We now check for success of the read function prior to
+ using its result as an index. * manager.c: Check for failures in
+ mkstemp and lseek when handling the temporary file created for
+ processing data returned from a CLI command in action_command.
+ Also check that the result of an lseek is sanitized prior to
+ using it as the size of a memory map to allocate. (issue
+ ASTERISK-19655) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1863/ ........ Merged
+ revisions 362359 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362364 | mjordan | 2012-04-17 16:11:25 -0500 (Tue, 17 Apr 2012)
+ | 29 lines Fix places in resources where a negative return value
+ could impact execution This patch addresses a number of modules
+ in resources that did not handle the negative return value from
+ function calls adequately. This includes: * res_agi.c: if the
+ result of the read function is a negative number, indicating some
+ failure, the result would instead be treated as the number of
+ bytes read. This patch now treats negative results in the same
+ manner as an end of file condition, with the exception that it
+ also logs the error code indicated by the return. *
+ res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor
+ to srcfd, and instead assigns a negative value, that file
+ descriptor could later be passed to functions that require a
+ valid file descriptor. If spawn_mp3 fails, we now immediately
+ retry instead of continuing in the logic. * res_rtp_asterisk.c:
+ if no codec can be matched between two RTP instances in a peer to
+ peer bridge, we immediately return instead of attempting to use
+ the codec payload type as an index to determine the appropriate
+ negotiated codec. (issue ASTERISK-19655) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged
+ revisions 362362 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362377 | mjordan | 2012-04-17 16:22:37 -0500 (Tue, 17 Apr 2012)
+ | 13 lines Handle case where an unknown format is used to get the
+ preferred codec size In ast_codec_pref_getsize, if an unknown
+ format is passed to the method, no preferred codec will be
+ selected and a negative number will be used to index into the
+ format list. The method now logs an unknown format as a warning,
+ and returns an empty format list. (issue ASTERISK-19655) Reported
+ by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/
+ ........ r362429 | rmudgett | 2012-04-18 11:27:51 -0500 (Wed, 18
+ Apr 2012) | 19 lines Add ability to ignore layer 1 alarms for BRI
+ PTMP lines. Several telcos bring the BRI PTMP layer 1 down when
+ the line is idle. When layer 1 goes down, Asterisk cannot make
+ outgoing calls. Incoming calls could fail as well because the
+ alarm processing is handled by a different code path than the
+ Q.931 messages. * Add the layer1_presence configuration option to
+ ignore layer 1 alarms when the telco brings layer 1 down. This
+ option can be configured by span while the similar DAHDI driver
+ teignorered=1 option is system wide. This option unlike
+ layer2_persistence does not require libpri v1.4.13 or newer.
+ Related to JIRA AST-598 JIRA ABE-2845 ........ Merged revisions
+ 362428 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r362496 | mjordan | 2012-04-18 21:27:08 -0500 (Wed, 18
+ Apr 2012) | 50 lines Fix a variety of potential buffer overflows
+ * chan_mobile: Fixed an overrun where the cind_state buffer (an
+ integer array of size 16) would be overrun due to improper bounds
+ checking. At worst, the buffer can be overrun by a total of 48
+ bytes (assuming 4-byte integers), which would still leave it
+ within the allocated memory of struct hfp. This would corrupt
+ other elements in that struct but not necessarily cause any
+ further issues. * app_sms: The array imsg is of size 250, while
+ the array (ud) that the data is copied into is of size 160. If
+ the size of the inbound message is greater then 160, up to 90
+ bytes could be overrun in ud. This would corrupt the user data
+ header (array udh) adjacent to ud. * chan_unistim: A number of
+ invalid memmoves are corrected. These would move data (which may
+ or may not be valid) into the ends of these buffers. * asterisk:
+ ast_console_toggle_loglevel does not check that the console log
+ level being set is less then or equal to the allowed log levels
+ of 32. * format_pref: In ast_codec_pref_prepend, if any
+ occurrence of the specified codec is not found, the value used to
+ index into the array pref->order would be one greater then the
+ maximum size of the array. * jitterbuf: If the element being
+ placed into the jitter buffer lands in the last available slot in
+ the jitter history buffer, the insertion sort attempts to move
+ the last entry in the buffer into one slot past the maximum
+ length of the buffer. Note that this occurred for both the min
+ and max jitter history buffers. * tdd: If a read from fsk_serial
+ returns a character that is greater then 32, an attempt to read
+ past one of the statically defined arrays containing the values
+ that character maps to would occur. * localtime: struct ast_time
+ and tm are not the same size - ast_time is larger, although it
+ contains the elements of tm within it in the same layout. Hence,
+ when using memcpy to copy the contents of tm into ast_time, the
+ size of tm should be used, as opposed to the size of ast_time. *
+ extconf: this treats ast_timing's minmask array as if it had a
+ length of 48, when it has defined the size of the array as 24.
+ pbx.h defines minmask as having a size of 48. (issue
+ ASTERISK-19668) Reported by: Matt Jordan ........ Merged
+ revisions 362485 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362537 | twilson | 2012-04-19 09:31:59 -0500 (Thu, 19 Apr 2012)
+ | 14 lines Handle multiple commands per connection via netconsole
+ Asterisk would accept multiple NULL-delimited CLI commands via
+ the netconsole socket, but would occasionally miss a command due
+ to the command not being completely read into the buffer. This
+ patch ensures that any partial commands get moved to the front of
+ the read buffer, appended to, and properly sent. (closes issue
+ ASTERISK-18308) Review: https://reviewboard.asterisk.org/r/1876/
+ ........ Merged revisions 362536 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362587 | seanbright | 2012-04-19 11:04:21 -0500 (Thu, 19 Apr
+ 2012) | 12 lines Prevent a crash in ExternalIVR when the 'S'
+ command is sent first. If the first command sent from an
+ ExternalIVR client is an 'S' command, we were blindly removing
+ the first element from the play list and deferencing it, even if
+ it was NULL. This corrects that and also locks appropriately in
+ one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski
+ ........ Merged revisions 362586 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362678 | rmudgett | 2012-04-19 16:00:21 -0500 (Thu, 19 Apr 2012)
+ | 5 lines Update membermacro and membergosub documentation in
+ queues.conf.sample. ........ Merged revisions 362677 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362681 | elguero | 2012-04-19 16:11:35 -0500 (Thu, 19 Apr 2012)
+ | 9 lines Add leading and trailing backslashes A couple of unit
+ tests did not have have leading or trailing backslashes when
+ setting their test category resulting in a warning message being
+ displayed. Added the backslash where needed. ........ Merged
+ revisions 362680 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362730 | wdoekes | 2012-04-19 16:59:43 -0500 (Thu, 19 Apr 2012)
+ | 5 lines Fix documentation for ${VERSION(ASTERISK_VERSION_NUM)}.
+ ........ Merged revisions 362729 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362816 | twilson | 2012-04-20 09:49:42 -0500 (Fri, 20 Apr 2012)
+ | 13 lines Document Speech* apps hangup on failure and suggest
+ TryExec The Speech API apps return -1 on failure, which will hang
+ up the channel. This may not be desirable behavior for some, but
+ it isn't something that can be changed without breaking people's
+ dialplans or writing an option to all of the Speech apps that
+ does what TryExec already does. This patch documents the hangup
+ behavior of the apps, and suggests TryExec as the solution.
+ (closes issue AST-813) ........ Merged revisions 362815 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362869 | twilson | 2012-04-20 11:12:34 -0500 (Fri, 20 Apr 2012)
+ | 11 lines OpenBSD doesn't have rawmemchr, use strchr (closes
+ issue ASTERISK-19758) Reported by: Barry Miller Tested by: Terry
+ Wilson Patches: 362758-diff uploaded by Barry Miller (license
+ 5434) ........ Merged revisions 362868 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362918 | elguero | 2012-04-20 11:47:51 -0500 (Fri, 20 Apr 2012)
+ | 11 lines Add missing payload type to events API The Security
+ Events Framework API was changed while adding the generation of
+ security events in chan_sip. A payload type and name was missed
+ from being added to struct ie_maps. (closes issue ASTERISK-19759)
+ Reported by: Michael L. Young Patches: issue-asterisk-19759.diff
+ uploaded by Michael L. Young (license 5026) ........ r362998 |
+ rmudgett | 2012-04-20 20:45:13 -0500 (Fri, 20 Apr 2012) | 5 lines
+ Update app_dial M and U option GOTO return value documentation.
+ ........ Merged revisions 362997 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363212 | tilghman | 2012-04-23 11:06:53 -0500 (Mon, 23 Apr 2012)
+ | 8 lines On some platforms, O_RDONLY is not a flag to be
+ checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX
+ specification does not mandate how these 3 flags must be
+ specified, only that one of the three must be specified in every
+ call. ........ Merged revisions 363209 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363376 | rmudgett | 2012-04-24 19:01:21 -0500 (Tue, 24 Apr 2012)
+ | 5 lines Hangup affected channel in error paths of
+ bridge_call_thread(). ........ Merged revisions 363375 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363429 | rmudgett | 2012-04-24 20:23:08 -0500 (Tue, 24 Apr 2012)
+ | 27 lines Fix recalled party B feature flags for a failed DTMF
+ atxfer. 1) B calls A with Dial option T 2) B DTMF atxfer to C 3)
+ B hangs up 4) C does not answer 5) B is called back 6) B answers
+ 7) B cannot initiate transfers anymore * Add dial features
+ datastore to recalled party B channel that is a copy of the
+ original party B channel's dial features datastore. * Extracted
+ add_features_datastore() from add_features_datastores(). *
+ Renamed struct ast_dial_features features_caller and
+ features_callee members to my_features and peer_features
+ respectively. These better names eliminate the need for some
+ explanatory comments. * Simplified code accessing the struct
+ ast_dial_features datastore. (closes issue ASTERISK-19383)
+ Reported by: lgfsantos ........ Merged revisions 363428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363688 | rmudgett | 2012-04-25 14:47:44 -0500 (Wed, 25 Apr 2012)
+ | 19 lines Clear ISDN channel resetting state if the peer
+ continues to use it. Some ISDN switches occasionally fail to send
+ a RESTART ACKNOWLEDGE in response to a RESTART request. * Made
+ the second SETUP received after sending a RESTART request clear
+ the channel resetting state as if the peer had sent the expected
+ RESTART ACKNOWLEDGE before continuing to process the SETUP. The
+ peer may not be sending the expected RESTART ACKNOWLEDGE. (issue
+ ASTERISK-19608) (issue AST-844) (issue AST-815) Patches:
+ jira_ast_815_v1.8.patch (license #5621) patch uploaded by
+ rmudgett (modified) ........ Merged revisions 363687 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363734 | rmudgett | 2012-04-25 15:48:22 -0500 (Wed, 25 Apr 2012)
+ | 18 lines Make DAHDISendCallreroutingFacility wait 5 seconds for
+ a reply before disconnecting the call. Some switches may not
+ handle the call-deflection/call-rerouting message if the call is
+ disconnected too soon after being sent. Asteisk was not waiting
+ for any reply before disconnecting the call. * Added a 5 second
+ delay before disconnecting the call to wait for a potential
+ response if the peer does not disconnect first. (closes issue
+ ASTERISK-19708) Reported by: mehdi Shirazi Patches:
+ jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett ........ Merged revisions 363730
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363789 | rmudgett | 2012-04-25 17:59:46 -0500 (Wed, 25 Apr 2012)
+ | 5 lines Update Pickup application documentation. ........
+ Merged revisions 363788 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363876 | rmudgett | 2012-04-25 22:11:45 -0500 (Wed, 25 Apr 2012)
+ | 5 lines Update Pickup application documentation. (Even better)
+ ........ Merged revisions 363875 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363935 | alecdavis | 2012-04-26 04:46:38 -0500 (Thu, 26 Apr
+ 2012) | 14 lines chan_sip: [general] maxforwards, not checked for
+ a value greater than 255 The peer maxforwards is checked for both
+ '< 1' and '> 255', but the default 'maxforwards' in the [general]
+ section is only checked for '< 1' alecdavis (license 585)
+ Reported by: alecdavis Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1888/ ........ Merged
+ revisions 363934 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363987 | kmoore | 2012-04-26 08:27:34 -0500 (Thu, 26 Apr 2012) |
+ 15 lines Fix reference leaks involving SIP Replaces transfers The
+ reference held for SIP blind transfers using the Replaces header
+ in an INVITE was never freed on success and also failed to be
+ freed in some error conditions. This caused a file descriptor
+ leak since the RTP structures in use at the time of the transfer
+ were never freed. This reference leak and another relating to
+ subscriptions in the same code path have now been corrected.
+ (Closes issue ASTERISK-19579) Reported by: Maciej Krajewski
+ Tested by: Maciej Karjewski ........ Merged revisions 363986 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364047 | twilson | 2012-04-26 14:30:55 -0500 (Thu, 26 Apr 2012)
+ | 8 lines Add more constness to the end_buf pointer in the
+ netconsole issue ASTERISK-18308 Review:
+ https://reviewboard.asterisk.org/r/1876/ ........ Merged
+ revisions 364046 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364065 | rmudgett | 2012-04-26 15:25:05 -0500 (Thu, 26 Apr 2012)
+ | 24 lines Fix DTMF atxfer running h exten after the wrong bridge
+ ends. When party B does an attended transfer of party A to party
+ C, the attending bridge between party B and C should not be
+ running an h exten when the bridge ends. Running an h exten now
+ sets a softhangup flag to ensure that an AGI will run in dead AGI
+ mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B
+ channel for the attending bridge between party B and C. (closes
+ issue AST-870) (closes issue ASTERISK-19717) Reported by: Mario
+ (closes issue ASTERISK-19633) Reported by: Andrey Solovyev
+ Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario
+ ........ Merged revisions 364060 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364109 | rmudgett | 2012-04-26 16:10:46 -0500 (Thu, 26 Apr 2012)
+ | 5 lines Update Pickup application documentation. (With feeling
+ this time.) ........ Merged revisions 364108 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364163 | schmidts | 2012-04-27 07:54:19 -0500 (Fri, 27 Apr 2012)
+ | 3 lines fix a wrong behavior of alarm timezones in caldav and
+ icalendar when an alarm doesnt use utc. This change uses the same
+ timezone from the start time. ........ r364204 | mjordan |
+ 2012-04-27 09:44:13 -0500 (Fri, 27 Apr 2012) | 23 lines Allow for
+ reloading SRTP crypto keys within the same SIP dialog As a
+ continuation of the patch in r356604, which allowed for the
+ reloading of SRTP keys in re-INVITE transfer scenarios, this
+ patch addresses the more common case where a new key is requested
+ within the context of a current SIP dialog. This can occur, for
+ example, when certain phones request a SIP hold. Previously, once
+ a dialog was associated with an SRTP object, any subsequent
+ attempt to process crypto keys in any SDP offer - either the
+ current one or a new offer in a new SIP request - were ignored.
+ This patch changes this behavior to only ignore subsequent crypto
+ keys within the current SDP offer, but allows future SDP offers
+ to change the keys. (issue ASTERISK-19253) Reported by: Thomas
+ Arimont Tested by: Thomas Arimont Review:
+ https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions
+ 364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r364259 | kmoore | 2012-04-27 13:58:34 -0500 (Fri, 27
+ Apr 2012) | 14 lines Allow SIP pvts involved in Replaces
+ transfers to fall out of reference sooner Unref the SIP pvt
+ stored in the refer structure as soon as it is no longer needed
+ so that the pvt and associated file descriptors can be freed
+ sooner. This change makes a reference decrement unnecessary in
+ code that handles SIP BYE/Also transfers which should not touch
+ the reference anyway. (Closes issue ASTERISK-19579) Reported by:
+ Maciej Krajewski Tested by: Maciej Krajewski ........ Merged
+ revisions 364258 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364285 | mjordan | 2012-04-27 14:30:19 -0500 (Fri, 27 Apr 2012)
+ | 43 lines Prevent overflow in calculation in ast_tvdiff_ms on
+ 32-bit machines The method ast_tvdiff_ms attempts to calculate
+ the difference, in milliseconds, between two timeval structs, and
+ return the difference in a 64-bit integer. Unfortunately, it
+ assumes that the long tv_sec/tv_usec members in the timeval
+ struct are large enough to hold the calculated values before it
+ returns. On 64-bit machines, this might be the case, as a long
+ may be 64-bits. On 32-bit machines, however, a long may be less
+ (32-bits), in which case, the calculation can overflow. This
+ overflow caused significant problems in MixMonitor, which uses
+ the method to determine if an audio factory, which has not
+ presented audio to an audiohook, is merely late in providing said
+ audio or will never provide audio. In an overflow situation, the
+ audiohook would incorrectly determine that an audio factory that
+ will never provide audio is merely late instead. This led to
+ situations where a MixMonitor never recorded any audio. Note that
+ this happened most frequently when that MixMonitor was started by
+ the ConfBridge application itself, or when the MixMonitor was
+ attached to a Local channel. (issue ASTERISK-19497) Reported by:
+ Ben Klang Tested by: Ben Klang Patches:
+ 32-bit-time-overflow-10-2012-04-26.diff (license #6283) by
+ mjordan (closes issue ASTERISK-19727) Reported by: Mark Murawski
+ Tested by: Michael L. Young Patches:
+ 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
+ (closes issue ASTERISK-19471) Reported by: feyfre Tested by:
+ feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review:
+ https://reviewboard.asterisk.org/r/1889/ ........ Merged
+ revisions 364277 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364342 | mmichelson | 2012-04-27 16:58:06 -0500 (Fri, 27 Apr
+ 2012) | 10 lines Don't attempt to make use of the
+ dynamic_exclude_static ACL if DNS lookup fails. (closes issue
+ ASTERISK-18321) Reported by Dan Lukes Patches:
+ ASTERISK-18321.patch by Mark Michelson (license #5049) ........
+ Merged revisions 364341 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364365 | twilson | 2012-04-27 17:31:01 -0500 (Fri, 27 Apr 2012)
+ | 11 lines Fix ast_parse_arg numeric type range checking and add
+ tests ast_parse_arg wasn't checking for strto* parse errors or
+ limiting the results by the actual range of the numeric types.
+ This patch fixes that and adds unit tests as well. Review:
+ https://reviewboard.asterisk.org/r/1879/ ........ Merged
+ revisions 364340 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012)
+ | 2 lines Add missing test_config.c ........ r364536 | elguero |
+ 2012-04-28 21:21:10 -0500 (Sat, 28 Apr 2012) | 13 lines Fix
+ configuring custom sound_leader_has_left in confbridge.conf The
+ configuration option to specify a custom sound_leader_has_left
+ file for a conference bridge was not being parsed. This patch
+ fixes it so that a custom sound file will now be used. (closes
+ issue ASTERISK-19771) Reported by: Pawel Kuzak Tested by: Pawel
+ Kuzak, Michael L. Young Patches: leaderhasleft_sound.dpatch
+ uploaded by Pawel Kuzak (license 6380) Review:
+ https://reviewboard.asterisk.org/r/1884/ ........ r364579 |
+ mjordan | 2012-04-29 14:43:53 -0500 (Sun, 29 Apr 2012) | 15 lines
+ Fix error that caused truncate operations to fail Another very
+ inappropriate placement of a ')' (again introduced in r362151)
+ caused the various truncate operations to attempt to truncate the
+ sound file at a position of '0'. (issue ASTERISK-19655) Reported
+ by: Matt Jordan (issue ASTERISK-19810) Reported by: colbec
+ ........ Merged revisions 364578 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364650 | markm | 2012-04-30 11:43:11 -0500 (Mon, 30 Apr 2012) |
+ 15 lines Merged revisions 364635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) |
+ 10 lines Sanatize result from bfd_find_nearest_line
+ (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file
+ to null resulting in a crash when strrchr(file) runs (closes
+ issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark
+ Murawski ........ ........ r364651 | may | 2012-04-30 11:48:57
+ -0500 (Mon, 30 Apr 2012) | 10 lines Fix use freed pointer in
+ return value from call thread (issue ASTERISK-19663) Reported by:
+ Matt Jordan Patches: ASTERISK-19663-ooh323.patch (License #5415)
+ ........ Merged revisions 364649 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364777 | jrose | 2012-05-01 13:23:08 -0500 (Tue, 01 May 2012) |
+ 13 lines Fix bad check in voicemail functions for
+ ast_inboxcount2_func Check looks for ast_inboxcount_func instead
+ of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes
+ issue ASTERISK-19718) Reported by: Corey Farrell Patches:
+ ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell
+ (license 5909) ........ Merged revisions 364769 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364787 | kmoore | 2012-05-01 14:07:09 -0500 (Tue, 01 May 2012) |
+ 12 lines Play conf-placeintoconf message to the correct channel
+ Correct the code in app_confbridge to play the conf-placeintoconf
+ message to the marked user entering the bridge instead of to the
+ conference while the marked user hears silence. (closes issue
+ ASTERISK-19641) Reported-by: Mark A Walters ........ Merged
+ revisions 364786 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364845 | rmudgett | 2012-05-01 16:50:32 -0500 (Tue, 01 May 2012)
+ | 7 lines * Fix error path resouce leak in local_request(). *
+ Restructure local_request() to reduce indentation. ........
+ Merged revisions 364840 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364900 | mmichelson | 2012-05-01 18:10:16 -0500 (Tue, 01 May
+ 2012) | 16 lines Fix Coverity-reported ARRAY_VS_SINGLETON error.
+ As it turned out, this wasn't a huge deal. We were calling
+ ast_app_parse_options() for a set of options of which none took
+ arguments. The proper thing to do for this case is to pass NULL
+ for the "args" parameter here. We were instead passing a
+ seemingly-randomly chosen char * from the function. While this
+ would never get written to, you can rest assured things would
+ have gotten bad had new options (which took arguments) been added
+ to func_volume. (closes issue ASTERISK-19656) ........ Merged
+ revisions 364899 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364903 | rmudgett | 2012-05-01 18:14:12 -0500 (Tue, 01 May 2012)
+ | 10 lines Fixed __ao2_ref() validating user_data twice. (closes
+ issue ASTERISK-19755) Reported by: Gunther Kelleter Patches:
+ ao2_ref.patch (license #6372) patch uploaded by Gunther Kelleter
+ ........ Merged revisions 364902 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364965 | mjordan | 2012-05-01 21:44:15 -0500 (Tue, 01 May 2012)
+ | 11 lines Only log a failure to get read/write samples from
+ factories if it didn't happen In audiohook_read_frame_both,
+ anytime samples are obtained from the read/write factories a
+ debug statement is logged stating that samples were not obtained
+ from the factories. This statement used to only occur if
+ option_debug was turned on and no samples were obtained; in some
+ refactoring when the option_debug statement was removed, the
+ "else" clause was removed as well. This patch makes it so that
+ those debug log statements only occur if the condition leading up
+ to them actually happened. ........ r365014 | elguero |
+ 2012-05-02 11:16:03 -0500 (Wed, 02 May 2012) | 18 lines Update
+ security events unit tests The security events framework API was
+ changed in Asterisk 10 but the unit tests were not updated at the
+ same time. This patch does the following: * Adds two more
+ security events that were added to the API * Add challenge,
+ received_challenge and received_hash in the inval_password
+ security event unit test (issue ASTERISK-19760) Reported by:
+ Michael L. Young Tested by: Michael L. Young Patches:
+ issue-asterisk-19760-branch10.diff uploaded by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/1877/
+ ........ r365083 | twilson | 2012-05-02 12:29:54 -0500 (Wed, 02
+ May 2012) | 33 lines Multiple revisions 365006,365068 ........
+ r365006 | twilson | 2012-05-02 10:49:03 -0500 (Wed, 02 May 2012)
+ | 12 lines Fix a CEL LINKEDID_END race and local channel
+ linkedids This patch has the ;2 channel inherit the linkedid of
+ the ;1 channel and fixes the race condition by no longer scanning
+ the channel list for "other" channels with the same linkedid.
+ Instead, cel.c has an ao2 container of linkedid strings and uses
+ the refcount of the string as a counter of how many channels with
+ the linkedid exist. Not only does this eliminate the race
+ condition, but it also allows us to look up the linkedid by the
+ hashed key instead of traversing the entire channel list. Review:
+ https://reviewboard.asterisk.org/r/1895/ ........ r365068 |
+ twilson | 2012-05-02 12:02:39 -0500 (Wed, 02 May 2012) | 11 lines
+ Don't leak a ref if out of memory and can't link the linkedid If
+ the ao2_link fails, we are most likely out of memory and bad
+ things are going to happen. Before those bad things happen, make
+ sure to clean up the linkedid references. This patch also adds a
+ comment explaining why linkedid can't be passed to both local
+ channel allocations and combines two ao2_ref calls into 1.
+ Review: https://reviewboard.asterisk.org/r/1895/ ........ Merged
+ revisions 365006,365068 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions
+ 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083
+ from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-04 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.5.0-digiumphones Released.
+
+2012-05-30 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.5.0-digiumphones-rc2 Released.
+
+ * Resolve crash in subscribing for MWI notifications.
+
+ ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the
+ variable shoudl definitely not be used after that. To solve this in
+ the two cases that affect subscribing for MWI notifications, we
+ instead save the ref locally, and unref them in the error
+ conditions.
+
+ (closes issue ASTERISK-19827)
+ Reported by: B. R.
+ Review: https://reviewboard.asterisk.org/r/1940/
+
+ * Fix crash in ConfBridge when user announcement is played for more
+ than 2 users
+
+ A patch introduced in r354938 made it so that ConfBridge would not
+ attempt to play sound files if those files did not exist.
+ Unfortunately, ConfBridge uses the same underlying fucntion,
+ play_sound_helper, to playback both the sound files and numbers to
+ callers. When a number is being played back, the name of the sound
+ file is expected to be NULL. This NULL value was passed into a
+ function that tested for the existance of a sound file and is not
+ tolerant to NULL file names, causing a crash.
+
+ This patch fixes the behavior, such that if a sound file does not
+ exist we do not attempt to play it, but we only attempt that check
+ if the sound file was specified in the first place. If a sound file
+ was not specified, we use the 'play number' logic in the helper
+ function.
+
+ (closes issue ASTERISK-19899)
+ Reported by: Florian Gilcher
+ Tested by: Florian Gilcher
+ patches:
+ ASTERISK-19899.diff uploaded by mjordan (license 6283)
+
+ * AST-2012-007
+
+ * AST-2012-008
+
+2012-05-03 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.5.0-digiumphones-rc1 Released.
+
+2012-05-03 20:06 +0000 [r365264] Jason Parker <jparker@digium.com>
+
+ * main/jitterbuf.c, configs/queues.conf.sample,
+ configs/usbradio.conf.sample (removed),
+ res/res_calendar_caldav.c, apps/rpt_flow.pdf (removed),
+ apps/app_queue.c, main/cel.c, res/res_config_sqlite.c,
+ res/res_calendar_ews.c, main/config.c, formats/format_siren7.c,
+ channels/chan_dahdi.c, formats/format_vox.c, funcs/func_volume.c,
+ configure, formats/format_h263.c, main/event.c,
+ apps/app_chanspy.c, formats/format_g719.c, channels/chan_sip.c,
+ funcs/func_env.c, channels/chan_agent.c, funcs/func_strings.c,
+ channels/console_video.c, Makefile.rules, main/astfd.c,
+ formats/format_wav_gsm.c, bridges/bridge_multiplexed.c,
+ channels/chan_iax2.c, funcs/func_global.c,
+ apps/confbridge/conf_config_parser.c, res/res_config_curl.c,
+ build_tools/cflags.xml, main/cdr.c, funcs/func_curl.c,
+ main/manager.c, main/tdd.c, channels/console_gui.c,
+ formats/format_pcm.c, main/app.c, main/stdtime/localtime.c,
+ utils/extconf.c, makeopts.in, main/message.c,
+ formats/format_gsm.c, res/res_clioriginate.c,
+ include/asterisk/time.h, res/res_rtp_asterisk.c,
+ res/res_config_pgsql.c, apps/app_meetme.c, /,
+ formats/format_wav.c, configure.ac, res/res_musiconhold.c,
+ channels/chan_gtalk.c, tests/test_linkedlists.c, apps/app_ices.c,
+ channels/sig_pri.c, res/res_srtp.c, formats/format_ilbc.c,
+ channels/sig_pri.h, Makefile, apps/app_forkcdr.c,
+ res/res_config_odbc.c, bridges/bridge_builtin_features.c,
+ codecs/gsm/src/k6opt.s, build_tools/menuselect-deps.in,
+ funcs/func_channel.c, apps/app_directed_pickup.c,
+ main/features.c, res/res_agi.c, main/http.c, main/logger.c,
+ apps/app_confbridge.c, apps/app_sms.c, main/audiohook.c,
+ formats/format_h264.c, apps/app_voicemail.c,
+ codecs/lpc10/Makefile, apps/app_dial.c, formats/format_sln.c,
+ codecs/gsm/Makefile, funcs/func_sysinfo.c,
+ formats/format_ogg_vorbis.c, CHANGES, main/astobj2.c,
+ main/format_pref.c, apps/app_speech_utils.c,
+ tests/test_security_events.c, main/tcptls.c,
+ addons/ooh323cDriver.c, formats/format_g723.c,
+ apps/app_externalivr.c, tests/test_config.c, tests/test_poll.c,
+ addons/chan_mobile.c, formats/format_siren14.c,
+ funcs/func_devstate.c, main/asterisk.c, main/xmldoc.c,
+ channels/chan_mgcp.c, formats/format_g729.c,
+ channels/chan_unistim.c, configs/chan_dahdi.conf.sample,
+ main/pbx.c, res/res_calendar_icalendar.c, channels/chan_local.c,
+ funcs/func_version.c, configs/rpt.conf.sample (removed): Multiple
+ revisions
+ 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083
+ ........ r361208 | jrose | 2012-04-04 14:30:09 -0500 (Wed, 04 Apr
+ 2012) | 10 lines Make 'help devstate change' display properly
+ (get rid of excess comma) (closes issue ASTERISK-19444) Reported
+ by: Makoto Dei Patches: devstate-change-usage-truncate.patch
+ uploaded by Makoto Dei (license 5027) ........ Merged revisions
+ 361201 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r361211 | jrose | 2012-04-04 15:00:23 -0500 (Wed, 04 Apr
+ 2012) | 12 lines Fix some stuff involving calls to memcpy and
+ memset The important parts of the patch were already applied
+ through other updates. (closes issue ASTERISK-19445) Reported by:
+ Makoto Dei Patches: memset-memcpy-length.patch uploaded by Makoto
+ Dei (license 5027) ........ Merged revisions 361210 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361270 | jrose | 2012-04-05 11:53:35 -0500 (Thu, 05 Apr 2012) |
+ 10 lines Fix MusicOnHold in MeetMe so that it always uses the
+ class if it's been defined There were a few instances of
+ restarting music on hold in meetme that would cause Asterisk to
+ revert to the default class of music on hold for no adequate
+ reason. Review: https://reviewboard.asterisk.org/r/1844/ ........
+ Merged revisions 361269 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361330 | kmoore | 2012-04-06 08:31:51 -0500 (Fri, 06 Apr 2012) |
+ 11 lines Remove unnecessary error message in app_dial.c The error
+ message for failure to stop autoservice after a gosub or macro
+ call during a dial was removed for macro while Asterisk 1.4 was
+ still being actively developed. The corresponding gosub error
+ message was never removed. (closes issue ASTERISK-19551) ........
+ Merged revisions 361329 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361333 | mjordan | 2012-04-06 09:01:33 -0500 (Fri, 06 Apr 2012)
+ | 11 lines Fix a typo in the warning messages for an ignored
+ media stream Added a '\n' to the warning messages when we ignore
+ a media stream due to the port number being '0'. (closes issue
+ ASTERISK-19646) Reported by: Badalian Vyacheslav ........ Merged
+ revisions 361332 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361381 | russell | 2012-04-06 10:49:19 -0500 (Fri, 06 Apr 2012)
+ | 5 lines Remove a few more files related to chan_usbradio and
+ app_rpt. ........ Merged revisions 361380 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361422 | pabelanger | 2012-04-06 11:31:18 -0500 (Fri, 06 Apr
+ 2012) | 14 lines Multiple revisions 361403,361412 ........
+ r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri, 06 Apr
+ 2012) | 2 lines Fix typo in svn:keywords ........ r361412 |
+ pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2
+ lines Fix typo in svn:keywords ........ Merged revisions
+ 361403,361412 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361472 | kmoore | 2012-04-06 13:13:04 -0500 (Fri, 06 Apr 2012) |
+ 5 lines Add missing newlines to CLI logging ........ Merged
+ revisions 361471 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361522 | rmudgett | 2012-04-06 14:47:29 -0500 (Fri, 06 Apr 2012)
+ | 8 lines Don't add an empty MESSAGE_DATA(key) header if it
+ doesn't already exist. Doing Set(MESSAGE_DATA(key)=) would add an
+ empty key header if the key header did not already exist. If it
+ already existed it would delete it. * Made msg_set_var_full()
+ exit early if the named variable did not already exist and the
+ value to set is empty. ........ r361560 | mjordan | 2012-04-06
+ 15:32:13 -0500 (Fri, 06 Apr 2012) | 13 lines Fix memory leak when
+ using MeetMeAdmin 'e' option with user specified A memory
+ leak/reference counting leak occurs if the MeetMeAdmin 'e'
+ command (eject last user that joined) is used in conjunction with
+ a specified user. Regardless of the command being executed, if a
+ user is specified for the command, MeetMeAdmin will look up that
+ user. Because the 'e' option kicks the last user that joined, as
+ opposed to the one specified, the reference to the user specified
+ by the command would be leaked when the user variable was
+ assigned to the last user that joined. ........ Merged revisions
+ 361558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r361607 | mjordan | 2012-04-06 17:00:11 -0500 (Fri, 06
+ Apr 2012) | 12 lines Fix memory leak in res_calendar_ews when
+ event email address node is empty If the XML calendar data
+ returned by a Microsoft Exchange Web Service specifies an XML
+ Event E-Mail Address ("EmailAddress"), and no e-mail address is
+ provided, a condition existed where an ast_calendar_attendee
+ struct would be allocated but not appended to the list of
+ attendees. Because of that, the memory associated with the
+ attendee would never be freed. This patch frees the memory if no
+ e-mail address is provided. ........ Merged revisions 361606 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361658 | mjordan | 2012-04-09 14:42:53 -0500 (Mon, 09 Apr 2012)
+ | 15 lines Change SHARED function to use a safe traversal when
+ modifying a variable When the SHARED function modifies a
+ variable, it removes it from its list of variables and reinserts
+ the new value at the head of the list of variables. Doing this
+ inside a standard list traversal can be dangerous, as the
+ standard list traversal does not account for the list being
+ changed. While the code in question should not cause a use after
+ free violation due to its breaking out of the loop after freeing
+ the variable, it could lead to a maintenance issue if the loop
+ was modified. This also fixes a violation reported by a static
+ analysis tool, which also makes this code easier to maintain in
+ the future. ........ Merged revisions 361657 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361706 | mjordan | 2012-04-09 15:54:55 -0500 (Mon, 09 Apr 2012)
+ | 17 lines Prevent invalid access of free'd memory if DAHDI
+ channel during an MWI event In the MWI processing loop, when a
+ valid event occurs the temporary caller ID information is
+ deallocated. If a new DAHDI channel is successfully created, the
+ event is passed up to the analog_ss_thread without error and the
+ loop exits. If, however, the DAHDI channel is not created, then
+ the caller ID struct has been free'd, and the gains reset to
+ their previous level. This will almost certainly cause an invalid
+ access to the free'd memory, either in subsequent calls to
+ callerid_free or calls to callerid_feed. This patch makes it so
+ that we only free the caller ID structure if a DAHDI channel is
+ successfully created, and we bump the gains back up if we fail to
+ make a DAHDI channel. ........ Merged revisions 361705 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361754 | mjordan | 2012-04-09 16:44:30 -0500 (Mon, 09 Apr 2012)
+ | 12 lines Allow func_curl to exit gracefully if list allocation
+ fails during write If the global_curl_info data structure could
+ not be allocated, the datastore associated with the operation
+ would be free'd, but the function would not return. This would
+ later dereference the datastore, almost certainly causing
+ Asterisk to crash. With this patch, if the data structure is not
+ allocated the method will return an error code, and not attempt
+ any further operation. ........ Merged revisions 361753 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361804 | mjordan | 2012-04-10 14:57:30 -0500 (Tue, 10 Apr 2012)
+ | 10 lines Fix crash caused by unloading or reloading of
+ res_http_post When unlinking itself from the registered HTTP
+ URIs, res_http_post could inadvertently free all URIs registered
+ with the HTTP server. This patch modifies the unregister method
+ to only free the URI that is actually being unregistered, as
+ opposed to all of them. ........ Merged revisions 361803 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361855 | rmudgett | 2012-04-10 16:47:42 -0500 (Tue, 10 Apr 2012)
+ | 19 lines Prevent invalid access of free'd memory if DAHDI
+ channel during an MWI event In the MWI processing loop, when a
+ valid event occurs the temporary caller ID information is
+ deallocated. If a new DAHDI channel is successfully created, the
+ event is passed up to the analog_ss_thread without error and the
+ loop exits. If, however, the DAHDI channel is not created, then
+ the caller ID struct has been free'd, and the gains reset to
+ their previous level. This will almost certainly cause an invalid
+ access to the free'd memory, either in subsequent calls to
+ callerid_free or calls to callerid_feed. * Rework the -r361705
+ patch to better manage the cs and mtd allocated resources. *
+ Fixed use of mwimonitoractive flag to be correct if the
+ mwi_thread() fails to start. ........ Merged revisions 361854
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361907 | jrose | 2012-04-11 11:07:50 -0500 (Wed, 11 Apr 2012) |
+ 10 lines Change default value of 'ignorebusy' on Queue members so
+ that behavior is more like 1.8 Prior to this patch, in order to
+ restore that behavior, a function would have to be used on the
+ QueueMember to make the ringinuse option do anything, which is
+ pretty unreasonable. (closes issue ASTERISK-19536) reported by:
+ Philippe Lindheimer Review:
+ https://reviewboard.asterisk.org/r/1860/ ........ r361956 |
+ kmoore | 2012-04-12 10:01:13 -0500 (Thu, 12 Apr 2012) | 13 lines
+ Simplify build system architecture optimization This change to
+ the build system rips out any usage of PROC along with
+ architecture-specific optimizations in favor of using
+ -march=native where it is supported. This fixes broken builds on
+ 64bit Intel systems and results in better optimized code on
+ systems running GCC 4.2+. Review:
+ https://reviewboard.asterisk.org/r/1852/ (closes issue
+ ASTERISK-19462) ........ Merged revisions 361955 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r361981 | kmoore | 2012-04-12 11:22:28 -0500 (Thu, 12 Apr 2012) |
+ 12 lines Make trunkfreq take effect when set Previously, setting
+ trunkfreq had no effect on initial load or on reload and only
+ ever used the default value. This causes trunkfreq to be used
+ appropriately on initial load and reload. (closes issue
+ ASTERISK-19521) Patch-by: Jaco Kroon ........ Merged revisions
+ 361972 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r362080 | jrose | 2012-04-13 10:30:22 -0500 (Fri, 13 Apr
+ 2012) | 10 lines Send relative path named recordings to the
+ meetme directory instead of sounds Prior to this patch, no effort
+ was made to parse the path name to determine a proper destination
+ for recordings of MeetMe's r option. This fixes that. Review:
+ https://reviewboard.asterisk.org/r/1846/ ........ Merged
+ revisions 362079 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362084 | jrose | 2012-04-13 11:04:22 -0500 (Fri, 13 Apr 2012) |
+ 15 lines Make ForkCDR e option not set end time of the newly
+ forked CDR log Prior to this patch, ForkCDR's e option would
+ immediately set the end time of the forked CDR to that of the CDR
+ that is being terminated. This resulted in the new CDR's end time
+ being roughly the same as it's beginning time (which is in turn
+ roughly the same as the original's end time). (closes issue
+ ASTERISK-19164) Reported by: Steve Davies Patches:
+ cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)
+ ........ Merged revisions 362082 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362152 | mjordan | 2012-04-16 14:39:32 -0500 (Mon, 16 Apr 2012)
+ | 19 lines Check for IO stream failures in various format's
+ truncate/seek operations For the formats that support seek and/or
+ truncate operations, many of the C library calls used to
+ determine or set the current position indicator in the file
+ stream were not being checked. In some situations, if an error
+ occurred, a negative value would be returned from the library
+ call. This could then be interpreted inappropriately as
+ positional data. This patch checks the return values from these
+ library calls before using them in subsequent operations. (issue
+ ASTERISK-19655) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1863/ ........ Merged
+ revisions 362151 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362202 | mjordan | 2012-04-16 16:40:29 -0500 (Mon, 16 Apr 2012)
+ | 18 lines Fix handling of negative return code when storing
+ voicemails in ODBC storage When storing a voicemail message using
+ an ODBC connection to a database, the voicemail message is first
+ stored on disk. The sound file associated with the message is
+ read into memory before being transmitted to the database. When
+ this occurs, a failure in the C library's lseek function would
+ cause a negative value to be passed to the mmap as the size of
+ the memory map to create. This would almost certainly cause the
+ creation of the memory map to fail, resulting in the message
+ being lost. (issue ASTERISK-19655) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/1863 ........ Merged
+ revisions 362201 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362205 | mjordan | 2012-04-16 16:57:19 -0500 (Mon, 16 Apr 2012)
+ | 25 lines Fix negative return handling in channel drivers In
+ chan_agent, while handling a channel indicate, the agent channel
+ driver must obtain a lock on both the agent channel, as well as
+ the channel the agent channel is using. To do so, it attempts to
+ lock the other channel first, then unlock the agent channel which
+ is locked prior to entry into the indicate handler. If this
+ unlock fails with a negative return value, which can occur if the
+ object passed to agent_indicate is an invalid ao2 object or is
+ NULL, the return value is passed directly to strerror, which can
+ only accept positive integer values. In chan_dahdi, the return
+ value of dahdi_get_index is used to directly index into the
+ sub-channel array. If dahd_get_index returns a negative value, it
+ would use that value to index into the array, which could cause
+ an invalid memory access. If dahdi_get_index returns a negative
+ number, we now default to SUB_REAL. (issue ASTERISK-19655)
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1863/ ........ Merged
+ revisions 362204 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362264 | elguero | 2012-04-17 09:53:04 -0500 (Tue, 17 Apr 2012)
+ | 23 lines Turn off warning message when bind address is set to
+ any. When a bind address is set to an ANY address
+ (udpbindport=::), a warning message is displayed stating that
+ "Address remapping activated in sip.conf but we're using IPv6,
+ which doesn't need it. Please remove 'localnet' and/or
+ 'externaddr' settings." But if one is running dual stack, we
+ shouldn't be told to turn those settings off. This patch checks
+ if the bind address is an ANY address or not. The warning message
+ will now only be displayed if the bind address is NOT an ANY
+ address and IPv6 is being used. Also, updated the copyright year.
+ (closes issue ASTERISK-19456) Reported by: Michael L. Young
+ Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff
+ uploaded by Michael L. Young (license 5026) ........ Merged
+ revisions 362253 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362305 | mjordan | 2012-04-17 13:27:44 -0500 (Tue, 17 Apr 2012)
+ | 15 lines Fix error that caused seek format operations to set
+ max file size to '1' or '0' A very inappropriate placement of a
+ ')' (introduced in r362151) caused the maximum size of a file to
+ be set as the result of a comparison operation, as opposed to the
+ result of the ftello operation. This resulted in seeking being
+ restricted to the beginning of the file, or 1 byte into the file.
+ Thanks to the Asterisk Test Suite for properly freaking out about
+ this on at least one test. (issue ASTERISK-19655) Reported by:
+ Matt Jordan ........ Merged revisions 362304 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362356 | mjordan | 2012-04-17 15:56:05 -0500 (Tue, 17 Apr 2012)
+ | 17 lines Fix places where a negative return from ftello could
+ be used as invalid input In a variety of locations in both
+ reading and writing a file, the result from the C library
+ function ftello is used as input to other functions. For the
+ parameters and functions in question, a negative value is invalid
+ input. This patch checks the return value from the ftello
+ function to determine if we were able to determine the current
+ position in the file stream and, if not, fail gracefully. (issue
+ ASTERISK-19655) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1863/ ........ Merged
+ revisions 362355 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362357 | jrose | 2012-04-17 15:57:36 -0500 (Tue, 17 Apr 2012) |
+ 12 lines Make use of va_args more appropriate to form in various
+ res_config modules plus utils. A number of va_copy operations
+ weren't matched with a corresponding va_end in res_config_odbc.
+ Also, there was a potential for va_end to be invoked twice on the
+ same va_arg in utils, which would mean invoking va_end on an
+ undefined variable... which is bad. va_end is removed from
+ various functions in config_pgsql and config_curl since they
+ aren't making their own copy. The invokers of those functions are
+ responsible for calling va_end on them. (issue ASTERISK-19451)
+ Reported by: Walter Doekes Review:
+ https://reviewboard.asterisk.org/r/1848/ ........ Merged
+ revisions 362354 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362360 | mjordan | 2012-04-17 16:07:29 -0500 (Tue, 17 Apr 2012)
+ | 24 lines Fix places in main where a negative return value could
+ impact execution This patch addresses a number of modules in main
+ that did not handle the negative return value from function calls
+ adequately, or were not sufficiently clear that the conditions
+ leading to improper handling of the return values could not
+ occur. This includes: * asterisk.c: A negative return value from
+ the read function would be used directly as an index into a
+ buffer. We now check for success of the read function prior to
+ using its result as an index. * manager.c: Check for failures in
+ mkstemp and lseek when handling the temporary file created for
+ processing data returned from a CLI command in action_command.
+ Also check that the result of an lseek is sanitized prior to
+ using it as the size of a memory map to allocate. (issue
+ ASTERISK-19655) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1863/ ........ Merged
+ revisions 362359 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362364 | mjordan | 2012-04-17 16:11:25 -0500 (Tue, 17 Apr 2012)
+ | 29 lines Fix places in resources where a negative return value
+ could impact execution This patch addresses a number of modules
+ in resources that did not handle the negative return value from
+ function calls adequately. This includes: * res_agi.c: if the
+ result of the read function is a negative number, indicating some
+ failure, the result would instead be treated as the number of
+ bytes read. This patch now treats negative results in the same
+ manner as an end of file condition, with the exception that it
+ also logs the error code indicated by the return. *
+ res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor
+ to srcfd, and instead assigns a negative value, that file
+ descriptor could later be passed to functions that require a
+ valid file descriptor. If spawn_mp3 fails, we now immediately
+ retry instead of continuing in the logic. * res_rtp_asterisk.c:
+ if no codec can be matched between two RTP instances in a peer to
+ peer bridge, we immediately return instead of attempting to use
+ the codec payload type as an index to determine the appropriate
+ negotiated codec. (issue ASTERISK-19655) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged
+ revisions 362362 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362377 | mjordan | 2012-04-17 16:22:37 -0500 (Tue, 17 Apr 2012)
+ | 13 lines Handle case where an unknown format is used to get the
+ preferred codec size In ast_codec_pref_getsize, if an unknown
+ format is passed to the method, no preferred codec will be
+ selected and a negative number will be used to index into the
+ format list. The method now logs an unknown format as a warning,
+ and returns an empty format list. (issue ASTERISK-19655) Reported
+ by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/
+ ........ r362429 | rmudgett | 2012-04-18 11:27:51 -0500 (Wed, 18
+ Apr 2012) | 19 lines Add ability to ignore layer 1 alarms for BRI
+ PTMP lines. Several telcos bring the BRI PTMP layer 1 down when
+ the line is idle. When layer 1 goes down, Asterisk cannot make
+ outgoing calls. Incoming calls could fail as well because the
+ alarm processing is handled by a different code path than the
+ Q.931 messages. * Add the layer1_presence configuration option to
+ ignore layer 1 alarms when the telco brings layer 1 down. This
+ option can be configured by span while the similar DAHDI driver
+ teignorered=1 option is system wide. This option unlike
+ layer2_persistence does not require libpri v1.4.13 or newer.
+ Related to JIRA AST-598 JIRA ABE-2845 ........ Merged revisions
+ 362428 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r362496 | mjordan | 2012-04-18 21:27:08 -0500 (Wed, 18
+ Apr 2012) | 50 lines Fix a variety of potential buffer overflows
+ * chan_mobile: Fixed an overrun where the cind_state buffer (an
+ integer array of size 16) would be overrun due to improper bounds
+ checking. At worst, the buffer can be overrun by a total of 48
+ bytes (assuming 4-byte integers), which would still leave it
+ within the allocated memory of struct hfp. This would corrupt
+ other elements in that struct but not necessarily cause any
+ further issues. * app_sms: The array imsg is of size 250, while
+ the array (ud) that the data is copied into is of size 160. If
+ the size of the inbound message is greater then 160, up to 90
+ bytes could be overrun in ud. This would corrupt the user data
+ header (array udh) adjacent to ud. * chan_unistim: A number of
+ invalid memmoves are corrected. These would move data (which may
+ or may not be valid) into the ends of these buffers. * asterisk:
+ ast_console_toggle_loglevel does not check that the console log
+ level being set is less then or equal to the allowed log levels
+ of 32. * format_pref: In ast_codec_pref_prepend, if any
+ occurrence of the specified codec is not found, the value used to
+ index into the array pref->order would be one greater then the
+ maximum size of the array. * jitterbuf: If the element being
+ placed into the jitter buffer lands in the last available slot in
+ the jitter history buffer, the insertion sort attempts to move
+ the last entry in the buffer into one slot past the maximum
+ length of the buffer. Note that this occurred for both the min
+ and max jitter history buffers. * tdd: If a read from fsk_serial
+ returns a character that is greater then 32, an attempt to read
+ past one of the statically defined arrays containing the values
+ that character maps to would occur. * localtime: struct ast_time
+ and tm are not the same size - ast_time is larger, although it
+ contains the elements of tm within it in the same layout. Hence,
+ when using memcpy to copy the contents of tm into ast_time, the
+ size of tm should be used, as opposed to the size of ast_time. *
+ extconf: this treats ast_timing's minmask array as if it had a
+ length of 48, when it has defined the size of the array as 24.
+ pbx.h defines minmask as having a size of 48. (issue
+ ASTERISK-19668) Reported by: Matt Jordan ........ Merged
+ revisions 362485 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362537 | twilson | 2012-04-19 09:31:59 -0500 (Thu, 19 Apr 2012)
+ | 14 lines Handle multiple commands per connection via netconsole
+ Asterisk would accept multiple NULL-delimited CLI commands via
+ the netconsole socket, but would occasionally miss a command due
+ to the command not being completely read into the buffer. This
+ patch ensures that any partial commands get moved to the front of
+ the read buffer, appended to, and properly sent. (closes issue
+ ASTERISK-18308) Review: https://reviewboard.asterisk.org/r/1876/
+ ........ Merged revisions 362536 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362587 | seanbright | 2012-04-19 11:04:21 -0500 (Thu, 19 Apr
+ 2012) | 12 lines Prevent a crash in ExternalIVR when the 'S'
+ command is sent first. If the first command sent from an
+ ExternalIVR client is an 'S' command, we were blindly removing
+ the first element from the play list and deferencing it, even if
+ it was NULL. This corrects that and also locks appropriately in
+ one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski
+ ........ Merged revisions 362586 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362678 | rmudgett | 2012-04-19 16:00:21 -0500 (Thu, 19 Apr 2012)
+ | 5 lines Update membermacro and membergosub documentation in
+ queues.conf.sample. ........ Merged revisions 362677 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362681 | elguero | 2012-04-19 16:11:35 -0500 (Thu, 19 Apr 2012)
+ | 9 lines Add leading and trailing backslashes A couple of unit
+ tests did not have have leading or trailing backslashes when
+ setting their test category resulting in a warning message being
+ displayed. Added the backslash where needed. ........ Merged
+ revisions 362680 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362730 | wdoekes | 2012-04-19 16:59:43 -0500 (Thu, 19 Apr 2012)
+ | 5 lines Fix documentation for ${VERSION(ASTERISK_VERSION_NUM)}.
+ ........ Merged revisions 362729 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362816 | twilson | 2012-04-20 09:49:42 -0500 (Fri, 20 Apr 2012)
+ | 13 lines Document Speech* apps hangup on failure and suggest
+ TryExec The Speech API apps return -1 on failure, which will hang
+ up the channel. This may not be desirable behavior for some, but
+ it isn't something that can be changed without breaking people's
+ dialplans or writing an option to all of the Speech apps that
+ does what TryExec already does. This patch documents the hangup
+ behavior of the apps, and suggests TryExec as the solution.
+ (closes issue AST-813) ........ Merged revisions 362815 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362869 | twilson | 2012-04-20 11:12:34 -0500 (Fri, 20 Apr 2012)
+ | 11 lines OpenBSD doesn't have rawmemchr, use strchr (closes
+ issue ASTERISK-19758) Reported by: Barry Miller Tested by: Terry
+ Wilson Patches: 362758-diff uploaded by Barry Miller (license
+ 5434) ........ Merged revisions 362868 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r362918 | elguero | 2012-04-20 11:47:51 -0500 (Fri, 20 Apr 2012)
+ | 11 lines Add missing payload type to events API The Security
+ Events Framework API was changed while adding the generation of
+ security events in chan_sip. A payload type and name was missed
+ from being added to struct ie_maps. (closes issue ASTERISK-19759)
+ Reported by: Michael L. Young Patches: issue-asterisk-19759.diff
+ uploaded by Michael L. Young (license 5026) ........ r362998 |
+ rmudgett | 2012-04-20 20:45:13 -0500 (Fri, 20 Apr 2012) | 5 lines
+ Update app_dial M and U option GOTO return value documentation.
+ ........ Merged revisions 362997 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363212 | tilghman | 2012-04-23 11:06:53 -0500 (Mon, 23 Apr 2012)
+ | 8 lines On some platforms, O_RDONLY is not a flag to be
+ checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX
+ specification does not mandate how these 3 flags must be
+ specified, only that one of the three must be specified in every
+ call. ........ Merged revisions 363209 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363376 | rmudgett | 2012-04-24 19:01:21 -0500 (Tue, 24 Apr 2012)
+ | 5 lines Hangup affected channel in error paths of
+ bridge_call_thread(). ........ Merged revisions 363375 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363429 | rmudgett | 2012-04-24 20:23:08 -0500 (Tue, 24 Apr 2012)
+ | 27 lines Fix recalled party B feature flags for a failed DTMF
+ atxfer. 1) B calls A with Dial option T 2) B DTMF atxfer to C 3)
+ B hangs up 4) C does not answer 5) B is called back 6) B answers
+ 7) B cannot initiate transfers anymore * Add dial features
+ datastore to recalled party B channel that is a copy of the
+ original party B channel's dial features datastore. * Extracted
+ add_features_datastore() from add_features_datastores(). *
+ Renamed struct ast_dial_features features_caller and
+ features_callee members to my_features and peer_features
+ respectively. These better names eliminate the need for some
+ explanatory comments. * Simplified code accessing the struct
+ ast_dial_features datastore. (closes issue ASTERISK-19383)
+ Reported by: lgfsantos ........ Merged revisions 363428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363688 | rmudgett | 2012-04-25 14:47:44 -0500 (Wed, 25 Apr 2012)
+ | 19 lines Clear ISDN channel resetting state if the peer
+ continues to use it. Some ISDN switches occasionally fail to send
+ a RESTART ACKNOWLEDGE in response to a RESTART request. * Made
+ the second SETUP received after sending a RESTART request clear
+ the channel resetting state as if the peer had sent the expected
+ RESTART ACKNOWLEDGE before continuing to process the SETUP. The
+ peer may not be sending the expected RESTART ACKNOWLEDGE. (issue
+ ASTERISK-19608) (issue AST-844) (issue AST-815) Patches:
+ jira_ast_815_v1.8.patch (license #5621) patch uploaded by
+ rmudgett (modified) ........ Merged revisions 363687 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363734 | rmudgett | 2012-04-25 15:48:22 -0500 (Wed, 25 Apr 2012)
+ | 18 lines Make DAHDISendCallreroutingFacility wait 5 seconds for
+ a reply before disconnecting the call. Some switches may not
+ handle the call-deflection/call-rerouting message if the call is
+ disconnected too soon after being sent. Asteisk was not waiting
+ for any reply before disconnecting the call. * Added a 5 second
+ delay before disconnecting the call to wait for a potential
+ response if the peer does not disconnect first. (closes issue
+ ASTERISK-19708) Reported by: mehdi Shirazi Patches:
+ jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett ........ Merged revisions 363730
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363789 | rmudgett | 2012-04-25 17:59:46 -0500 (Wed, 25 Apr 2012)
+ | 5 lines Update Pickup application documentation. ........
+ Merged revisions 363788 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363876 | rmudgett | 2012-04-25 22:11:45 -0500 (Wed, 25 Apr 2012)
+ | 5 lines Update Pickup application documentation. (Even better)
+ ........ Merged revisions 363875 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363935 | alecdavis | 2012-04-26 04:46:38 -0500 (Thu, 26 Apr
+ 2012) | 14 lines chan_sip: [general] maxforwards, not checked for
+ a value greater than 255 The peer maxforwards is checked for both
+ '< 1' and '> 255', but the default 'maxforwards' in the [general]
+ section is only checked for '< 1' alecdavis (license 585)
+ Reported by: alecdavis Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1888/ ........ Merged
+ revisions 363934 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r363987 | kmoore | 2012-04-26 08:27:34 -0500 (Thu, 26 Apr 2012) |
+ 15 lines Fix reference leaks involving SIP Replaces transfers The
+ reference held for SIP blind transfers using the Replaces header
+ in an INVITE was never freed on success and also failed to be
+ freed in some error conditions. This caused a file descriptor
+ leak since the RTP structures in use at the time of the transfer
+ were never freed. This reference leak and another relating to
+ subscriptions in the same code path have now been corrected.
+ (Closes issue ASTERISK-19579) Reported by: Maciej Krajewski
+ Tested by: Maciej Karjewski ........ Merged revisions 363986 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364047 | twilson | 2012-04-26 14:30:55 -0500 (Thu, 26 Apr 2012)
+ | 8 lines Add more constness to the end_buf pointer in the
+ netconsole issue ASTERISK-18308 Review:
+ https://reviewboard.asterisk.org/r/1876/ ........ Merged
+ revisions 364046 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364065 | rmudgett | 2012-04-26 15:25:05 -0500 (Thu, 26 Apr 2012)
+ | 24 lines Fix DTMF atxfer running h exten after the wrong bridge
+ ends. When party B does an attended transfer of party A to party
+ C, the attending bridge between party B and C should not be
+ running an h exten when the bridge ends. Running an h exten now
+ sets a softhangup flag to ensure that an AGI will run in dead AGI
+ mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B
+ channel for the attending bridge between party B and C. (closes
+ issue AST-870) (closes issue ASTERISK-19717) Reported by: Mario
+ (closes issue ASTERISK-19633) Reported by: Andrey Solovyev
+ Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario
+ ........ Merged revisions 364060 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364109 | rmudgett | 2012-04-26 16:10:46 -0500 (Thu, 26 Apr 2012)
+ | 5 lines Update Pickup application documentation. (With feeling
+ this time.) ........ Merged revisions 364108 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364163 | schmidts | 2012-04-27 07:54:19 -0500 (Fri, 27 Apr 2012)
+ | 3 lines fix a wrong behavior of alarm timezones in caldav and
+ icalendar when an alarm doesnt use utc. This change uses the same
+ timezone from the start time. ........ r364204 | mjordan |
+ 2012-04-27 09:44:13 -0500 (Fri, 27 Apr 2012) | 23 lines Allow for
+ reloading SRTP crypto keys within the same SIP dialog As a
+ continuation of the patch in r356604, which allowed for the
+ reloading of SRTP keys in re-INVITE transfer scenarios, this
+ patch addresses the more common case where a new key is requested
+ within the context of a current SIP dialog. This can occur, for
+ example, when certain phones request a SIP hold. Previously, once
+ a dialog was associated with an SRTP object, any subsequent
+ attempt to process crypto keys in any SDP offer - either the
+ current one or a new offer in a new SIP request - were ignored.
+ This patch changes this behavior to only ignore subsequent crypto
+ keys within the current SDP offer, but allows future SDP offers
+ to change the keys. (issue ASTERISK-19253) Reported by: Thomas
+ Arimont Tested by: Thomas Arimont Review:
+ https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions
+ 364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r364259 | kmoore | 2012-04-27 13:58:34 -0500 (Fri, 27
+ Apr 2012) | 14 lines Allow SIP pvts involved in Replaces
+ transfers to fall out of reference sooner Unref the SIP pvt
+ stored in the refer structure as soon as it is no longer needed
+ so that the pvt and associated file descriptors can be freed
+ sooner. This change makes a reference decrement unnecessary in
+ code that handles SIP BYE/Also transfers which should not touch
+ the reference anyway. (Closes issue ASTERISK-19579) Reported by:
+ Maciej Krajewski Tested by: Maciej Krajewski ........ Merged
+ revisions 364258 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364285 | mjordan | 2012-04-27 14:30:19 -0500 (Fri, 27 Apr 2012)
+ | 43 lines Prevent overflow in calculation in ast_tvdiff_ms on
+ 32-bit machines The method ast_tvdiff_ms attempts to calculate
+ the difference, in milliseconds, between two timeval structs, and
+ return the difference in a 64-bit integer. Unfortunately, it
+ assumes that the long tv_sec/tv_usec members in the timeval
+ struct are large enough to hold the calculated values before it
+ returns. On 64-bit machines, this might be the case, as a long
+ may be 64-bits. On 32-bit machines, however, a long may be less
+ (32-bits), in which case, the calculation can overflow. This
+ overflow caused significant problems in MixMonitor, which uses
+ the method to determine if an audio factory, which has not
+ presented audio to an audiohook, is merely late in providing said
+ audio or will never provide audio. In an overflow situation, the
+ audiohook would incorrectly determine that an audio factory that
+ will never provide audio is merely late instead. This led to
+ situations where a MixMonitor never recorded any audio. Note that
+ this happened most frequently when that MixMonitor was started by
+ the ConfBridge application itself, or when the MixMonitor was
+ attached to a Local channel. (issue ASTERISK-19497) Reported by:
+ Ben Klang Tested by: Ben Klang Patches:
+ 32-bit-time-overflow-10-2012-04-26.diff (license #6283) by
+ mjordan (closes issue ASTERISK-19727) Reported by: Mark Murawski
+ Tested by: Michael L. Young Patches:
+ 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
+ (closes issue ASTERISK-19471) Reported by: feyfre Tested by:
+ feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review:
+ https://reviewboard.asterisk.org/r/1889/ ........ Merged
+ revisions 364277 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364342 | mmichelson | 2012-04-27 16:58:06 -0500 (Fri, 27 Apr
+ 2012) | 10 lines Don't attempt to make use of the
+ dynamic_exclude_static ACL if DNS lookup fails. (closes issue
+ ASTERISK-18321) Reported by Dan Lukes Patches:
+ ASTERISK-18321.patch by Mark Michelson (license #5049) ........
+ Merged revisions 364341 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364365 | twilson | 2012-04-27 17:31:01 -0500 (Fri, 27 Apr 2012)
+ | 11 lines Fix ast_parse_arg numeric type range checking and add
+ tests ast_parse_arg wasn't checking for strto* parse errors or
+ limiting the results by the actual range of the numeric types.
+ This patch fixes that and adds unit tests as well. Review:
+ https://reviewboard.asterisk.org/r/1879/ ........ Merged
+ revisions 364340 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012)
+ | 2 lines Add missing test_config.c ........ r364536 | elguero |
+ 2012-04-28 21:21:10 -0500 (Sat, 28 Apr 2012) | 13 lines Fix
+ configuring custom sound_leader_has_left in confbridge.conf The
+ configuration option to specify a custom sound_leader_has_left
+ file for a conference bridge was not being parsed. This patch
+ fixes it so that a custom sound file will now be used. (closes
+ issue ASTERISK-19771) Reported by: Pawel Kuzak Tested by: Pawel
+ Kuzak, Michael L. Young Patches: leaderhasleft_sound.dpatch
+ uploaded by Pawel Kuzak (license 6380) Review:
+ https://reviewboard.asterisk.org/r/1884/ ........ r364579 |
+ mjordan | 2012-04-29 14:43:53 -0500 (Sun, 29 Apr 2012) | 15 lines
+ Fix error that caused truncate operations to fail Another very
+ inappropriate placement of a ')' (again introduced in r362151)
+ caused the various truncate operations to attempt to truncate the
+ sound file at a position of '0'. (issue ASTERISK-19655) Reported
+ by: Matt Jordan (issue ASTERISK-19810) Reported by: colbec
+ ........ Merged revisions 364578 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364650 | markm | 2012-04-30 11:43:11 -0500 (Mon, 30 Apr 2012) |
+ 15 lines Merged revisions 364635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) |
+ 10 lines Sanatize result from bfd_find_nearest_line
+ (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file
+ to null resulting in a crash when strrchr(file) runs (closes
+ issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark
+ Murawski ........ ........ r364651 | may | 2012-04-30 11:48:57
+ -0500 (Mon, 30 Apr 2012) | 10 lines Fix use freed pointer in
+ return value from call thread (issue ASTERISK-19663) Reported by:
+ Matt Jordan Patches: ASTERISK-19663-ooh323.patch (License #5415)
+ ........ Merged revisions 364649 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364777 | jrose | 2012-05-01 13:23:08 -0500 (Tue, 01 May 2012) |
+ 13 lines Fix bad check in voicemail functions for
+ ast_inboxcount2_func Check looks for ast_inboxcount_func instead
+ of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes
+ issue ASTERISK-19718) Reported by: Corey Farrell Patches:
+ ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell
+ (license 5909) ........ Merged revisions 364769 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364787 | kmoore | 2012-05-01 14:07:09 -0500 (Tue, 01 May 2012) |
+ 12 lines Play conf-placeintoconf message to the correct channel
+ Correct the code in app_confbridge to play the conf-placeintoconf
+ message to the marked user entering the bridge instead of to the
+ conference while the marked user hears silence. (closes issue
+ ASTERISK-19641) Reported-by: Mark A Walters ........ Merged
+ revisions 364786 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364845 | rmudgett | 2012-05-01 16:50:32 -0500 (Tue, 01 May 2012)
+ | 7 lines * Fix error path resouce leak in local_request(). *
+ Restructure local_request() to reduce indentation. ........
+ Merged revisions 364840 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364900 | mmichelson | 2012-05-01 18:10:16 -0500 (Tue, 01 May
+ 2012) | 16 lines Fix Coverity-reported ARRAY_VS_SINGLETON error.
+ As it turned out, this wasn't a huge deal. We were calling
+ ast_app_parse_options() for a set of options of which none took
+ arguments. The proper thing to do for this case is to pass NULL
+ for the "args" parameter here. We were instead passing a
+ seemingly-randomly chosen char * from the function. While this
+ would never get written to, you can rest assured things would
+ have gotten bad had new options (which took arguments) been added
+ to func_volume. (closes issue ASTERISK-19656) ........ Merged
+ revisions 364899 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364903 | rmudgett | 2012-05-01 18:14:12 -0500 (Tue, 01 May 2012)
+ | 10 lines Fixed __ao2_ref() validating user_data twice. (closes
+ issue ASTERISK-19755) Reported by: Gunther Kelleter Patches:
+ ao2_ref.patch (license #6372) patch uploaded by Gunther Kelleter
+ ........ Merged revisions 364902 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364965 | mjordan | 2012-05-01 21:44:15 -0500 (Tue, 01 May 2012)
+ | 11 lines Only log a failure to get read/write samples from
+ factories if it didn't happen In audiohook_read_frame_both,
+ anytime samples are obtained from the read/write factories a
+ debug statement is logged stating that samples were not obtained
+ from the factories. This statement used to only occur if
+ option_debug was turned on and no samples were obtained; in some
+ refactoring when the option_debug statement was removed, the
+ "else" clause was removed as well. This patch makes it so that
+ those debug log statements only occur if the condition leading up
+ to them actually happened. ........ r365014 | elguero |
+ 2012-05-02 11:16:03 -0500 (Wed, 02 May 2012) | 18 lines Update
+ security events unit tests The security events framework API was
+ changed in Asterisk 10 but the unit tests were not updated at the
+ same time. This patch does the following: * Adds two more
+ security events that were added to the API * Add challenge,
+ received_challenge and received_hash in the inval_password
+ security event unit test (issue ASTERISK-19760) Reported by:
+ Michael L. Young Tested by: Michael L. Young Patches:
+ issue-asterisk-19760-branch10.diff uploaded by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/1877/
+ ........ r365083 | twilson | 2012-05-02 12:29:54 -0500 (Wed, 02
+ May 2012) | 33 lines Multiple revisions 365006,365068 ........
+ r365006 | twilson | 2012-05-02 10:49:03 -0500 (Wed, 02 May 2012)
+ | 12 lines Fix a CEL LINKEDID_END race and local channel
+ linkedids This patch has the ;2 channel inherit the linkedid of
+ the ;1 channel and fixes the race condition by no longer scanning
+ the channel list for "other" channels with the same linkedid.
+ Instead, cel.c has an ao2 container of linkedid strings and uses
+ the refcount of the string as a counter of how many channels with
+ the linkedid exist. Not only does this eliminate the race
+ condition, but it also allows us to look up the linkedid by the
+ hashed key instead of traversing the entire channel list. Review:
+ https://reviewboard.asterisk.org/r/1895/ ........ r365068 |
+ twilson | 2012-05-02 12:02:39 -0500 (Wed, 02 May 2012) | 11 lines
+ Don't leak a ref if out of memory and can't link the linkedid If
+ the ao2_link fails, we are most likely out of memory and bad
+ things are going to happen. Before those bad things happen, make
+ sure to clean up the linkedid references. This patch also adds a
+ comment explaining why linkedid can't be passed to both local
+ channel allocations and combines two ao2_ref calls into 1.
+ Review: https://reviewboard.asterisk.org/r/1895/ ........ Merged
+ revisions 365006,365068 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions
+ 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083
+ from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-02 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.4.0 Released.
+
+2012-05-01 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.4.0-rc3 Released.
+
+ * channels/chan_sip.c: Revert revision 360862
+
+ 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)
+
+2012-04-24 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.4.0-rc2 Released.
+
+ * AST-2012-004
+
+ * AST-2012-005
+
+ * AST-2012-006
+
+2012-04-04 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.4.0-rc1 Released.
+
+2012-04-04 16:38 +0000 [r361091-361143] Jonathan Rose <jrose@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@digium.com>
+
+ * /, channels/chan_sip.c: Make transfer not ignore port information
+ with SIP. Attempting to transfer with SIP to an address like
+ 1XXXXX@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@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@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@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@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@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@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@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@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@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@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
+ 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/
+ ........ Merged revisions 357940 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-02 18:37 +0000 [r357895] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /, channels/sig_pri.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 ........ Merged revisions 357894 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-02 15:59 +0000 [r357812] Sean Bright <sean@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." ........ Merged revisions
+ 357811 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-02 15:50 +0000 [r357810] Richard Mudgett <rmudgett@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) ........ Merged
+ revisions 357809 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-02 01:05 +0000 [r357762] Mark Michelson <mmichelson@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 ........ Merged revisions
+ 357761 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-01 14:18 +0000 [r357667] Kinsey Moore <kmoore@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) ........
+ Merged revisions 357665 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-29 20:39 +0000 [r357576-357620] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * include/asterisk/stringfields.h, main/utils.c: Update stringfield
+ documentation for removed second va_list in favor of va_copy. In
+ r320946, the second va_list that was passed to
+ ast_string_field_build_va and friends, was removed. This patch
+ updates the documentation to reflect that.
+
+ * 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 ........ Merged revisions 357575
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-28 22:29 +0000 [r357458-357497] Jonathan Rose <jrose@digium.com>
+
+ * /, configs/sip.conf.sample, UPGRADE-1.8.txt: 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) ........ Merged revisions 357490
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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) ........ Merged
+ revisions 357455 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-28 21:21 +0000 [r357421] Tilghman Lesher <tilghman@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. ........ Merged revisions 357416 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-28 20:58 +0000 [r357408] Richard Mudgett <rmudgett@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 ........ Merged revisions 357407 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-28 20:42 +0000 [r357357-357405] Jonathan Rose <jrose@digium.com>
+
+ * UPGRADE.txt: revision 357386 -- oops, accidentally made it 10.3
+ to 10.4 instead of 10.2 to 10.3 (issue ASTERISK-19352) reported
+ by: jamicque
+
+ * /, UPGRADE.txt, UPGRADE-1.8.txt: Moves UPGRADE.txt notes from
+ r357356 to a new section specific to 1.8.12 (issue
+ ASTERISK-19352) reported by: jamicque ........ Merged revisions
+ 357386 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, UPGRADE-1.8.txt: Adds UPGRADE.txt notes to r357266 indicating
+ changes to transport option (issue ASTERISK-19352) Reported by:
+ jamicque ........ Merged revisions 357356 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-28 19:35 +0000 [r357353] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_page.c: Remove dupliate 'i' option table entry in
+ app_page.c. (closes issue ASTERISK-19310) Reported by: Makoto Dei
+ Patches: app_page-duplicate-i-option.patch (license #5027) patch
+ uploaded by Makoto Dei ........ Merged revisions 357352 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-28 18:51 +0000 [r357318] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/security_events.c: Add a security event for the case
+ where fake authentication challenge is sent.
+
+2012-02-28 18:11 +0000 [r357271] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Changes transport option in sip.conf so
+ that using multiple instances doesn't stack. Prior to this patch,
+ Using "transport=" multiple times would cause them to add to one
+ another like allow/deny. This patch changes that behavior to
+ simply use the transport option specified last. Also, if no
+ transport option is applied now, the default will automatically
+ be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches:
+ asterisk-19352-transport-warning-message-v1.patch uploaded by
+ Michael L. Young (license 5026)
+ issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes
+ (license 5674) Review:
+ https://reviewboard.asterisk.org/r/1745/diff/#index_header
+ ........ Merged revisions 357266 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-28 14:46 +0000 [r357213] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, Makefile.rules: Make COMPILE_DOUBLE magic actually work. The
+ build system has some special magic to ensure that if Asterisk is
+ built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the
+ source is still compiled with the optimizer enabled (even though
+ the result will be thrown away), because the compiler is able to
+ find a great deal of coding errors and bugs as a result of
+ running its optimizers. Unfortunately at some point this mode got
+ broken, and the 'throwaway' compile of the code was no longer
+ done with the optimizer enabled. This patch corrects that
+ problem. ........ Merged revisions 357212 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-29 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.3.0 Released.
+
+2012-03-26 Asterisk Development Team <asteriskteam@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ autoconf/ast_c_declare_check.m4 (added), configure.ac,
+ formats/format_ogg_vorbis.c: Fix compile problem when old version
+ of libvorbisfile v1.1.2 is used. The principle difference between
+ libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition
+ of the predefined callbacks OV_CALLBACKS_xxx in
+ vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the
+ configure script to detect if libvorbisfile.h declares
+ OV_CALLBACKS_NOCLOSE. * Copied the declaration of
+ OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile.
+ (closes issue ASTERISK-19370) Reported by: Jonn Taylor ........
+ Merged revisions 355608 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-16 20:01 +0000 [r355623] Sean Bright <sean@malleable.com>
+
+ * /, main/audiohook.c: Revert a change to
+ audio_audiohook_write_list that had no affect. When I made this
+ change initially, I was under the false impression that the
+ audiohooks structure remained on the channel after all of the
+ hooks had been detached. This is not the case, ast ast_read takes
+ care of removing the audiohooks structure if the lists are empty.
+ ........ Merged revisions 355622 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-17 19:06 +0000 [r355733] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Fix regressions with regards to route-set
+ creation on early dialogs. This fixes two main issues: 1.
+ Asterisk would send a CANCEL to the route created by the
+ provisional response instead of using the same destination it did
+ in the initial INVITE. 2. If a new route set arrives in a 200 OK
+ than was in the 1XX response (perfectly possible if our outbound
+ INVITE gets forked), then the route set in the 200 OK needs to
+ overwrite the route set in the 1XX response. (closes issue
+ ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten
+ Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson
+ (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt
+ (license 6034) Review: https://reviewboard.asterisk.org/r/1749
+ ........ Merged revisions 355732 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-17 19:34 +0000 [r355794-355747] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_iax2.c: Pass the correct value to
+ ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq
+ variable to determine how often to send trunk packets, but this
+ value is in milliseconds while ast_timer_set_rate() expects the
+ rate argument to be ticks per second. So we divide 1000 by
+ trunkfreq and pass that in instead. With a default of 20ms, this
+ change makes IAX2 send trunk packets every 20ms instead of every
+ 50ms. Tracked down by myself and Bob Wienholt. ........ Merged
+ revisions 355746 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * configs/iax.conf.sample, /, channels/chan_iax2.c: Don't allow
+ trunkfreq to be greater than 1000ms. ........ Merged revisions
+ 355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-18 07:58 +0000 [r355851] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_ss7.h, /, channels/sig_analog.h, channels/sig_pri.c,
+ channels/sig_ss7.c: push 'outgoing' flag from sig_XXX up to
+ chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept
+ in sync, particulary FXS ast_hangup didn't clear the 'outgoing'
+ flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync.
+ Now provides a callback for all the low level sig_XXX modules.
+ (issue ASTERISK-19316) alecdavis (license 585) Reported by:
+ Jeremy Pepper Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1747/ ........ Merged
+ revisions 355850 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-18 17:02 +0000 [r355896-355895] Paul Belanger <pabelanger@digium.com>
+
+ * /: Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)
+ ........ Merged revisions 355839 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /: Revert commit
+
+2012-02-19 17:50 +0000 [r355998-355902] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_iax2.c: Set the length of the ast_sockaddr, so
+ that we can set it's port later. Without this, the call to
+ ast_sockaddr_set_port a few lines later is a noop. ........
+ Merged revisions 355901 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: Add some boilerplate documentation for
+ IAXVAR and IAXPEER. ........ Merged revisions 355904 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_dahdi.c, /: Change some debug messages from
+ LOG_DEBUG to ast_debug. ........ Merged revisions 355949 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_dahdi.c, /: This was a LOG_NOTICE, so roll it back.
+ ........ Merged revisions 355952 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: Remove spurious warning when
+ 'qualifyfreqnotok' is set successfully. (closes issue
+ ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright
+ Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John
+ Covert (license 5512) ........ Merged revisions 355997 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-21 04:30 +0000 [r356074] Kinsey Moore <kmoore@digium.com>
+
+ * main/ccss.c: Add missing newline to ccss state change
+ notification Move along, nothing to see here...
+
+2012-02-21 11:17 +0000 [r356108] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_iax2.c: Make 'iax2 show callnumber usage' output
+ make sense when an IP is passed in. ........ Merged revisions
+ 356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-22 14:53 +0000 [r356215] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 356214 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012)
+ | 27 lines Fix potential buffer overrun and memory leak when
+ executing "sip show peers" The "sip show peers" command uses a
+ fix sized array to sort the current peers in the peers
+ ao2_container. The size of the array is based on the current
+ number of peers in the container. However, once the size of the
+ array is determined, the number of peers in the container can
+ change, as the peers container is not locked. This could cause a
+ buffer overrun when populating the array, if peers were added to
+ the container after the array was created. Additionally, a memory
+ leak of the allocated array would occur if a user caused the
+ _show_peers method to return CLI_SHOWUSAGE. We now create a
+ snapshot of the current peers using an ao2_callback with the
+ OBJ_MULTIPLE flag. This size of the array is set to the number of
+ peers that the iterator will iterate over; hence, if peers are
+ added or removed from the peers container it will not affect the
+ execution of the "sip show peers" command. Review:
+ https://reviewboard.asterisk.org/r/1738/ (closes issue
+ ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas
+ Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey
+ Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan
+ (license 6283) ........
+
+2012-02-22 21:18 +0000 [r356297] Terry Wilson <twilson@digium.com>
+
+ * main/loader.c, res/res_calendar.c, /,
+ include/asterisk/calendar.h: Track module use count for
+ res_calendar If the res_calendar module was followed immediately
+ by one of the calendar tech modules and "core stop gracefully"
+ was run, Asterisk would crash. This patch adds use count tracking
+ for res_calendar so that it is unloaded after the tech modules
+ when shutting down gracefully. It is now not possible to unload
+ all the of the calendar modules via "module unload
+ res_calednar.so", but it is still possible to unload them all via
+ "module unload -h res_calendar.so". Review:
+ https://reviewboard.asterisk.org/r/1752/ ........ Merged
+ revisions 356291 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-23 03:23 +0000 [r356431-356428] Paul Belanger <pabelanger@digium.com>
+
+ * /, apps/app_rpt.c: Multiple revisions 356290,356335,356337
+ ........ r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed,
+ 22 Feb 2012) | 4 lines Fix -Werror=unused-but-set-variable
+ compiler error (gcc 4.6.2) Review:
+ https://reviewboard.asterisk.org/r/1763/ ........ r356335 |
+ pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2
+ lines Add back strsep() function for previous commit ........
+ r356337 | pabelanger | 2012-02-22 16:36:37 -0500 (Wed, 22 Feb
+ 2012) | 2 lines Missed one strsep() function ........ Merged
+ revisions 356290,356335,356337 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * addons/chan_ooh323.c, /: Fix -Werror=unused-but-set-variable
+ compiler error (gcc 4.6.2) ........ Merged revisions 356430 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-23 15:40 +0000 [r356476] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Fix ACK routing for non-2xx responses.
+ When we send an ACK for a 2xx response to an INVITE, we are
+ supposed to use the learned route set. However, when we receive a
+ non-2xx final response to an INVITE, we are supposed to send the
+ ACK to the same place we initially sent the INVITE. We had been
+ doing this up until the changes went in that would build a route
+ set from provisional responses. That introduced a regression
+ where we would use the learned route set under all circumstances.
+ With this change, we now will set the destination of our ACK
+ based on the invitestate. If it is INV_COMPLETED then that means
+ that we have received a non-2xx final response (INV_TERMINATED
+ indicates a 2xx response was received). If it is INV_CANCELLED,
+ then that means the call is being canceled, which means that we
+ should be ACKing a 487 response. The other change introduced here
+ is setting the invitestate to INV_CONFIRMED when we send an ACK
+ *after* the reqprep instead of before. This way, we can tell in
+ reqprep more easily what the invitestate is prior to sending the
+ ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer
+ patches: ASTERISK-19389v2.patch uploaded by Mark Michelson
+ (license #5049) (with some slight modifications prior to commit)
+ ........ Merged revisions 356475 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-23 19:52 +0000 [r356522] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c, main/features.c: Fix blind transfer
+ parking issues if the dialed extension is not recognized as a
+ parking extension. Custom parking extensions may not be coded
+ such that the first and only extension priority is the Park
+ application. These custom parking extensions will not be
+ recognized as parking extensions. When a call is blind
+ transferred to an extension that is not recognized as a parking
+ extension, the normal blind transfer code causes the transferred
+ channel to start executing dialplan. Calls that get parked in
+ this manner do not know the original channel name that parked the
+ call so the original parker could never be called back if the
+ parked call is not retrieved before the timeout time. The parking
+ space is also announced to the call being parked as a side effect
+ of not knowing the original parking channel. * Fix handling of
+ BLINDTRANSFER channel variable for call parking. * Fixed SIP
+ blind transfer using the wrong dialplan context variable to check
+ for the parking extension. (closes issue ASTERISK-19322) Reported
+ by: aragon Tested by: rmudgett, jparker Review:
+ https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 ........
+ Merged revisions 356521 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-24 15:07 +0000 [r356651-356605] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_srtp.c, channels/sip/sdp_crypto.c,
+ include/asterisk/res_srtp.h, main/rtp_engine.c, /,
+ include/asterisk/rtp_engine.h: Allow SRTP policies to be reloaded
+ Currently, when using res_srtp, once the SRTP policy has been
+ added to the current session the policy is locked into place. Any
+ attempt to replace an existing policy, which would be needed if
+ the remote endpoint negotiated a new cryptographic key, is
+ instead rejected in res_srtp. This happens in particular in
+ transfer scenarios, where the endpoint that Asterisk is
+ communicating with changes but uses the same RTP session. This
+ patch modifies res_srtp to allow remote and local policies to be
+ reloaded in the underlying SRTP library. From the perspective of
+ users of the SRTP API, the only change is that the adding of
+ remote and local policies are now added in a single method call,
+ whereas they previously were added separately. This was changed
+ to account for the differences in handling remote and local
+ policies in libsrtp. Review:
+ https://reviewboard.asterisk.org/r/1741/ (closes issue
+ ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas
+ Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt
+ Jordan (license 6283) (with some small modifications for this
+ check-in) ........ Merged revisions 356604 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * res/res_srtp.c, /: Remove srtp_shutdown from res_srtp The patch
+ for ASTERISK-19253 included properly shutting down the libsrtp
+ library in the case of module unload. Unfortunately, not all
+ distributions have the srtp_shutdown call. As such, this patch
+ removes calling srtp_shutdown. ........ Merged revisions 356650
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-24 18:27 +0000 [r356690] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c, include/asterisk/tcptls.h,
+ channels/sip/include/sip.h: Fix worker thread resource leak in
+ SIP TCP/TLS. The SIP TCP/TLS worker threads were created joinable
+ but noone could join them if they died on their own. * Fix the
+ SIP TCP/TLS worker threads to not be created joinable. *
+ _sip_tcp_helper_thread() only needs one parameter since the pvt
+ parameter is only passed in as NULL and never used. (closes issue
+ ASTERISK-19203) Reported by: Steve Davies Review:
+ https://reviewboard.asterisk.org/r/1714/ ........ Merged
+ revisions 356677 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-25 17:22 +0000 [r356798] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c, /: Fix crash in app_voicemail during
+ close_mailbox In r354890, a memory leak in app_voicemail was
+ fixed by properly disposing of the allocated heard/deleted
+ pointers. However, there are situations, particularly when no
+ messages are found in a folder, where these pointers are not
+ allocated and not NULL. In that case, an invalid free would be
+ attempted, which could crash app_voicemail. As there are a number
+ of code paths where this could occur, this patch uses the number
+ of messages detected in the folder before it attempts to free the
+ pointers. This resolves the crash detected in the Asterisk Test
+ Suite's check_voicemail_nominal test. ........ Merged revisions
+ 356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-27 15:30 +0000 [r356961] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_odbc.c: Remove possible segfaults from res_odbc by
+ adding locks around usage of odbc handle (closes issue
+ ASTERISK-19011) Reported by: Walter Doekes Patches:
+ issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch
+ uploaded by Walter Doekes (license 5674) review:
+ https://reviewboard.asterisk.org/r/1719/ review:
+ https://reviewboard.asterisk.org/r/1622/ ........ Merged
+ revisions 356917 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-27 16:05 +0000 [r356964] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Copy CDR variables when set during a bridge
+ This patch makes sure amaflags, accountcode, and userfield get
+ copied to the bridge CDR when set during a bridge (like via a
+ custom feature). (closes issue ASTERISK-16990) Review:
+ https://reviewboard.asterisk.org/r/1721/ ........ Merged
+ revisions 356963 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-27 23:36 +0000 [r357095] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Fix callerid of Originated calls. Thanks to
+ Matt Riddell for tracking this down. (closes issue
+ ASTERISK-19385) Reported by: ornix ........ Merged revisions
+ 357093 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-05 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.2.0 Released.
+
+2012-03-01 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.2.0-rc4 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.
+
+2012-02-28 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.2.0-rc3 Released.
+
+ * main/channel.c: Fix callerid of Originated calls.
+
+ The callerid of originated calls (independent of mechanism) was not
+ being passed to the outbound channel. This patch fixes that. Thanks
+ to Matt Riddell for tracking this down.
+ (closes issue ASTERISK-19385)
+ Reported by: ornix
+
+ * channels/chan_sip.c: Fix ACK routing for non-2xx responses.
+
+ When we send an ACK for a 2xx response to an INVITE, we are supposed
+ to use the learned route set. However, when we receive a non-2xx
+ final response to an INVITE, we are supposed to send the ACK to the
+ same place we initially sent the INVITE.
+
+ We had been doing this up until the changes went in that would build
+ a route set from provisional responses. That introduced a regression
+ where we would use the learned route set under all circumstances.
+
+ With this change, we now will set the destination of our ACK based on
+ the invitestate. If it is INV_COMPLETED then that means that we have
+ received a non-2xx final response (INV_TERMINATED indicates a 2xx
+ response was received). If it is INV_CANCELLED, then that means the
+ call is being canceled, which means that we should be ACKing a 487
+ response.
+
+ The other change introduced here is setting the invitestate to
+ INV_CONFIRMED when we send an ACK *after* the reqprep instead of
+ before. This way, we can tell in reqprep more easily what the
+ invitestate is prior to sending the ACK.
+
+ (closes issue ASTERISK-19389)
+ reported by Karsten Wemheuer
+ patches:
+ ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049)
+
+ * channels/chan_sip.c: Fix regressions with regards to route-set
+ creation on early dialogs.
+
+ This fixes two main issues:
+ 1. Asterisk would send a CANCEL to the route created by the provisional
+ response instead of using the same destination it did in the initial
+ INVITE.
+ 2. If a new route set arrives in a 200 OK than was in the 1XX response
+ (perfectly possible if our outbound INVITE gets forked), then the
+ route set in the 200 OK needs to overwrite the route set in the 1XX
+ response.
+ (closes issue ASTERISK-19358)
+ Reported by: Karsten Wemheuer
+ Tested by: Karsten Wemheuer
+ patches:
+ ASTERISK-19358.patch uploaded by Mark Michelson (license 5049)
+ ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034)
+
+ Review: https://reviewboard.asterisk.org/r/1749
+
+
+2012-02-10 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.2.0-rc2 Released.
+
+ * channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric
+ codes. In ASTERISK-18924, SIP INFO DTMF handling was 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 against '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
+
+ * apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce from
+ uninitialized caller_id storage (closes issue ASTERISK-19311)
+ Reported by: tootai
+ Tested by: rmudgett
+
+ * 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/
+
+ * 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
+
+ * channels/chan_sip.c,include/asterisk/dnsmgr.h,main/dnsmgr.c: 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.
+
+2012-02-01 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.2.0-rc1 Released.
+
+ * Test results:
+ http://bamboo.asterisk.org/browse/TESTING-ASTERISK1020RCS-2
+
+2012-01-30 12:48 +0000 [r353261] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Clarify log WARNING message when
+ port-zero SDP 'm' lines received. Previously, if an m-line in an
+ SDP offer or answer had a port number of zero, that line was
+ skipped, and resulted in an 'Unsupported SDP media type...'
+ warning message. This was misleading, as the media type was not
+ unsupported, but was ignored because the m-line indicated that
+ the media stream had been rejected (in an answer) or was not
+ going to be used (in an offer). ........ Merged revisions 353260
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-29 02:44 +0000 [r353176] Russell Bryant <russell@russellbryant.com>
+
+ * main/netsock.c, /: Find even more network interfaces. The
+ previous change made the code look for emN and pciN in addition
+ to what it did originally, which was search for ethN. However, it
+ needed to be looking for pciN#N, so that's what it does now. This
+ also moves the memset() to be before every ioctl(). ........
+ Merged revisions 353175 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-28 14:51 +0000 [r353127] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/rtp_engine.c, /: Add 'L16-256' MIME subtype alias for
+ slin16. Asterisk has supported the 'L16' MIME subtype for 16kHz
+ signed linear (PCM) audio for quite some time, but some endpoints
+ refer to it as 'L16-256'. This commit adds this as an alias for
+ the existing format. ........ Merged revisions 353126 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-28 04:27 +0000 [r353078] Russell Bryant <russell@russellbryant.com>
+
+ * main/netsock.c, /: Update ast_set_default_eid() to find more
+ network interfaces. As of Fedora 15, ethN is not the name of
+ ethernet interfaces. The names are emN or pciN. Update some code
+ that searched for interfaces named ethN to look for the new
+ names, as well. For more information about why this change was
+ made, see this page: http://domsch.com/blog/?p=455 ........
+ Merged revisions 353077 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-27 21:37 +0000 [r352992-353039] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c: Audit of ao2_iterator_init() usage for v10.
+ Missed one.
+
+ * tests/test_format_api.c: Audit of ao2_iterator_init() usage for
+ v10. Fix double format_cap iterator cleanup.
+
+2012-01-27 19:19 +0000 [r352965] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_monitor.c: Make failed PauseMonitor and UnpauseMonitor
+ with no valid channel not close AMI session. I also went ahead
+ and took a little time to make sure that the manager value
+ AMI_SUCCESS was used instead of just return 0 being thrown around
+ everywhere since that's how we handle this stuff these days.
+ (closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches:
+ res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey
+ (license 5766) ........ Merged revisions 352959 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-27 18:36 +0000 [r352956] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_srtp.c, main/pbx.c, /, channels/chan_sip.c,
+ include/asterisk/indications.h, res/snmp/agent.c,
+ main/taskprocessor.c, apps/app_queue.c, channels/chan_iax2.c,
+ apps/app_chanspy.c, main/indications.c, res/res_odbc.c: Audit of
+ ao2_iterator_init() usage for v1.8. Fixes numerous reference
+ leaks and missing ao2_iterator_destroy() calls as a result.
+ Review: https://reviewboard.asterisk.org/r/1697/ ........ Merged
+ revisions 352955 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-27 00:08 +0000 [r352863] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
+ revisions 352862 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r352862 | alecdavis | 2012-01-27 13:05:30 +1300 (Fri, 27 Jan
+ 2012) | 12 lines rfc4235 - Section 4.1: Versions MUST be
+ representable using a non-negative 32 bit integer. If a BLF
+ subscription exists for long enough, using %d may print negative
+ version numbers. Unlikely, as 2^32 at 1 update per second is ~137
+ years, or half that before the versions number started going
+ negative. Tested with Asterisk 1.8.8.2 with Grandstream phones.
+ alecdavis (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1694/ ........
+
+2012-01-26 20:22 +0000 [r352817] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: Fix outbound DTMF for inband mode (tell
+ asterisk core to generate DTMF sounds). (Closes issue
+ ASTERISK-19233) Reported by: Matt Behrens Patches:
+ chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)
+ ........ Merged revisions 352807 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-26 19:07 +0000 [r352756] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Copy amaflags to sip_pvt from peer during
+ create_addr_from_peer For whatever reason, we don't have a single
+ function for copying data like this from SIP peers to the SIP
+ pvt. This patch adds the copying of amaflags to the sip_pvt, but
+ it would probably be worth discussing this function along with
+ the others that essentially just copy some amount of data from a
+ peer to a private. (Closes issue ASTERISK-19029) Reported by:
+ Matt Lehner ........ Merged revisions 352755 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-26 06:33 +0000 [r352705] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_sip.c: Merged revisions 352704 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r352704 | alecdavis | 2012-01-26 19:27:07 +1300 (Thu, 26 Jan
+ 2012) | 20 lines Cleanup dialog-info+xml Notify dialog Make
+ similar to other Notify messages. sample output: <?xml
+ version="1.0"?> <dialog-info
+ xmlns="urn:ietf:params:xml:ns:dialog-info" version="715"
+ state="full" entity="sip:8523@192.168.x.xx"> <dialog id="8523">
+ <state>terminated</state> </dialog> </dialog-info> Tested with
+ Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
+ Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1693/ ........
+
+2012-01-25 22:23 +0000 [r352651] Paul Belanger <pabelanger@digium.com>
+
+ * apps/app_voicemail.c, /: Fix -Werror=unused-but-set-variable
+ compiler error (gcc 4.6.2) ........ Merged revisions 352643 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-25 21:18 +0000 [r352616] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/test.c: Avoid unnecessary rebuilds of main/test.c.
+ main/test.c includes "asterisk/version.h", when it should include
+ "asterisk/ast_version.h" instead (and it should use the
+ ast_get_version() and ast_get_version_num() functions). This
+ commit modifies it to extract the Asterisk version information
+ using the proper APIs, and as a result means that main/test.c no
+ longer needs to be rebuilt when a Subversion checkout is updated
+ or modified. ........ Merged revisions 352612 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-25 17:30 +0000 [r352556] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Remove some extraneous debugging from
+ registry memleak fix ........ Merged revisions 352551 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-25 17:16 +0000 [r352520] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c, CHANGES, main/message.c,
+ channels/sip/include/sip.h: Fixes for sending SIP MESSAGE outside
+ of calls. * Fix authenticate MESSAGE losing custom headers added
+ by the MESSAGE_DATA function in the authorization attempt. * Pass
+ up better From header contents for SIP to use. Now is in the
+ "display-name" <URI> format expected by MessageSend. (Note that
+ this is a behavior change that could concievably affect some
+ people.) * Block user from adding standard headers that are added
+ automatically. (To, From,...) * Allow the user to override the
+ Content-Type header contents sent by MessageSend. * Decrement
+ Max-Forwards header if the user transferred it from an incoming
+ message. * Expand SIP short header names so the dialplan and
+ other code only has to deal with the full names. * Documents what
+ SIP expects in the MessageSend(from) parameter. (closes issue
+ ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917)
+ Reported by: Shaun Clark Review:
+ https://reviewboard.asterisk.org/r/1683/
+
+2012-01-25 16:54 +0000 [r352516] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/format.c, main/format_cap.c, main/format_pref.c: Eliminate
+ unnecessary rebuilds of main/format*.c. These files have no need
+ to include "asterisk/version.h", and doing so forces them to be
+ rebuilt each time a Subversion checkout moves between 'modified'
+ and 'unmodified' states.
+
+2012-01-25 16:49 +0000 [r352515] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Clean up some SIP registry-related memory
+ leaks 1) Be sure and free at unload the epa_backend we allocate
+ at startup 2) Do the same sip_registry cleanup at unload we do at
+ reload Review: https://reviewboard.asterisk.org/r/1689/ ........
+ Merged revisions 352514 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-25 16:41 +0000 [r352512] Jonathan Rose <jrose@digium.com>
+
+ * /, configs/sip.conf.sample: Redocuments sip types peer, user,
+ friend in sip.conf.sample There was faulty information in the
+ sample config describing user as a synonym for friend so it has
+ been changed to better elaborate on the differences between the
+ three entity types. (closes issue ASTERISK-15537) Reported by:
+ yarique ........ Merged revisions 352511 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-24 22:22 +0000 [r352430] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Don't do a DNS lookup on an outbound
+ REGISTER host if there is an outbound proxy configured. (closes
+ issue ASTERISK-16550) reported by: Olle Johansson ........ Merged
+ revisions 352424 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-24 20:35 +0000 [r352373] Jonathan Rose <jrose@digium.com>
+
+ * /, sounds/Makefile: Set core sounds version to 1.4.22. Now that
+ we have the right license for the Russian 1.4.22 sounds as well
+ as the sounds for the Australian English 1.4.22 sounds, we can
+ finally set the sounds to use 1.4.22! (closes issue
+ ASTERISK-18978) Reported by: Cameron Twomey Patches:
+ confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002
+ uploaded by Cameron Twomey ........ Merged revisions 352367 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-24 17:02 +0000 [r352292] Richard Mudgett <rmudgett@digium.com>
+
+ * /, funcs/func_odbc.c: Fix locking issues with channel datastores
+ in func_odbc.c. * Fixed a potential memory leak when an existing
+ datastore is manually destroyed by inline code instead of calling
+ ast_datastore_free(). (closes issue ASTERISK-17948) Reported by:
+ Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/
+ ........ Merged revisions 352291 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-23 20:30 +0000 [r352228-352231] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/features.c: Fix grammar of comment. ........ Merged
+ revisions 352230 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/features.c: Fix blind transfers from failing if an 'h'
+ extension is present. This prevents the 'h' extension from being
+ run on the transferee channel when it is transferred via a native
+ transfer mechanism such as SIP REFER. (closes ASTERISK-19173)
+ Reported by: Ross Beer Tested by: Kristjan Vrban Patches:
+ ASTERISK-19173 by Mark Michelson (license 5049) Review:
+ https://reviewboard.asterisk.org/r/1685 ........ Merged revisions
+ 352199 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-23 19:18 +0000 [r352149] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_fax_spandsp.c: Correctly apply FAXOPT settings (V17,
+ V27, V29) before starting spandsp layer While the FAXOPT function
+ could be used to set the modem capabilities, the input to that
+ function was not being applied correctly to the spandsp layer.
+ This patch applies the current model capabilities before starting
+ the spandsp layer. (closes issue: ASTERISK-16409) Reported by:
+ Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson
+ Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license
+ 5081) spandsp-modems-10.diff uploaded by mnicholson (license
+ 5081) ........ Merged revisions 352144 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-23 17:34 +0000 [r352091] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Fix sip_cfg.notifycid to be set with the
+ defined enum values. The invalid value used when notifycid was
+ enabled was benign. As far as the code was concerned -1 and 1 are
+ equivalent. (closes issue ASTERISK-19232) Reported by: Eike
+ Kuiper ........ Merged revisions 352090 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-21 00:21 +0000 [r352035] Richard Mudgett <rmudgett@digium.com>
+
+ * /, funcs/func_timeout.c, main/app.c: Fix ast_app_dtget() time
+ unit inconsistency. Note: Noone calls ast_app_dtget() with the
+ timeout parameter of zero so the bad code normally will never get
+ executed. * Fix unnecessary floating point division in
+ func_timeout.c timeout_write() when all other values are
+ integers. (closes issue ASTERISK-16817) Reported by: Dmitry
+ Andrianov ........ Merged revisions 352029 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-21 00:08 +0000 [r352015-352017] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Remove XXX comment that is not necessary.
+ ........ Merged revisions 352016 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Fix RTP reference leak. If a blind
+ transfer were initiated using a REFER without a prior reINVITE to
+ place the call on hold, AND if Asterisk were sending RTCP
+ reports, then there was a reference for the RTP instance of the
+ transferer. This fixes the issue by merging two similar but
+ slightly conflicting sections of code into a single area. It also
+ adds a stop_media_flows() call in the case that the transferer's
+ UA never sends a BYE to us like it is supposed to. (issue
+ ASTERISK-19192) Review: https://reviewboard.asterisk.org/r/1681/
+ ........ Merged revisions 352014 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-20 19:35 +0000 [r351816-351861] Kinsey Moore <kmoore@digium.com>
+
+ * /, codecs/ilbc/iLBC_test.c: More corrections for the ilbc code
+ These changes are in a file that is not compiled by default, and
+ so were missed on earlier checks. ........ Merged revisions
+ 351860 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /: Recorded merge of revisions 351858 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Allow
+ ilbc code to build under dev mode GCC 4.6.3 found some set/unused
+ variables in the ILBC code.
+
+ * codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Restore
+ LSF_check function calls from set/unused variable removal These
+ functions are not noops and modify the array that is passed in.
+ Thanks for the catch Richard.
+
+ * codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Remove more
+ set, but unused variables in the ilbc codec GCC 4.6.3 caught
+ these in dev mode as well.
+
+2012-01-20 15:59 +0000 [r351762] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Adds setting of mwi_from field to
+ check_auth_result check_peer_ok (closes ASTERISK-19057) Reported
+ By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license
+ 5242) ........ Merged revisions 351759 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-20 15:54 +0000 [r351761] Matthew Jordan <mjordan@digium.com>
+
+ * codecs/ilbc/helpfun.c, /: Remove unused variable 'tmp' from
+ helpfun in ilbc codec gcc version 4.6.2 caught an unused variable
+ in the ilbc codec library. This would prevent compilation with
+ --enable-dev-mode; variable removed. ........ Merged revisions
+ 351760 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-20 13:01 +0000 [r351708] Stefan Schmidt <sst@sil.at>
+
+ * /, contrib/asterisk-ng-doxygen: enable doxygen build for files in
+ the channels/sip folder like reqresp_parser.c ........ Merged
+ revisions 351707 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-19 23:25 +0000 [r351646] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Misc minor
+ fixes in reqresp_parser.c and chan_sip.c. * Fix corner cases in
+ get_calleridname() parsing and ensure that the output buffer is
+ nul terminated. * Make get_calleridname() truncate the name it
+ parses if the given buffer is too small rather than abandoning
+ the parse and not returning anything for the name. Adjusted
+ get_calleridname_test() unit test to handle the truncation
+ change. * Fix get_in_brackets_test() unit test to check the
+ results of get_in_brackets() correctly. * Fix
+ parse_name_andor_addr() to not return the address of a local
+ buffer. This function is currently not used. * Fix potential NULL
+ pointer dereference in sip_sendtext(). * No need to
+ memset(calleridname) in check_user_full() or tmp_name in
+ get_name_and_number() because get_calleridname() ensures that it
+ is nul terminated. * Reply with an accurate response if
+ get_msg_text() fails in receive_message(). This is academic in
+ v1.8 because get_msg_text() can never fail. ........ Merged
+ revisions 351618 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-19 22:43 +0000 [r351612] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Correct output of RTCP jitter
+ statistics in SR and RR reports Change the RTCP RR and SR
+ generation code to convert Asterisk's internal jitter statistics
+ to be represented in RTP timestamp units based on the rate of the
+ codec in use instead of in seconds. (closes issue ASTERISK-14530)
+ ........ Merged revisions 351611 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-19 21:47 +0000 [r351560] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c, include/asterisk/netsock2.h: Eliminates
+ doubling the :port part of SIP Notify Message-Account headers.
+ This patch prevents the domain string from getting mangled during
+ the initreqprep step by moving the initialization to before its
+ immediate use. It also documents this pitfall for the
+ ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported
+ by: Yuri Review: https://reviewboard.asterisk.org/r/1678/
+ ........ Merged revisions 351559 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-19 21:12 +0000 [r351505] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Prevent crash when an SDP offer is
+ received with an encrypted video stream when support for video is
+ disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
+ Reported by: Catalin Sanda ........ Merged revisions 351504 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-18 21:05 +0000 [r351451] Matthew Jordan <mjordan@digium.com>
+
+ * codecs/ilbc/helpfun.c (added), codecs/ilbc/LICENSE_ADDENDUM
+ (added), codecs/ilbc/doCPLC.c (added), codecs/ilbc/anaFilter.c
+ (added), codecs/ilbc/helpfun.h (added), codecs/ilbc/createCB.c
+ (added), codecs/ilbc/doCPLC.h (added), codecs/ilbc/anaFilter.h
+ (added), codecs/ilbc/constants.c (added),
+ codecs/ilbc/iLBC_decode.c (added), codecs/ilbc/createCB.h
+ (added), codecs/ilbc/constants.h (added),
+ codecs/ilbc/iLBC_decode.h (added), codecs/ilbc/iCBSearch.c
+ (added), codecs/ilbc/filter.c (added), codecs/ilbc/hpInput.c
+ (added), codecs/ilbc/gainquant.c (added), codecs/ilbc/iCBSearch.h
+ (added), codecs/ilbc/hpOutput.c (added), codecs/ilbc/rfc3951.txt
+ (added), codecs/ilbc/filter.h (added), codecs/ilbc/hpInput.h
+ (added), codecs/ilbc/LPCencode.c (added), codecs/ilbc/gainquant.h
+ (added), codecs/codec_ilbc.c, codecs/ilbc/hpOutput.h (added),
+ codecs/ilbc/StateSearchW.c (added), codecs/ilbc/PATENTS (added),
+ contrib/scripts/get_ilbc_source.sh, codecs/ilbc/LPCencode.h
+ (added), codecs/ilbc/LICENSE (added), codecs/ilbc/StateSearchW.h
+ (added), codecs/ilbc/iCBConstruct.c (added),
+ codecs/ilbc/syntFilter.c (added), /, codecs/ilbc/iCBConstruct.h
+ (added), codecs/ilbc/iLBC_test.c (added),
+ codecs/ilbc/syntFilter.h (added), codecs/ilbc/StateConstructW.c
+ (added), codecs/ilbc/packing.c (added),
+ codecs/ilbc/StateConstructW.h (added), codecs/ilbc/packing.h
+ (added), codecs/ilbc/getCBvec.c (added), codecs/ilbc/LPCdecode.c
+ (added), codecs/ilbc/enhancer.c (added), codecs/ilbc/lsf.c
+ (added), codecs/ilbc/iLBC_encode.c (added),
+ codecs/ilbc/getCBvec.h (added), codecs/ilbc/LPCdecode.h (added),
+ codecs/ilbc/enhancer.h (added), codecs/ilbc/FrameClassify.c
+ (added), codecs/ilbc/iLBC_define.h (added), codecs/ilbc/lsf.h
+ (added), codecs/ilbc/extract-cfile.awk (added),
+ codecs/ilbc/iLBC_encode.h (added), codecs/ilbc/Makefile,
+ codecs/ilbc/FrameClassify.h (added): Include iLBC source code for
+ distribution with Asterisk This patch includes the iLBC source
+ code for distribution with Asterisk. Clarification regarding the
+ iLBC source code was provided by Google, and the appropriate
+ licenses have been included in the codecs/ilbc folder. Review:
+ https://reviewboard.asterisk.org/r/1675 Review:
+ https://reviewboard.asterisk.org/r/1649 (closes issue:
+ ASTERISK-18943) Reporter: Leif Madsen Tested by: Matt Jordan
+ ........ Merged revisions 351450 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-18 15:57 +0000 [r351408] Stefan Schmidt <sst@sil.at>
+
+ * /, channels/chan_sip.c: The get_pai function in chan_sip.c didn't
+ recognized a proper callerid name and number from a
+ P-Asserted-Identity cause the header parsing logic was wrong.
+ Changing the parsing functions to the sip header parsing APIs in
+ reqresp_parser.h solves this problem. Review:
+ https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and
+ Mark Michelson ........ Merged revisions 351396 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-17 17:22 +0000 [r351308] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Eliminate odd initialization of
+ probation variable. ........ Merged revisions 351306 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-17 17:08 +0000 [r351289] Jonathan Rose <jrose@digium.com>
+
+ * res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES: Adds
+ pjmedia probation concepts to res_rtp_asterisk's learning mode.
+ In order to better handle RTP sources with strictrtp enabled
+ (which is now default in 10) using the learning mode to figure
+ out new sources when they change is handled by checking for a
+ number of consecutive (by sequence number) packets received to an
+ rtp struct based on a new configurable value called 'probation'.
+ Also, during learning mode instead of liberally accepting all
+ packets received, we now reject packets until a clear source has
+ been determined. Review: https://reviewboard.asterisk.org/r/1663/
+ ........ Merged revisions 351287 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-17 16:54 +0000 [r351286] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Use built-in parsing functions for
+ Contact and Record-Route headers. If a Contact or a Record-Route
+ header had a quoted string with an item in angle brackets, then
+ we would mis-parse it. For instance, "Bob <1234>"
+ <1234@example.org> would be misparsed as having the URI "1234"
+ The fix for this is to use parsing functions from
+ reqresp_parser.h since they are heavily tested and are awesome.
+ (issue ASTERISK-18990) ........ Merged revisions 351284 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-17 16:07 +0000 [r351234] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Fix udptl issue with initial INVITE
+ introduced by r351027 When an inital INVITE occurs that contains
+ image media, a channel is not yet associated with the SIP dialog.
+ The file descriptor associated with the udptl session needs to be
+ set in initialize_udptl or in sip_new to account for this
+ scenario. ........ Merged revisions 351233 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-17 01:43 +0000 [r351183] Russell Bryant <russell@russellbryant.com>
+
+ * /, channels/chan_sip.c: Merged revisions 351182 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012)
+ | 22 lines Add some missing locking in chan_sip. This patch adds
+ some missing locking to the function
+ send_provisional_keepalive_full(). This function is called from
+ the scheduler, which is processed in the SIP monitor thread. The
+ associated channel (or pbx) thread will also be using the same
+ sip_pvt and ast_channel so locking must be used. The
+ sip_pvt_lock_full() function is used to ensure proper locking
+ order in a safe manner. In passing, document a suspected
+ reference counting error in this function. The "fix" is left
+ commented out because when the "fix" is present, crashes occur.
+ My theory is that fixing it is exposing a reference counting
+ error elsewhere, but I don't know where. (Or my analysis of this
+ being a problem could have been completely wrong in the first
+ place). Leave the comment in the code for so that someone may
+ investigate it again in the future. Also add a bit of doxygen to
+ transmit_provisional_response(). (closes issue ASTERISK-18979)
+ Review: https://reviewboard.asterisk.org/r/1648 ........
+
+2012-01-16 21:17 +0000 [r351081-351131] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Ensure ACK retransmit & hangup on non-200
+ response to INVITE When handling a non-2xx final response on an
+ INVITE transaction, we have to keep the transaction around after
+ we send an ACK in case we receive a retransmission of the
+ response so we can re-transmit the ACK, but also tear down the
+ ast_channel as soon as we transmit the ACK. Before this patch, we
+ could fail at both of these things. Calling
+ sip_alreadygone/needdestroy prevented us from keeping the
+ transaction up and retransmitting the ACK, and queueing
+ CONGESTION was not sufficient to cause the channel to be torn
+ down when originating calls via the CLI, for example. This patch
+ queues a hangup with CONGESTION instead of just queueing
+ CONGESTION for these responses and removes the sip_alreadygone
+ and sip_needdestroy calls from handle_response_invite on non-2xx
+ responses. It relies on the hangup calling sip_scheddestroy. For
+ more information, see section 17.1.1.1 of RFC 3261. (closes issue
+ ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/
+ ........ Merged revisions 351130 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Don't prematurely stop SIP session timer
+ When Asterisk is the UAS (incoming call, endpoint is re-inviting)
+ the SIP session timer expires after half the time the sip
+ endpoint indicates in the Session-expires header in
+ proc_session_timer(). The session timer was being stopped totally
+ and being handled as an error case instead of running again until
+ the second expiry. This patch treats the half-time expiry as a
+ non-error case and continues the timer until the true expiry.
+ (closes issue ASTERISK-18996) Reported by: Thomas Arimont Tested
+ by: Thomas Arimont Patches: session_timer_fix.diff by Terry
+ Wilson (License #5357) based on session_timer.patch by Thomas
+ Arimont (License #5525) ........ Merged revisions 351080 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-16 19:12 +0000 [r351028] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Create and initialize udptl only when
+ dialog negotiates for image media Prior to this patch, the udptl
+ struct was allocated and initialized when a dialog was associated
+ with a peer that supported T.38, when a new SIP channel was
+ allocated, or what an INVITE request was received. This resulted
+ in any dialog associated with a peer that supported T.38 having
+ udptl support assigned to it, including the UDP ports needed for
+ communication. This occurred even in non-INVITE dialogs that
+ would never send image media. This patch creates and initializes
+ the udptl structure only when the SDP for a dialog specifies that
+ image media is supported, or when Asterisk indicates through the
+ appropriate control frame that a dialog is to support T.38.
+ (closes issue ASTERISK-16698) Reported by: under Tested by:
+ Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan
+ (License #6283) (closes issue ASTERISK-16794) Reported by: Elazar
+ Broad Tested by: Stefan Schmidt review:
+ https://reviewboard.asterisk.org/r/1668/ ........ Merged
+ revisions 351027 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-16 17:11 +0000 [r350978] Sean Bright <sean@malleable.com>
+
+ * main/db.c: Sort the output of 'database showkey' as well. You can
+ pass wildcards (%) to the database CLI commands, so this will
+ sort the returned list of matches.
+
+2012-01-16 17:06 +0000 [r350976] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp_engine.c, /: Add missing code to set direct RTP setup
+ information during dialing. ........ Merged revisions 350975 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-16 14:27 +0000 [r350938] Sean Bright <sean@malleable.com>
+
+ * main/db.c: Sort the output of 'database show' by key. This more
+ closely mimics the behavior of 'database show' before the
+ conversion to sqlite3.
+
+2012-01-15 20:12 +0000 [r350886-350889] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, main/asterisk.c: Allow only one thread at a time to do
+ asterisk cleanup/shutdown. Add locking around the
+ really-really-quit part of the core stop/restart part. Previously
+ more than one thread could be called to do cleanup, causing
+ atexit handlers to be run multiple times, in turn causing
+ segfaults. (issue ASTERISK-18883) Reviewed by: Terry Wilson
+ Review: https://reviewboard.asterisk.org/r/1662/ Review:
+ https://reviewboard.asterisk.org/r/1658/ ........ Merged
+ revisions 350888 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, utils/extconf.c: Fix -Werror=unused-but-set-variable compile
+ error in utils/extconf.c. Note that I'm not confirming legitimacy
+ of having that file in tree at all. Is anyone using
+ aelparse/conf2ael? (issue ASTERISK-15350) ........ Merged
+ revisions 350885 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-14 16:41 +0000 [r350790-350838] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac,
+ autoconf/libcurl.m4: Ensure that all AC_LANG_PROGRAM calls in the
+ configure script are properly quoted. Recent versions of autoconf
+ (2.68 on my system) won't properly process the configure script
+ unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in
+ the script were, but many were not. This patch corrects the
+ unquoted calls. ........ Merged revisions 350837 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * contrib/scripts/install_prereq, /, channels/chan_h323.c,
+ addons/chan_mobile.c, res/res_pktccops.c: Multiple revisions
+ 350788-350789 ........ r350788 | kpfleming | 2012-01-14 09:22:33
+ -0600 (Sat, 14 Jan 2012) | 8 lines Ensure that two prerequisites
+ are properly installed on Debian-style distributions. * Don't
+ specify a specific version of libgmime; newer versions are
+ available now and acceptable. * Install libsrtp so that res_srtp
+ can be built. ........ r350789 | kpfleming | 2012-01-14 09:23:32
+ -0600 (Sat, 14 Jan 2012) | 3 lines Correct some
+ 'set-but-not-used' variable warnings. ........ Merged revisions
+ 350788-350789 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-13 22:10 +0000 [r350737] Kinsey Moore <kmoore@digium.com>
+
+ * /, include/asterisk/autoconfig.h.in: Run bootstrap.sh for the for
+ the ASTERISK-18929 fix configure and autoconfig.h.in were not
+ regenerated when the fix was committed. ........ Merged revisions
+ 350736 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-13 21:51 +0000 [r350734] Richard Mudgett <rmudgett@digium.com>
+
+ * /, configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample:
+ Correct eventtype names in cel_odbc and cel_pgsql sample files
+ ........ Merged revisions 350733 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-13 21:41 +0000 [r350731] Kinsey Moore <kmoore@digium.com>
+
+ * /, configure.ac, bootstrap.sh, main/asterisk.c: Make sure
+ asterisk builds on OpenBSD OpenBSD defines SO_PEERCRED, but it
+ returns a 'struct sockpeercred', not 'struct ucred', which causes
+ compilation of main/asterisk.c to fail in read_credentials().
+ This allows configure to check for sockpeercred and asterisk to
+ deal with it properly. (closes issue ASTERISK-18929) Reported-by:
+ Barry Miller Patch-by: Barry Miller ........ Merged revisions
+ 350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-13 20:31 +0000 [r350680] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/sip/config_parser.c: Set port to a default sane value
+ if a bogus one is provided when parsing hostnames. ........
+ Merged revisions 350679 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-13 17:29 +0000 [r350585] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c,
+ configs/cel.conf.sample, /, cel/cel_manager.c,
+ configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample,
+ main/cel.c, configs/cel_custom.conf.sample: Add missing CEL
+ logging fields to various CEL backends. Multiple revisions
+ 350555,350571 ........ r350555 | rmudgett | 2012-01-13 11:12:51
+ -0600 (Fri, 13 Jan 2012) | 12 lines Add missing CEL logging
+ fields to various CEL backends. * Add missing eventextra to
+ cel_psql.c and cel_odbc.c. * Add missing PeerAccount and
+ EventExtra to cel_manager.c. * Add missing userdeftype support
+ for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample.
+ (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman
+ ........ r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13
+ Jan 2012) | 8 lines Use compatible names for event extra data for
+ various CEL backends. * Change eventextra to extra in cel_psql.c
+ and cel_odbc.c. * Change EventExtra to Extra in cel_manager.c.
+ (issue ASTERISK-17190) ........ Merged revisions 350555,350571
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-13 16:59 +0000 [r350550-350553] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_queue.c: Realtime queues failed to load queue
+ information without queue member table Previously, realtime
+ queues could be loaded without defining the queue member table.
+ This allowed for queue members to be dynamic, while the realtime
+ queue definitions could exist in some backing storage. Revision
+ 342223 broke this when it changed the return value for
+ realtime_multientry to return NULL when no results are returned.
+ Previously, an empty ast_config object was expected. (closes
+ issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene
+ Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt
+ Jordan (license 6283) ........ Merged revisions 350552 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * bridges/bridge_builtin_features.c, channels/chan_bridge.c,
+ include/asterisk/bridging.h, apps/app_confbridge.c,
+ main/bridging.c: Fix crash from bridge channel hangup race
+ condition in ConfBridge This patch addresses two issues in
+ ConfBridge and the channel bridge layer: 1. It fixes a race
+ condition wherein the bridge channel could be hung up 2. It
+ removes the deadlock avoidance from the bridging layer and makes
+ the bridge_pvt an ao2 ref counted object Patch by David Vossel
+ (mjordan was merely the commit monkey) (issue ASTERISK-18988)
+ (closes issue ASTERISK-18885) Reported by: Dmitry Melekhov Tested
+ by: Matt Jordan Patches: chan_bridge_cleanup_v.diff uploaded by
+ David Vossel (license 5628) (closes issue ASTERISK-19100)
+ Reported by: Matt Jordan Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1654/
+
+2012-01-12 16:04 +0000 [r350502] Jonathan Rose <jrose@digium.com>
+
+ * /, main/features.c: Adds peer to CEL report on CEL_BRIDGE_START
+ and CEL_BRIDGE_END (closes issue ASTERISK-17940) Reporter: Nic
+ Colledge Patches: features_18.patch uploaded by Nic Colledge
+ (license 6245) ........ Merged revisions 350501 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-11 22:51 +0000 [r350312-350453] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/cel.c: Remove extraneous BRIDGEPEER AMI VarSet event on a
+ CEL dummy channel. (closes issue ASTERISK-19180) Reported by:
+ Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license
+ #5909) patch uploaded by Corey Farrell ........ Merged revisions
+ 350452 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_dial.c, /, CHANGES, apps/app_followme.c: Make FollowMe
+ optionally update connected line information when the accepting
+ endpoint is bridged. Like Dial and Queue, FollowMe needs to deal
+ with AST_CONTROL_CONNECTED_LINE information so when the parties
+ are initially bridged, the connected line information will be
+ correct. * Added the 'I' option just like the app_dial and
+ app_queue 'I' option. * Made 'N' option ignored if the call is
+ already answered. (closes issue ASTERISK-18969) Reported by:
+ rmudgett Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1656/ ........ Merged
+ revisions 350364 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, funcs/func_lock.c: Fix absolute/relative time mismatch in LOCK
+ function. The time passed by the LOCK function to an internal
+ function was relative time when the function expected absolute
+ time. * Don't use C++ keywords in get_lock(). (closes issue
+ ASTERISK-16868) Reported by: Andrey Solovyev Patches:
+ 20101102__issue18207.diff.txt (license #5003) patch uploaded by
+ Andrey Solovyev (modified) ........ Merged revisions 350311 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-09 21:55 +0000 [r350221] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_iax2.c: Fix joinable thread terminating without
+ joiner memory leak in chan_iax.c. The iax2_process_thread() can
+ exit without anyone waiting to join the thread. If noone is
+ waiting to join the thread then a large memory leak occurs. *
+ Made iax2_process_thread() deatach itself if nobody is waiting to
+ join the thread. (closes issue ASTERISK-17339) Reported by:
+ Tzafrir Cohen Patches:
+ asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch
+ (license #5617) patch uploaded by Alex Villacis Lasso (modified)
+ (closes issue ASTERISK-17825) Reported by: wangjin ........
+ Merged revisions 350220 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-09 19:34 +0000 [r350180] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * main/db.c: Fix shutdown handling of sqlite3 astdb. If a db_sync
+ was scheduled just before shutdown, the atexit code calling
+ db_sync would have no effect, causing the astdb commit thread to
+ stay alive. This caused the SIP/realtime_sipregs test to fail.
+ (The fallback kill would run the atexit code again and that would
+ wreak havoc.) This fixes that the atexit kill condition is picked
+ up properly. (closes issue ASTERISK-18883) Reviewed by: Terry
+ Wilson Review: https://reviewboard.asterisk.org/r/1659
+
+2012-01-09 18:57 +0000 [r350076-350129] Richard Mudgett <rmudgett@digium.com>
+
+ * /, contrib/scripts/valgrind_compare (added): Multiple revisions
+ 350127-350128 ........ r350127 | rmudgett | 2012-01-09 12:40:33
+ -0600 (Mon, 09 Jan 2012) | 12 lines Update contrib script
+ live_ast to invoke Asterisk with valgrind and suppression file. *
+ Added valgrind_compare script to compare two valgrind log files
+ for differences. (issue ASTERISK-17339) Reported by: Tzafrir
+ Cohen Patches: valgrind_compare (license #5035) script uploaded
+ by Tzafrir Cohen live_ast_valgrind.diff (license #5035) patch
+ uploaded by Tzafrir Cohen live_ast_valgrind_v2.diff (license
+ #5185) patch uploaded by Paul Belanger ........ r350128 |
+ rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11
+ lines live_ast: valgrind: run asterisk under valgrind Adds a new
+ sub-command, "valgrind" to live_ast. It runs asterisk under
+ valgrind. The extra command-line parameters are passed to
+ Asterisk as usual, and parameters to valgrind are passed through
+ LIVE_AST_VALGRIND_ARGS in live.conf . Review:
+ https://reviewboard.asterisk.org/r/1109/ Merged revisions 326636
+ from http://svn.asterisk.org/svn/asterisk/branches/10 ........
+ Merged revisions 350127-350128 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/asterisk.c: Make Asterisk -x command line parameter imply
+ -r parameter presence. The Asterisk -x command line parameter is
+ documented inconsistently. * Made the -x documentation and
+ behavior consistent. * Since this is also a new year, updated the
+ copyright notices while here. (closes issue ASTERISK-19094)
+ Reported by: Eugene Patches:
+ issueA19094_correct_asterisk_option_x.patch (license #5674) patch
+ uploaded by Walter Doekes (modified) Tested by: Eugene ........
+ Merged revisions 350075 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-09 15:39 +0000 [r350024] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_meetme.c: Prevent SLA settings from getting wiped out
+ on reload If SLA was reloaded without the config file being
+ changed, current settings got wiped out before the SLA reload
+ code decided it wasn't going to reload the file since nothing was
+ changed. Moving the settings reset later in the reload process
+ fixes this. (closes issue AST-744) ........ Merged revisions
+ 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-06 23:25 +0000 [r349977] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Don't leak CID in From header when
+ presentation=unavailable When someone does
+ Set(CALLERPRES()=unavailable) (or
+ Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From
+ header shows "Anonymous" <anonymous@anonymous.invalid>. When
+ sendrpid=yes/pai, the From header will still display the callerid
+ info, even though we supply an rpid header with the anonymous
+ info. It seems like we shouldn't leak that info in any case.
+ Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04
+ seems to indicate that one shouldn't send identifying info in the
+ From in this case. This patch anonymizes the From header as well
+ even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review:
+ https://reviewboard.asterisk.org/r/1649/ ........ Merged
+ revisions 349968 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-06 21:25 +0000 [r349928] Kinsey Moore <kmoore@digium.com>
+
+ * pbx/pbx_lua.c: Fix lua goto detection to prevent unexpected
+ behavior with confbridge A bug in the pbx_lua goto detection was
+ causing the dialplan to hangup unexpectedly after confbridge
+ exited if it had called lua dialplan code during execution.
+ Patch-by: Timo Teras Acked-by: Matt Nicholson (closes issue
+ ASTERISK-18976)
+
+2012-01-06 16:48 +0000 [r349873] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_followme.c: Fix memory leaks in app_followme
+ find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt
+ Jordan ........ Merged revisions 349872 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-05 23:56 +0000 [r349822] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_fax.c: Fix premature free'ing of the frame committed in
+ r349608 Even though we set the frame to the ast_null_frame and
+ return that, the caller of the frame hook may still need the
+ frame. This now is a bit more careful about when it frees the
+ frame, i.e., only under the same conditions that applied when we
+ duplicated it in the first place.
+
+2012-01-05 23:46 +0000 [r349820] Richard Mudgett <rmudgett@digium.com>
+
+ * /, cel/cel_sqlite3_custom.c: Make not assume that the
+ cel_sqlite3_custom SQL table primary key is AcctId. If a table is
+ created by some other application and the primary key is not
+ named "AcctId", cel/cel_sqlite3_custom.c will always try to
+ create the table and fail because it already exists. * Change the
+ SQL table query to not require AcctId as the primary key. (closes
+ issue ASTERISK-18963) Reported by: socketpair Patches: fix.patch
+ (license #6337) patch uploaded by socketpair ........ Merged
+ revisions 349819 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-05 22:10 +0000 [r349732] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/file.c: Allow playback of formats that don't support
+ seeking ast_streamfile previously did unconditional seeking on
+ files that broke playback of formats that don't support that
+ functionality. This patch avoids the seek that was causing the
+ problem. This regression was introduced in r158062. (closes issue
+ ASTERISK-18994) Patch-by: Timo Teras ........ Merged revisions
+ 349731 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-05 21:55 +0000 [r349673-349729] Jonathan Rose <jrose@digium.com>
+
+ * main/dsp.c, /: Fix an issue where dsp.c would interpret multiple
+ dtmf events from a single key press. When receiving calls from a
+ mobile phone into a DISA system on a connection with significant
+ interference, the reporter's Asterisk system would interpret DTMF
+ incorrectly and replicate digits received. This patch resolves
+ that by increasing the number of frames a mismatch has to be
+ detected before assuming the DTMF is over by 1 frame and adjusts
+ dtmf_detect function to reset hits and misses only when an edge
+ is detected. (closes issue ASTERISK-17493) Reported by: Alec
+ Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis
+ (license 5546) Review: https://reviewboard.asterisk.org/r/1130/
+ ........ Merged revisions 349728 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/asterisk.c: Ensures Asterisk closes when receiving
+ terminal signals in 'no fork' mode. When catching a signal, in no
+ fork mode the console thread is identical to the thread
+ responsible for catching the signal and closing Asterisk, which
+ requires it to first dispense with the console thread. Prior to
+ this patch, if these threads were identical, upon receiving a
+ killing signal, the thread will send an URG signal to itself,
+ which we also catch and then promptly do nothing with. Obviously
+ this isn't useful behavior. (closes issue ASTERISK-19127)
+ Reported By: Bryon Clark Patches: quit_on_signals.patch uploaded
+ by Bryon Clark (license 6157) ........ Merged revisions 349672
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-04 22:19 +0000 [r349608-349619] Matthew Jordan <mjordan@digium.com>
+
+ * apps/confbridge/conf_config_parser.c: Fix for ConfBridge config
+ parser unlocking channel mutex too many times When looking up a
+ ConfBridge profile, the config parser would, if it found a
+ channel datastore on the channel requesting the bridge profile,
+ unlock the channel mutex twice. Since that's a little aggressive,
+ it now only unlocks it once. (closes issue ASTERISK-19042)
+ Reported by: Matt Jordan Tested by: Matt Jordan Patches: 19042
+ uploaded by David Vossel (license 5628)
+
+ * res/res_fax.c: Free successfully translated frame in
+ fax_gateway_framehook A frame that is translated via
+ ast_translate is also duplicated via ast_frdup. This will
+ allocate a new frame on the heap, which needs to be free'd at the
+ appropriate time. This issue reporter used valgrind to find that
+ this occurred in res_fax's fax_gateway_framehook; a quick search
+ through the code showed that only place this was currently not
+ handling the translatted frame properly. (closes issue
+ ASTERISK-19133) Reported by: Sylvain Rochet
+
+2012-01-04 20:50 +0000 [r349559] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Fix segfault in chan_dahdi for
+ CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private
+ pointer checks in the following chan_dahdi channel callbacks:
+ dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and
+ dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by:
+ Diego Aguirre Tested by: rmudgett ........ Merged revisions
+ 349558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-04 20:23 +0000 [r349505-349532] Kinsey Moore <kmoore@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Make debian init script
+ conform to the LSB standard Previously, this init script would
+ return 1 if Asterisk was already running. This is incorrect
+ behavior according to the LSB standard and has been fixed by
+ returning 0 instead. (closes issue ASTERISK-17958) Reported-by:
+ johnc ........ Merged revisions 349529 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * contrib/scripts/autosupport, /, contrib/scripts/autosupport.8:
+ Update autosupport script and man page Added information
+ collection from the output of the utilities: top, free, uptime,
+ ifconfig Added information collection from the output of the
+ Asterisk command 'dahdi show status' Added option / flag '-n,
+ --non-interactive' Updated man page to reflect new option / flag
+ '-n, --non-interactive' Patch-by: John Bigelow (itzanger) (closes
+ issue AST-749) ........ Merged revisions 349504 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-04 19:44 +0000 [r349451-349502] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Adds Subscription-State header to notify
+ with call completion. per RFC3265 (Closes issue ASTERISK-17953)
+ Reported by: George Konopacki Patches: 19400.patch uploaded by
+ mmichelson (license 5049) ........ Merged revisions 349482 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/pbx.c, /: Fix documentation for SayNumber to reflect the
+ fact that language is changed in CHANNEL() (closes issue
+ ASTERISK-18962) reported by: Nir Simionovich ........ Merged
+ revisions 349450 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-27 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.1.0 Released.
+
+ * Test results:
+ http://bamboo.asterisk.org/browse/TESTING-ASTERISK1010RCS-4
+
+2012-01-24 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.1.0-rc2 Released.
+
+ * Test results:
+ http://bamboo.asterisk.org/browse/TESTING-ASTERISK1010RCS-3
+
+ * main/file.c: Allow playback of formats that don't support
+ seeking. ast_streamfile previously did unconditional seeking
+ on files that broke playback of formats that don't support that
+ functionality. This patch avoids the seek that was causing the
+ problem. (closes issue ASTERISK-18994) Patch-by: Timo Teras
+
+ * channels/chan_sip.c: AST-2012-001: prevent crash when an SDP offer
+ is received with an encrypted video stream when support for video
+ is disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
+ Reported by: Catalin Sanda
+
+ * channels/chan_sip.c: Fix RTP reference leak. If a blind transfer
+ were initiated using a REFER without a prior reINVITE to place the
+ call on hold, AND if Asterisk were sending RTCP reports, then there
+ was a reference leak for the RTP instance of the transferer.
+ (closes issue ASERISK-19192) Reported by: Tyuta Vitali
+
+ * res/res_rtp_asterisk: Add pjmedia probation concepts to
+ res_rtp_asterisk's learning mode. In order to better handle RTP
+ sources with strictrtp enabled (which is the default setting in 10)
+ using the learning mode to figure out new sources when they change is
+ handled by checking for a number of consecutive (by sequence number)
+ packets received to an rtp struct based on a new configurable value
+ called 'probation'. Also, during learning mode instead of liberally
+ accepting all packets received, we now reject packets until a clear
+ source has been determined.
+
+ * main/features.c: Fix blind transfers from failing if an 'h' extension
+ is present. This prevents the 'h' extension from being run on the
+ transferee channel when it is transferred via a native transfer
+ mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported
+ by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
+ Mark Michelson (license 5049)
+
+ * apps/app_queue.c: Realtime queues failed to load queue
+ information without queue member table. Revision 342223
+ broke this when it changed the return value for
+ realtime_multientry to return NULL when no results are
+ returned. (closes issue ASTERISK-19170) Reported by: Rene
+ Mendoza Tested by: Rene Mendoza
+
+2011-12-30 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.1.0-rc1 Released.
+
+ * Test results:
+ http://bamboo.asterisk.org/browse/TESTING-ASTERISK1010RCS-1
+
+2011-12-29 15:14 +0000 [r349340] Matthew Jordan <mjordan@digium.com>
+
+
+ * main/rtp_engine.c, /: Handle AST_CONTROL_UPDATE_RTP_PEER frames
+ in local bridge loop Failing to handle
+ AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
+ causes the loop to exit prematurely. This causes a variety of
+ negative side effects, depending on when the loop exits. This
+ patch handles the frame by essentially swallowing the frame in
+ the local loop, as the current channel drivers expect the RTP
+ bridge to handle the frame, and, in the case of the local bridge
+ loop, no additional action is necessary. (issue ASTERISK-19040)
+ (issue ASTERISK-19128) (issue ASTERISK-17725) (issue
+ ASTERISK-18340) (closes issue ASTERISK-19095) Reported by: Stefan
+ Schmidt Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1640/ ........ Merged
+ revisions 349339 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-28 21:33 +0000 [r349290] Sean Bright <sean@malleable.com>
+
+ * /, main/audiohook.c: Use ast_audiohook_write_list_empty to
+ determine if our lists are empty instead of duplicating that
+ logic. ........ Merged revisions 349289 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-28 19:00 +0000 [r349248-349250] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils: Tell Subversion to gnore the 'astdb2bdb' binary file if it
+ exists.
+
+ * main/dsp.c, res/res_fax.c, include/asterisk/dsp.h,
+ include/asterisk/res_fax.h, res/res_fax_spandsp.c: Improve T.38
+ gateway V.21 preamble detection. This commit removes the V.21
+ preamble detection code previously added to the generic DSP
+ implementation in Asterisk, and instead enhances the res_fax
+ module to be able to utilize V.21 preamble detection
+ functionality made available by FAX technology modules. This
+ commit also adds such support to res_fax_spandsp, which uses the
+ Spandsp modem tone detection code to do the V.21 preamble
+ detection. There should be no functional change here, other than
+ much more reliable V.21 preamble detection (and thus T.38 gateway
+ initiation).
+
+2011-12-27 20:53 +0000 [r349195] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_timing_pthread.c, include/asterisk/module.h,
+ res/res_timing_dahdi.c, res/res_timing_timerfd.c,
+ res/res_musiconhold.c: Fix timing source dependency issues with
+ MOH Prior to this patch, res_musiconhold existed at the same
+ module priority level as the timing sources that it depends on.
+ This would cause a problem when music on hold was reloaded, as
+ the timing source could be changed after res_musiconhold was
+ processed. This patch adds a new module priority level,
+ AST_MODPRI_TIMING, that the various timing modules are now loaded
+ at. This now occurs before loading other resource modules, such
+ that the timing source is guaranteed to be set prior to resolving
+ the timing source dependencies. (closes issue ASTERISK-17474)
+ Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf,
+ Wes Van Tlghem, elguero, Thomas Arimont Patches:
+ asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff
+ uploaded by elguero (License #5026)
+ asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff
+ uploaded by elguero (License #5026)
+ asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by
+ elguero (License #5026) Review:
+ https://reviewboard.asterisk.org/r/1578/ ........ Merged
+ revisions 349194 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-27 17:17 +0000 [r349145] Sean Bright <sean@malleable.com>
+
+ * /, main/audiohook.c: Once an audiohook is attached to a channel,
+ we continue to transcode all of the frames, even after all of the
+ hooks are detached. This patch short-cicuits us out before we
+ transcode unnecessarily. ........ Merged revisions 349144 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-23 17:32 +0000 [r349045] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 349044 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r349044 | seanbright | 2011-12-23 12:25:01 -0500 (Fri, 23 Dec
+ 2011) | 18 lines In ChanSpy, don't create audiohooks that will
+ never be used. When ChanSpy is initialized it creates and
+ attaches 3 audiohooks: 1) Read audio off of the channel that we
+ are spying on 2) Write audio to the channel that we are spying on
+ 3) Write audio to the channel that is bridged to the channel that
+ we are spying on. The first is always necessary, but the others
+ are used only when specific options are passed to the ChanSpy
+ application (B, d, w, and W to be specific). When those flags are
+ not passed, neither of those audiohooks are ever sent frames, but
+ we still try to process the hooks for each voice frame that we
+ recieve on the channel. So in short - only create and attach
+ audiohooks that we actually need. ........
+
+2011-12-23 15:25 +0000 [r348993] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_dial.c, /: Fix missing doc tags found while fixing
+ ASTERISK-18689 Add missing <variable></variable> tags in app_dial
+ documentation. ........ Merged revisions 348992 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-23 02:30 +0000 [r348952] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /, channels/chan_sip.c, include/asterisk/pbx.h: Fix
+ extension state callback references in chan_sip. Chan_sip gives a
+ dialog reference to the extension state callback and assumes that
+ when ast_extension_state_del() returns, the callback cannot
+ happen anymore. Chan_sip then reduces the dialog reference count
+ associated with the callback. Recent changes (ASTERISK-17760)
+ have resulted in the potential for the callback to happen after
+ ast_extension_state_del() has returned. For chan_sip, this could
+ be very bad because the dialog pointer could have already been
+ destroyed. * Added ast_extension_state_add_destroy() so chan_sip
+ can account for the sip_pvt reference given to the extension
+ state callback when the extension state callback is deleted. *
+ Fix pbx.c awkward statecbs handling in
+ ast_extension_state_add_destroy() and handle_statechange() now
+ that the struct ast_state_cb has a destructor to call. * Ensure
+ that ast_extension_state_add_destroy() will never return -1 or 0
+ for a successful registration. * Fixed pbx.c statecbs_cmp() to
+ compare the correct information. The passed in value to compare
+ is a change_cb function pointer not an object pointer. * Make
+ pbx.c ast_merge_contexts_and_delete() not perform callbacks with
+ AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for
+ deadlocking when those locks are held during the callback. *
+ Removed unused lock declaration for the pbx.c store_hints list.
+ (closes issue ASTERISK-18844) Reported by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1635/ ........ Merged
+ revisions 348940 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-22 22:37 +0000 [r348846-348889] Matthew Jordan <mjordan@digium.com>
+
+ * cel/cel_pgsql.c, /: Fix for memory leaks / cleanup in cel_pgsql
+ There were a number of issues in cel_pgsql's pgsql_log method: *
+ If either sql or sql2 could not be allocated, the method would
+ return while the pgsql_lock was still locked * If the execution
+ of the log statement succeeded, the sql and sql2 structs were
+ never free'd * Reconnection successes were logged as ERRORs. In
+ general, the severity of several logging statements was reduced
+ (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested
+ by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/
+ ........ Merged revisions 348888 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/say.c, main/file.c, main/app.c, apps/app_confbridge.c,
+ main/bridging.c: Add Asterisk TestSuite event hooks to support
+ ConfBridge testing This patch adds initial testsuite event hooks
+ so that ConfBridge tests can be executed in the Asterisk
+ TestSuite. (issue ASTERISK-19059)
+
+2011-12-22 20:17 +0000 [r348845] Terry Wilson <twilson@digium.com>
+
+ * /, include/asterisk/format_pref.h: Allow packetization vaules >
+ 127 According to the RTP packetization documentation, and the
+ maximum values listed in AST_FORMAT_LIST, we should support
+ values > that the signed char array that ast_codec_pref makes
+ available to store the value. All places in the code treat the
+ framing field as though it were an int array instaead of a char
+ array anyway, so this just fixes the type of the array. (closes
+ issue ASTERISK-18876) Review:
+ https://reviewboard.asterisk.org/r/1639/ ........ Merged
+ revisions 348833 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-21 20:13 +0000 [r348736-348793] Richard Mudgett <rmudgett@digium.com>
+
+ * codecs/speex: Make codecs/speex ignore *.i files also.
+
+ * apps/confbridge: Make apps/confbridge ignore *.i files also.
+
+ * /, channels/chan_iax2.c: Fix chan_iax2 to not report an RDNIS
+ number if it is blank. Some ISDN switches complain or block the
+ call if the RDNIS number is empty. * Made chan_iax2 not save a
+ RDNIS number into the ast_channel if the string is blank. This is
+ what other channel drivers do. (closes issue ASTERISK-17152)
+ Reported by: rmudgett ........ Merged revisions 348735 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-19 21:37 +0000 [r348648] Richard Mudgett <rmudgett@digium.com>
+
+ * /, configure, configure.ac: Fix crashes on other platforms caused
+ by interference from Darwin weak symbol support. Support weak
+ symbols on a platform specific basis. The Mac OS X (Darwin)
+ support must be isolated from the other platforms because it has
+ caused other platforms to crash. Several other platforms
+ including Linux have GCC versions that define the weak attribute.
+ However, this attribute is only setup for use in the code by
+ Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang
+ Review: https://reviewboard.asterisk.org/r/1617/ ........ Merged
+ revisions 348647 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-19 19:48 +0000 [r348605] Leif Madsen <lmadsen@digium.com>
+
+ * main/message.c: Update documentation for MESSAGE_SEND_STATUS
+ variable. (Closes issue ASTERISK-19056) Reported by: Yuri
+ Patches: 348360.diff uploaded by Yuri (license #5242)
+
+2011-12-18 18:28 +0000 [r348517] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configs/sip.conf.sample: Correct two flaws in sip.conf.sample
+ related to AST-2011-013. * The sample file listed *two* values
+ for the 'nat' option as being the default. Only 'force_rport' is
+ the default. * The warning about having differing 'nat' settings
+ confusingly referred to both peers and users. ........ Merged
+ revisions 348515 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 348516 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-16 23:56 +0000 [r348311-348465] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /, main/features.c: Clean-up on isle five for
+ __ast_request_and_dial() and ast_call_forward(). * Add locking
+ when a channel inherits variables and datastores in
+ __ast_request_and_dial() and ast_call_forward(). Note: The
+ involved channels are not active so there was minimal potential
+ for problems. * Remove calls to ast_set_callerid() in
+ __ast_request_and_dial() and ast_call_forward() because the set
+ information is for the wrong direction. * Don't use C++ keywords
+ for variable names in ast_call_forward(). * Run the redirecting
+ interception macro if defined when forwarding a call in
+ ast_call_forward(). Note: Currently will never execute because
+ the only callers that supply a calling channel supply a hungup or
+ zombie channel. * Make feature_request_and_dial() put the
+ transferee into autoservice when it calls ast_call_forward() in
+ case a redirection interception macro is run. Note: Currently
+ will never happen because the caller channel (Party B) is always
+ hungup at this time. * Make feature_request_and_dial() ignore the
+ AST_CONTROL_PROCEEDING frame to silence a log message. ........
+ Merged revisions 348464 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/channel.c, /: Fix cut and past error in ast_call_forward().
+ (issue ASTERISK-18836) ........ Merged revisions 348401 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/channel.c, main/pbx.c, /, apps/app_authenticate.c,
+ funcs/func_cdr.c, main/features.c, include/asterisk/cdr.h,
+ apps/app_followme.c, apps/app_queue.c, res/res_monitor.c: Fix
+ crash during CDR update. The ast_cdr_setcid() and
+ ast_cdr_update() were shown in ASTERISK-18836 to be called by
+ different threads for the same channel. The channel driver thread
+ and the PBX thread running dialplan. * Add lock protection around
+ CDR API calls that access an ast_channel pointer. (closes issue
+ ASTERISK-18836) Reported by: gpluser Review:
+ https://reviewboard.asterisk.org/r/1628/ ........ Merged
+ revisions 348362 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_parkandannounce.c, /: Fix ParkAndAnnounce to pass the
+ CallerID to the announcing channel. ParkAndAnnounce tried to pass
+ the CallerID to the announcing channel but the ID was wiped out
+ by the channel masquerade done when parking the call. * Save the
+ CallerID before parking the channel to pass it to the announcing
+ channel. * Fixed a minor memory leak in ParkAndAnnounce. *
+ Updated some ParkAndAnnounce log messages. ........ Merged
+ revisions 348310 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-14 22:34 +0000 [r348265] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_originate.c: Added support for all slin formats to
+ app_originate Previously, app_originate could not originate a
+ call into a non-8kHz conference bridge as the formats for
+ non-8kHz slin codecs were not applied to the created channel.
+ This patch adds all of the formats by default, such that if a
+ created channel has a codec that supports a higher sampling rate,
+ a translation path can be built between it and other channels.
+
+2011-12-14 22:05 +0000 [r348213] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Don't clear LOCALSTATIONID before sending or
+ receiving. The user may set that variable. ASTERISK-18921
+ ........ Merged revisions 348212 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-14 21:58 +0000 [r348211] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_queue.c: Fixed Asterisk crash when function QUEUE_MEMBER
+ receives invalid input The function QUEUE_MEMBER has two required
+ parameters (queuename, option). It was only checking for the
+ presence of queuename. The patch checks for the existence of the
+ option parameter and provides better error logging when invalid
+ values are provided for the option parameter as well.
+
+2011-12-14 20:35 +0000 [r348155-348158] Jonathan Rose <jrose@digium.com>
+
+ * /, configs/features.conf.sample: Fix accidental use of tabs
+ instead of spaces from previous features.conf.sample change
+ ........ Merged revisions 348157 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, configs/features.conf.sample: Document PARKINGSLOT variable in
+ features.conf.sample (issue ASTERISK-16239) ........ Merged
+ revisions 348154 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-13 23:06 +0000 [r348102] Richard Mudgett <rmudgett@digium.com>
+
+ * /, bridges/bridge_builtin_features.c, apps/app_followme.c: Fix
+ FollowMe CallerID on outgoing calls. The addition of the
+ Connected Line support changed how CallerID is passed to outgoing
+ calls. The FollowMe application was not updated to pass CallerID
+ to the outgoing calls. * Fix FollowMe CallerID on outgoing calls.
+ * Restructured findmeexec() to fix several memory leaks and
+ eliminate some duplicated code. * Made check the return value of
+ create_followme_number(). Putting a NULL into the numbers list is
+ bad if create_followme_number() fails. * Fixed a couple uses of
+ ast_strdupa() inside loops. * The changes to
+ bridge_builtin_features.c fix a similar CallerID issue with the
+ bridging API attended and blind transfers. (Not used at this
+ time.) (closes issue ASTERISK-17557) Reported by: hamlet505a
+ Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1612/ ........ Merged
+ revisions 348101 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-13 15:20 +0000 [r348056] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: Fix possible misshandling of an incoming SIP
+ response as a peer poke response. Also make sure peer has even
+ qualify enabled when handle a peer poke response. (closes issue
+ ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and
+ UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed
+ by: David Vossel ........ Merged revisions 348048 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-12 19:24 +0000 [r347996] Terry Wilson <twilson@digium.com>
+
+ * res/res_srtp.c, /: Add a separate buffer for SRTCP packets The
+ function ast_srtp_protect used a common buffer for both SRTP and
+ SRTCP packets. Since this function can be called from multiple
+ threads for the same SRTP session (scheduler for SRTCP and
+ channel for SRTP) it was possible for the packets to become
+ corrupted as the buffer was used by both threads simultaneously.
+ This patch adds a separate buffer for SRTCP packets to avoid the
+ problem. (closes issue ASTERISK-18889, Reported/patch by Daniel
+ Collins) ........ Merged revisions 347995 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-12 18:13 +0000 [r347953-347955] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/extensions.conf.sample, configs/iax.conf.sample,
+ configs/chan_dahdi.conf.sample, configs/chan_ooh323.conf.sample,
+ configs/vpb.conf.sample, configs/extensions.lua.sample,
+ configs/sip.conf.sample: Reverting -r347953 for ASTERISK-14122
+
+ * configs/extensions.conf.sample, configs/iax.conf.sample,
+ configs/chan_dahdi.conf.sample, configs/chan_ooh323.conf.sample,
+ configs/vpb.conf.sample, configs/extensions.lua.sample,
+ configs/sip.conf.sample: Update sample configs to put incoming
+ calls into context public. * Add warning about the SIP allowguest
+ option in context public. (closes issue ASTERISK-14122) Reported
+ by: Alec Davis Review: https://reviewboard.asterisk.org/r/719/
+
+2011-12-09 01:29 +0000 [r347812] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /: Fix some parsing issues in
+ add_exten_to_pattern_tree(). * Simplify compare_char() and avoid
+ potential sign extension issue. * Fix infinite loop in
+ add_exten_to_pattern_tree() handling of character set escape
+ handling. * Added buffer overflow checks in
+ add_exten_to_pattern_tree() character set collection. * Made
+ ignore empty character sets. * Added escape character handling to
+ end-of-range character in character sets. This has a slight
+ change in behavior if the end-of-range character is an escape
+ character. You must now escape it. * Fix potential sign extension
+ issue when expanding character set ranges. * Made remove
+ duplicated characters from character sets. The duplicate
+ characters lower extension matching priority and prevent
+ duplicate extension detection. * Fix escape character handling
+ when the escape character is trying to escape the end-of-string.
+ We could have continued processing characters after the end of
+ the exten string. We could have added the previous character to
+ the pattern matching tree incorrectly. (closes issue
+ ASTERISK-18909) Reported by: Luke-Jr ........ Merged revisions
+ 347811 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-08 21:31 +0000 [r347727] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: Fix regression when using tcpenable=no
+ and tlsenable=yes. The tlsenable settings are tucked away in
+ main/tcptls.c, so I missed them when resolving ASTERISK-18837.
+ This should resolve the test suite breakage of the sip tls tests.
+ Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt
+ Jordan ........ Merged revisions 347718 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-08 20:43 +0000 [r347656] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_queue.c: Fix regressed behavior of queue set penalty to
+ work without specifying 'in <queuename>' r325483 caused a
+ regression in Asterisk 10+ that would make Asterisk segfault when
+ attempting to set penalty on an interface without specifying a
+ queue in the queue set penalty CLI command. In addition, no
+ attempt would be made whatsoever to perform the penalty setting
+ on all the queues in the core list with either the cli command or
+ the non-segfaulting ami equivalent. This patch fixes that and
+ also makes an attempt to document and rename some functions
+ required by this command to better represent what they actually
+ do. Oh yeah, and the use of this command without specifying a
+ specific queue actually works now. Review:
+ https://reviewboard.asterisk.org/r/1609/
+
+2011-12-08 17:53 +0000 [r347600] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Mark channel running the h exten with the
+ soft-hangup flag. When a bridge is broken, ast_bridge_call()
+ might execute the h exten on the calling channel. However, that
+ channel may not have been the channel that broke the bridge by
+ hanging up. The channel executing the h exten must be in a hung
+ up state so things like AGI run in the correct mode. * Make sure
+ ast_bridge_call() marks the channel it is executing the h exten
+ on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as
+ to match the pbx.c main dialplan execution loop when it executes
+ the h exten.) (closes issue ASTERISK-18811) Reported by: David
+ Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621)
+ patch uploaded by rmudgett Tested by: David Hajek, rmudgett
+ ........ Merged revisions 347595 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-08 16:20 +0000 [r347532] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Don't crash on INFO automon request with
+ no channel AST-2011-014. When automon was enabled in
+ features.conf, it was possible to crash Asterisk by sending an
+ INFO request if no channel had been created yet. (closes issue
+ ASTERISK-18805) ........ Merged revisions 347530 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 347531 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-07 21:39 +0000 [r347439] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, /: Update AMI Getvar and Setvar documentation
+ about supplying a channel name. (closes issue ASTERISK-18958)
+ Reported by: Red Patches: jira_asterisk_18958_v1.8.patch (license
+ #5621) patch uploaded by rmudgett ........ Merged revisions
+ 347438 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-07 20:27 +0000 [r347383] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_meetme.c: Fix: Meetme recording variables from
+ realtime DB use null entries over channel variables Meetme would
+ attempt to substitute the realtime values of RECORDING_FILE and
+ RECORDING_FORMAT from the meetme db entry instead of using the
+ channel variable set for those variables in spite of those
+ database entries being NULL or even lacking a column to represent
+ them. (closes issue ASTERISK-18873) Reported by: Byron Clark
+ Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license
+ 6157) ........ Merged revisions 347369 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-07 20:12 +0000 [r347344] Terry Wilson <twilson@digium.com>
+
+ * Makefile, include/asterisk/paths.h, configs/asterisk.conf.sample,
+ build_tools/make_defaults_h, main/asterisk.c, main/db.c: Add
+ ASTSBINDIR to the list of configurable paths This patch also
+ makes astdb2sqlite3 and astcanary use the configured directory
+ instead of relying on $PATH. (closes issue ASTERISK-18959)
+ Review: https://reviewboard.asterisk.org/r/1613/
+
+2011-12-06 23:56 +0000 [r347293] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Make SIP INFO messages for dtmf-relay
+ signals case insensitive. (closes issue ASTERISK-18924) Reported
+ by: Kevin Taylor ........ Merged revisions 347292 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-06 21:53 +0000 [r347240] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c, /: Documents CHANNEL(musicclass) taking priority over
+ m([x]) in waitExten If waitExten specifies a music class to use
+ with its music on hold option, it will use CHANNEL(musicclass)
+ instead if that channel variable has been set on the initiating
+ channel. This documents that behavior in the waitExten app so
+ that this can be known without checking the documentation of the
+ code in function local_ast_moh_start. (closes issue
+ ASTERISK-18804) ........ Merged revisions 347239 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-06 19:42 +0000 [r347124-347167] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: Don't allow transport=tcp when
+ tcpenable=no. When tcpenable=no, sending to transport=tcp hosts
+ was still allowed. Resolving the source address wasn't possible
+ and yielded the string "(null)" in SIP messages. Fixed that and a
+ couple of not-so-correct log messages. (closes issue
+ ASTERISK-18837) Reported by: Andreas Topp Review:
+ https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan
+ ........ Merged revisions 347166 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_voicemail.c, /: Add regression tests for issue
+ ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572
+ Reviewed by: Matt Jordan ........ Merged revisions 347131 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_voicemail.c, /: Move setting of voicemail zonetag and
+ locale up a bit. The voicemail [general] zonetag and locale
+ variables weren't loaded until after the mailboxes were
+ initialized. This caused the settings to be unset for those
+ mailboxes until a reload was performed. (closes issue
+ ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570
+ Reviewed by: Matt Jordan ........ Merged revisions 347111 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-06 17:24 +0000 [r347068] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Fixed crash from orphaned MWI
+ subscriptions in chan_sip This patch resolves the issue where MWI
+ subscriptions are orphaned by subsequent SIP SUBSCRIBE messages.
+ When a peer is removed, either by pruning realtime SIP peers or
+ by unloading / loading chan_sip, the MWI subscriptions that were
+ orphaned would still be on the event engine list of valid
+ subscriptions but have a pointer to a peer that no longer was
+ valid. When an MWI event would occur, this would cause a seg
+ fault. (closes issue ASTERISK-18663) Reported by: Ross Beer
+ Tested by: Ross Beer, Matt Jordan Patches:
+ blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
+ Review: https://reviewboard.asterisk.org/r/1610/ ........ Merged
+ revisions 347058 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-05 17:42 +0000 [r347007] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Restore call progress code for analog
+ ports. Extracting sig_analog from chan_dahdi lost call progress
+ detection functionality. * Fix analog ports from considering a
+ call answered immediately after dialing has completed if the
+ callprogress option is enabled. (closes issue ASTERISK-18841)
+ Reported by: Richard Miller Patches: chan_dahdi.diff (license
+ #5685) patch uploaded by Richard Miller (Modified by me)
+ sig_analog.c.diff (license #5685) patch uploaded by Richard
+ Miller (Modified by me) sig_analog.h.diff (license #5685) patch
+ uploaded by Richard Miller ........ Merged revisions 347006 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-05 15:02 +0000 [r346955] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c, /: Resolve duplicate label used in multiple
+ priorities for the same extension. Prior to this patch, if labels
+ with the same name were used for different priorities in the same
+ extension, the new label would be accepted, but it would be
+ unusable since attempts to reach that label would just go to the
+ first one. Now pbx.c detects this, generates a warning in logs,
+ and culls the label before adding it to the dialplan. (closes
+ issue ASTERISK-18807) Reported by: Kenneth Shumard Patches:
+ pbx.c.patch uploaded by Kenneth Shumard (License 5077) ........
+ Merged revisions 346954 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-05 14:46 +0000 [r346952] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_jabber.exports.in, /: Fix chan_jingle/gtalk load
+ regression introduced in r346087 Add missing symbol exports for
+ ast_aji_client_destroy and ast_aji_buddy_destroy for usage
+ outside res_jabber. Testing of these changes focused on
+ res_jabber itself, so this problem was missed. Reported-by:
+ Michael Spiceland ........ Merged revisions 346951 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-04 10:03 +0000 [r346900] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and
+ domain ACL bypass. The code that allowed admins to create users
+ with domain-only uri's had stopped to work in 1.8 because of the
+ reqresp parser rewrites. This is fixed now: if you have a
+ [mydomain.com] sip user, you can register with useraddr
+ sip:mydomain.com. Note that in that case -- if you're using
+ domain ACLs (a configured domain list) -- mydomain.com must be in
+ the allow list as well. Reviewboard r1606 shows a list of
+ registration combinations and which SIP response codes are
+ returned. Review: https://reviewboard.asterisk.org/r/1533/
+ Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes
+ issue ASTERISK-18741) ........ Merged revisions 346899 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-02 23:27 +0000 [r346856] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Update SIP MESSAGE To parsing to correctly
+ handle URI The previous patch (r346040) incorrectly parsed the
+ URI in the presence of a port, e.g., user@hostname:port would
+ fail as the port would be double appended to the SIP message.
+ This patch uses the parse_uri function to correctly parse the URI
+ into its username and hostname parts, and places them in the
+ correct fields in the sip_pvt structure. (issue ASTERISK-18903)
+ Review: https://reviewboard.asterisk.org/r/1597/
+
+2011-12-02 16:42 +0000 [r346763] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, /, channels/chan_h323.c: Merged revisions
+ 346762 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7
+ lines process null frame pointer returned by
+ ast_rtp_instance_read correctly (closes issue ASTERISK-16697)
+ Reported by: under Patches: segfault.diff (License #5871) patch
+ uploaded by under ........
+
+2011-12-01 21:14 +0000 [r346701] Richard Mudgett <rmudgett@digium.com>
+
+ * main/stun.c, /, res/res_stun_monitor.c,
+ configs/res_stun_monitor.conf.sample, include/asterisk/stun.h:
+ Re-resolve the STUN address if a STUN poll fails for
+ res_stun_monitor. The STUN socket must remain open between polls
+ or the external address seen by the STUN server is likely to
+ change. However, if the STUN request poll fails then the STUN
+ server address needs to be re-resolved and the STUN socket needs
+ to be closed and reopened. * Re-resolve the STUN server address
+ and create a new socket if the STUN request poll fails. * Fix
+ ast_stun_request() return value consistency. * Fix
+ ast_stun_request() to check the received packet for expected
+ message type and transaction ID. * Fix ast_stun_request() to read
+ packets until timeout or an associated response packet is found.
+ The stun_purge_socket() hack is no longer required. * Reduce
+ ast_stun_request() error messages to debug output. * No longer
+ pass in the destination address to ast_stun_request() if the
+ socket is already bound or connected to the destination. (closes
+ issue ASTERISK-18327) Reported by: Wolfram Joost Tested by:
+ rmudgett Review: https://reviewboard.asterisk.org/r/1595/
+ ........ Merged revisions 346700 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-01 20:37 +0000 [r346565-346698] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180
+ ringing. 183 Ringing isn't even a thing. 183 is actually a
+ session progress message. (closes issue ASTERISK-18925) Reported
+ by: Sebastian Denz Tested by: jrose Patches:
+ asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian
+ Denz (License #6139) ........ Merged revisions 346697 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
+ r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) |
+ 18 lines Cleaning up chan_sip/tcptls file descriptor closing.
+ This patch attempts to eliminate various possible instances of
+ undefined behavior caused by invoking close/fclose in situations
+ where fclose may have already been issued on a
+ tcptls_session_instance and/or closing file descriptors that
+ don't have a valid index for fd (-1). Thanks for more than a
+ little help from wdoekes. (closes issue ASTERISK-18700) Reported
+ by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane
+ Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas
+ Review: https://reviewboard.asterisk.org/r/1576/ ........ Merged
+ revisions 346564 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-30 19:37 +0000 [r346473] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/queues.conf.sample: Update queues.conf.sample
+ documentation. Update the documentation surrounding the use of
+ MONITOR_EXEC to make it more clear that it can be used for both
+ Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413)
+ Reported by: David Woolley Patches:
+ issue18817_mixmonitor_queues_doc.diff by Michael L. Young
+ (License #5026) ........ Merged revisions 346472 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-29 00:00 +0000 [r346349] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/message.h, main/message.c: Fixes memory leak in
+ message API. The ast_msg_get_var function did not properly
+ decrement the ref count of the var it retrieves. The way this is
+ implemented is a bit tricky, as we must decrement the var and
+ then return the var's value. As long as the documentation for the
+ function is followed, this will not result in a dangling pointer
+ as the ast_msg structure owns its own reference to the var while
+ it exists in the var container.
+
+2011-11-28 14:32 +0000 [r346293] Stefan Schmidt <sst@sil.at>
+
+ * res/res_rtp_asterisk.c, /: Fix regression that 'rtp/rtcp set
+ debup ip' only works when also a port was specified. (closes
+ issue ASTERISK-18693) Reported by: Davide Dal Fra Review:
+ https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter
+ Doekes ........ Merged revisions 346292 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-23 22:58 +0000 [r346240] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/acl.h, /, channels/chan_skinny.c,
+ channels/chan_h323.c, main/acl.c, channels/chan_iax2.c: Fix calls
+ to ast_get_ip() not initializing the address family. ........
+ Merged revisions 346239 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-23 20:45 +0000 [r346145-346198] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text()
+ function. In r116240, get_msg_text() got an extra parameter to
+ fix the unwanted addition of trailing newlines to SIP MESSAGE
+ bodies. This caused all linefeeds to be trimmed, which isn't
+ right either. This is a stop-gap; the right fix is to return the
+ original SIP request body. Review:
+ https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan
+ ........ Merged revisions 346147 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, include/asterisk/strings.h: Fix ast_str_truncate signedness
+ warning and documentation. Review:
+ https://reviewboard.asterisk.org/r/1594 ........ Merged revisions
+ 346144 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-23 17:14 +0000 [r346087] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_jingle.c, /, include/asterisk/jabber.h,
+ channels/chan_gtalk.c, res/res_jabber.c: Fix res_jabber resource
+ leaks This should fix almost all resource leaks in res_jabber
+ that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous
+ situation where ast_aji_get_client would sometimes bump an
+ object's refcount and sometimes not. Review:
+ https://reviewboard.asterisk.org/r/1553 ........ Merged revisions
+ 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-23 16:19 +0000 [r346040] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Fixed SendMessage stripping extension from
+ To: header in SIP MESSAGE When using the MessageSend application
+ to send a SIP MESSAGE to a non-peer, chan_sip attempted to
+ validate the hostname or IP Address. In the process, it stripped
+ off the extension and failed to add it back to the sip_pvt
+ structure before transmitting. This patch adds the full URI
+ passed in from the message core to the sip_pvt structure. (closes
+ issue ASTERISK-18903) Reported by: Shaun Clark Tested by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/1597/
+
+2011-11-23 16:10 +0000 [r346031] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_musiconhold.c: Resume playing existing hold music for
+ cached realtime MOH As a result of the fix for ASTERISK-18039,
+ realtime caching MOH no longer properly resumes playing back a
+ file between different holds in the same call. This is because
+ scanning for new files causes the existing file array to be
+ emptied and we were just comparing that the saved pointer to the
+ filename matched the pointer to the filename in a particular
+ position in the array. An easy fix is to save the filename
+ instead of a pointer to it and then do a strcmp instead of
+ comparing the addresses. (closes issue ASTERISK-18912) Review:
+ https://reviewboard.asterisk.org/r/1596/ ........ Merged
+ revisions 346030 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-23 16:06 +0000 [r346029] Paul Belanger <pabelanger@digium.com>
+
+ * res/res_format_attr_celt.c, res/res_format_attr_silk.c: Added
+ support level for new modules
+
+2011-11-22 23:00 +0000 [r345977] Richard Mudgett <rmudgett@digium.com>
+
+ * main/dnsmgr.c, /, include/asterisk/dnsmgr.h: Fix dnsmgr entries
+ to ask for the same address family each time. The dnsmgr refresh
+ would always get the first address found regardless of the
+ original address family requested. So if you asked for only IPv4
+ addresses originally, you might get an IPv6 address on refresh. *
+ Saved the original address family requested by
+ ast_dnsmgr_lookup() to be used when the address is refreshed.
+ ........ Merged revisions 345976 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-22 20:31 +0000 [r345924] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * include/asterisk/logger.h, /: Clarify why the AST_LOG_* macros
+ exist next to the LOG_* macros. (issue ASTERISK-17973) ........
+ Merged revisions 345923 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-22 16:40 +0000 [r345882] Paul Belanger <pabelanger@digium.com>
+
+ * apps/confbridge/conf_config_parser.c: Add missing sound_only_one
+ config variable (closes issue ASTERISK-18895) Reported by:
+ zvision Patches: conf_config_parser.diff (license #5755) patch
+ uploaded by zvision
+
+2011-11-21 21:07 +0000 [r345830] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Default
+ to nat=yes; warn when nat in general and peer differ It is
+ possible to enumerate SIP usernames when the general and
+ user/peer nat settings differ in whether to respond to the port a
+ request is sent from or the port listed for responses in the Via
+ header. In 1.4 and 1.6.2, this would mean if one setting was
+ nat=yes or nat=route and the other was either nat=no or
+ nat=never. In 1.8 and 10, this would mean when one was
+ nat=force_rport and the other was nat=no. In order to address
+ this problem, it was decided to switch the default behavior to
+ nat=yes/force_rport as it is the most commonly used option and to
+ strongly discourage setting nat per-peer/user when at all
+ possible. For more discussion of the issue, please see:
+ http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
+ (closes issue ASTERISK-18862) Review:
+ https://reviewboard.asterisk.org/r/1591/ ........ Merged
+ revisions 345776 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged
+ revisions 345800 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 345828 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-19 15:10 +0000 [r345640-345683] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/db.c: Update the documentation to better clarify how the
+ existing commands work. Review:
+ https://reviewboard.asterisk.org/r/1593/ ........ Merged
+ revisions 345682 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/db.c: Fix a change in behavior in 'database show' from 1.8.
+ In 1.8 and previous versions, one could use any fullword portion
+ of the key name, including the full key, to obtain the record.
+ Until this patch, this did not work for the full key. Closes
+ issue ASTERISK-18886 Patch: by tilghman Review: by twilson
+ (http://pastebin.com/7rtu6bpk) on #asterisk-dev
+
+2011-11-17 17:29 +0000 [r345558] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Remove dead code since pri_grab() can
+ never fail. Dead code makes programmers sick. I am sick of
+ looking at it. ........ Merged revisions 345546 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-16 14:49 +0000 [r345488] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_voicemail.c, /: Guarantee messages go into the right
+ folders with multiple recipients Before, using the U flag in
+ Voicemail with multiple recipients would put urgent messages in
+ the INBOX folder for all users past the first thanks to a bug
+ with the message copying function. This would also cause messages
+ to fail to be sent if the INBOX directory hadn't been created for
+ that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt
+ Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/1589/ ........ Merged
+ revisions 345487 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-15 20:10 +0000 [r345220-345432] Richard Mudgett <rmudgett@digium.com>
+
+ * /, res/res_agi.c: Make FastAGI HANGUP show up in AGI debug
+ output. * Change from using send() to ast_agi_send() so the
+ HANGUP shows up in the AGI debug output. (closes issue
+ ASTERISK-18723) Reported by: James Van Vleet Patches:
+ jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by
+ rmudgett ........ Merged revisions 345431 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/sig_pri.c: Fix typo in sig_pri using wrong structure
+ name. It is fortunate that the typo does not alter generated code
+ since the e->restart.channel and e->ring.channel members are in
+ the same position. (closes issue ASTERISK-18868) Reported by:
+ zvision Patches: sig_pri.c.diff (License #5755) patch uploaded by
+ zvision ........ Merged revisions 345370 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_queue.c: Make queue log indicate if ADDMEMBER is
+ paused for AMI and realtime. * Add parameter to queue log
+ ADDMEMBER to indicate if the member is paused. (closes issue
+ ASTERISK-18645) Reported by: garlew Patches: paused.diff (License
+ #5337) patch uploaded by garlew Tested by: rmudgett, garlew
+ Review: https://reviewboard.asterisk.org/r/1469/ ........ Merged
+ revisions 345285 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample, UPGRADE-1.8.txt,
+ channels/sip/include/sip.h: Restore SIP DTMF overlap dialing
+ method. The recent fix for ASTERISK-17288 to get RFC3578 SIP
+ overlap support working correctly removed a long standing ability
+ to do overlap dialing using DTMF in the early media phase of a
+ call. See ASTERISK-18702 it has a very good description of the
+ issue. I started with Pavel Troller's chan_sip.diff patch on
+ issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf
+ allowoverlap config option. The new option value causes the
+ Incomplte application to not send anything with chan_sip so the
+ caller can supply more digits via DTMF. * Renames
+ SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
+ since that is what it really means. * Fixed get_destination()
+ inconsistency with the pickup extension matching. * Fixed
+ initialization of PAGE3 of global_flags in reload_config().
+ (closes issue ASTERISK-18702) Reported by: Pavel Troller Review:
+ https://reviewboard.asterisk.org/r/1517/ Review:
+ https://reviewboard.asterisk.org/r/1582/ ........ Merged
+ revisions 345273 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/pbx.c, /: Fix Progress spelling error in main/pbx.c. (closes
+ issue ASTERISK-18857) Reported by: David M Patches:
+ mainpbx-trivial.patch (License #6326) patch uploaded by David M
+ ........ Merged revisions 345219 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-14 19:10 +0000 [r345164] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Don't read past end of input when calling
+ write() int blah = 1; ... write(chan->alertpipe[1], &blah,
+ new_frames * sizeof(blah)) != (new_frames * sizeof(blah))) is
+ only valid when new_frames == 1. Otherwise we start reading into
+ adjacent variables declared on the stack. The read end discards
+ what is read, so the values don't matter but it's not a good idea
+ to read past where we want even though new_frames is almost
+ always 1 and should never be large. This patch is basically taken
+ out of kpfleming's eventfd branch, as he mentioned that he
+ remembered fixing it there when I talked to him about this issue.
+ Review: https://reviewboard.asterisk.org/r/1583/ ........ Merged
+ revisions 345163 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-14 19:01 +0000 [r345161] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * channels/sip/include/reqresp_parser.h, /: Update reqresp_parser
+ parse_uri doxygen comments. The issue mentioned in the bug report
+ had been fixed recently by twilson. The reporter included this
+ documentation fix. (closes issue ASTERISK-18572) Reported by:
+ Richard Miller Patch by: Richard Miller (modified) ........
+ Merged revisions 345160 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-14 16:12 +0000 [r345117] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_voicemail.c, /: Moves voicemail setup password entry to
+ the end of the setup process. This change was made because
+ forcegreeting and forcename settings in voicemail could be
+ circumvented by hanging up after entering a password, because the
+ only way voicemail currently observes whether a mailbox is new or
+ not is by checking to see if the password is the same as the
+ mailbox number or not. (closes issue ASTERISK-18282) Reported by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/
+ ........ Merged revisions 345062 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-14 15:10 +0000 [r345064] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Ensure that a null vmexten does not cause
+ a segfault When sip_send_mwi_to_peer was modified recently to
+ avoid deadlocks, vmexten was not expected to be null. This change
+ handles that situation to avoid a segfault. ........ Merged
+ revisions 345063 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-12 16:17 +0000 [r344966] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * channels/chan_misdn.c, /: mISDN Round Robin break when no channel
+ is available Prevent channels been parsed repetitively. ........
+ Merged revisions 344965 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-12 00:34 +0000 [r344900] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_musiconhold.c: Don't forget to rescan MOH files for
+ cached realtime classes Realtime MOH class caching was
+ implemented because without it, you would build a completely new
+ MOH class and would start the music over at the beginning each
+ time hold was pressed in a conversation. Unfortunately, this
+ broke re-scanning for file changes for realtime MOH classes. This
+ patch corrects that issue. (closes issue ASTERISK-18039) Review:
+ https://reviewboard.asterisk.org/r/1579/ ........ Merged
+ revisions 344899 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-11 21:58 +0000 [r344845] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * include/asterisk/stringfields.h, include/asterisk/utils.h, /,
+ main/utils.c: Use __alignof__ instead of sizeof for stringfield
+ length storage. Kevin P Fleming suggested that r343157 should use
+ __alignof__ instead of sizeof. For most systems this won't be an
+ issue, but better fix it now while it's still fresh. Review:
+ https://reviewboard.asterisk.org/r/1573 ........ Merged revisions
+ 344843 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-11 21:50 +0000 [r344842] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/file.c: Video format was treated as audio when removed
+ from the file playback scheduler This patch fixes the format type
+ check in ast_closestream and filestream_destructor. Previously a
+ comparison operator was used, but since audio formats are no
+ longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats
+ that have a value greater than the video formats), a bitwise AND
+ operation is used instead. Duplicated code was also moved to
+ filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo
+ Bedrij Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1580/ ........ Merged
+ revisions 344823 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-11 21:36 +0000 [r344836-344839] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/sip/reqresp_parser.c: Remove unneeded if(params)
+ checks in reqresp_parser. Nick Lewis added them in
+ https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent
+ reason. There is no way that params could become NULL in that
+ piece of code, so I removed these excess checks again. ........
+ Merged revisions 344837 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/manager.c, /: Fix bad quoting of multiline mxml opaque_data
+ that caused invalid xml. The opaque_data was added and enclosed
+ in single quotes, assuming it would be only a single line. The
+ rest of the lines were appended after the closing quote. (closes
+ issue ASTERISK-18852) Reported by: peep_ on IRC Review:
+ https://reviewboard.asterisk.org/r/1577 ........ Merged revisions
+ 344835 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-11 20:14 +0000 [r344770] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Fix regression introduced by SDP fixups
+ If capability is adjusted when switching to UDPTL during fax
+ transmission, fax teardown fails. Make sure capability is only
+ touched if RTP is active. This regression was introduced in
+ R344385. ........ Merged revisions 344769 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-11 18:36 +0000 [r344662-344716] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Check sip.conf maxforwards parameter for
+ range 1 <= x <= 255. JIRA AST-710 ........ Merged revisions
+ 344715 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/cli.c, /: Make CLI "core show channel" not hold the channel
+ lock during console output. Holding the channel lock while the
+ CLI "core show channel" command is executing can slow down the
+ system. It could block the system if the console output is halted
+ or paused. * Made capture the CLI "core show channel" output into
+ a buffer to be output after the channel is unlocked. * Removed
+ use of C++ keyword as a variable name. out renamed to obuf. *
+ Checked allocation of obuf for failure so will not crash. (closes
+ issue ASTERISK-18571) Reported by: Pavel Troller Tested by:
+ rmudgett ........ Merged revisions 344661 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-11 15:33 +0000 [r344609] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c, /: Fix a segmentation fault when using an extension
+ with CID matching and no CID. Attempting to call an extension
+ which used Caller ID matching with a channel that has an empty
+ caller id string would result in a segmentation fault. (closes
+ issue ASTERISK-18392 Reported By: Ales Zelenik ........ Merged
+ revisions 344608 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-10 23:21 +0000 [r344537-344557] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_macro.c: Fix app_macro.c MODULEINFO section termination.
+ (closes issue ASTERISK-18848) Reported by: Tony Mountifield
+
+ * /, apps/app_queue.c: Fix potential deadlock calling ast_call()
+ with channel locks held. Fixed app_queue.c:ring_entry() calling
+ ast_call() with the channel locks held. Chan_local attempts to do
+ deadlock avoidance in its ast_call() callback and could deadlock
+ if a channel lock is already held. ........ Merged revisions
+ 344539 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_queue.c: Make AMI event AgentCalled get
+ CallerID/ConnectedLine info from the incoming channel. It was
+ strange that the AgentCalled AMI event would get most of its
+ information from the incoming channel but then get the CallerID
+ information from the outgoing channel. Before connected line
+ support was added, this information was always the same at this
+ point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham
+ Tested by: rmudgett ........ Merged revisions 344536 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-10 21:54 +0000 [r344493] David Vossel <dvossel@digium.com>
+
+ * main/bridging.c: Fixes issue with ConfBridge participants hanging
+ up during DTMF feature menu usage getting stuck in conference
+ forever. When a conference user enters the DTMF menu they are
+ suspended from the bridge while the channel is handed off to the
+ DTMF feature code. If a user entered this state and hungup, there
+ existed a race condition where the channel could not exit the
+ conference because it was waiting on a signal that would never
+ arrive. This patch fixes that, because it would stupid for me to
+ talk about the problem and commit a patch for something else.
+ (closes issue ASTERISK-18829) Reported by: zvision
+
+2011-11-10 21:14 +0000 [r344386-344440] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_meetme.c: Fix another incorrect case with meetme's
+ PIN logic and add documentation This fixes an issue where a user
+ of a dynamic conference was asked for a PIN twice. This also adds
+ documentation to assist in future modifications to the piece of
+ code responsible for PIN checking. (closes issue AST-670)
+ ........ Merged revisions 344439 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Fix several
+ bugs with SDP parsing and well-formedness of responses Fix bug
+ ASTERISK-16558 which dealt with the order of responses to
+ incoming streams defined by SDP. Fix unreported bug where
+ offering multiple same-type streams would cause Asterisk to reply
+ with an incorrect SDP response missing one or more streams
+ without a proper declination. Fix bugs related to a single
+ non-audio stream being offered with responses requesting codecs
+ that were not offered in the initial invite along with an
+ additional audio stream that was not in the initial invite.
+ Review: https://reviewboard.asterisk.org/r/1516/ ........ Merged
+ revisions 344385 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-10 16:25 +0000 [r344334] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_rtp_asterisk.c, /: only attempt to do stun handling on
+ ipv4 or ipv4 mapped to ipv6 addresses Patch by: jkonieczny
+ (modified) ASTERISK-18490 ........ Merged revisions 344330 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-09 20:53 +0000 [r344271] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Fix deadlock during dialplan reload.
+ Another deadlock between the conlock/hints and channels/channel
+ locking orders. * Don't hold the channel and private lock in
+ sip_new() when calling ast_exists_extension(). (closes issue
+ ASTERISK-18740) Reported by: Byron Clark Patches:
+ sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by
+ Gregory Hinton Nietsky ASTERISK-18740.patch (license #6157) patch
+ uploaded by Byron Clark Tested by: Byron Clark ........ Merged
+ revisions 344268 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-09 20:07 +0000 [r344175-344216] Terry Wilson <twilson@digium.com>
+
+ * channels/sip/include/reqresp_parser.h, /, channels/chan_sip.c,
+ channels/sip/reqresp_parser.c, channels/sip/include/sip.h: Don't
+ treat a host:port string as a domain The domain matching code
+ prior to 1.8 used to manually remove the port from the host:port
+ string when determining if an incoming request matched the list
+ of domains. When switching to the new parsing functions, the
+ documentation implied that the "domain" was being returned by
+ these functions, when instead it was returning the "hostport" as
+ defined by RFC 3261. This led to confusion and resulted in 1.8+
+ rejecting an incoming request from x.x.x.x:xxxxx when
+ domain=x.x.x.x was set in sip.conf. This patch renames the
+ "domain" variables in the parsing functions to "hostport" to more
+ accurately describe what it is that they are returning and also
+ properly truncates the resulting hostport strings when dealing
+ with domain matching. Review:
+ https://reviewboard.asterisk.org/r/1574/ ........ Merged
+ revisions 344215 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, tests/test_netsock2.c: Add a unit test for
+ ast_sockaddr_split_hostport Review:
+ https://reviewboard.asterisk.org/r/1575/ ........ Merged
+ revisions 344157 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-09 19:02 +0000 [r344159-344160] Alexandr Anikin <may@telecom-service.ru>
+
+ * /: delete svn:mergeinfo
+
+ * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooh323.c, /,
+ addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooq931.h,
+ addons/ooh323c/src/ootypes.h, addons/ooh323c/src/oochannels.c:
+ Generate response to Status Enquiry message with Status q.931
+ message. Some PBXes require this for call status checking (closes
+ issue ASTERISK-18748) Reported by: Fabrizio Lazzaretti Patches:
+ ASTERISK-18748-5.patch (License #5415) patch uploaded by may213
+ Tested by: Fabrizio Lazzaretti ........ Merged revisions 344158
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-09 17:14 +0000 [r344103] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_meetme.c: Fix pin parameter behavior regression in
+ MeetMe The last time this code was touched (by me), a subtlety
+ was missed based on the difference between needing to check a
+ pin's validity and the need to prompt for a pin. (closes issue
+ ASTERISK-18488) ........ Merged revisions 344102 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-09 15:26 +0000 [r344049] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, formats/format_wav.c: don't call ltohl() twice on the same
+ value ASTERISK-18739 Patch by: pawel (modified) ........ Merged
+ revisions 344048 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-08 21:59 +0000 [r344004] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Residual changes for Asterisk v10 branch
+ from ASTERISK-18747. Residual changes for Asterisk v10 branch
+ from ASTERISK-18747 after
+ https://reviewboard.asterisk.org/r/1564/ commit and associated
+ dialogs callid hash key change fix. * Make check_rtp_timeout()
+ return CMP_MATCH if need to delete dialog from dialogs_rtpcheck.
+ This is an optimization to avoid an unneeded lock/unlock and
+ object search when using ao2_unlink. * Prevent crash in
+ check_rtp_timeout() if dialog->rtp is NULL. Review:
+ https://reviewboard.asterisk.org/r/1557/
+
+2011-12-15 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.0.0 Released.
+
+2011-12-08 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.0.0-rc3 Released.
+
+ * Add ASTSBINDIR to the list of configurable paths
+
+ This patch also makes astdb2sqlite3 and astcanary use the configured
+ directory instead of relying on $PATH.
+
+ (closes issue ASTERISK-18959)
+ Review: https://reviewboard.asterisk.org/r/1613/
+
+ * Don't crash on INFO automon request with no channel
+
+ AST-2011-014. When automon was enabled in features.conf, it was possible
+ to crash Asterisk by sending an INFO request if no channel had been
+ created yet.
+
+ (closes issue ASTERISK-18805)
+
+ * Fixed crash from orphaned MWI subscriptions in chan_sip
+
+ This patch resolves the issue where MWI subscriptions are orphaned
+ by subsequent SIP SUBSCRIBE messages. When a peer is removed, either
+ by pruning realtime SIP peers or by unloading / loading chan_sip, the
+ MWI subscriptions that were orphaned would still be on the event engine
+ list of valid subscriptions but have a pointer to a peer that no longer
+ was valid. When an MWI event would occur, this would cause a seg fault.
+
+ (closes issue ASTERISK-18663)
+ Review: https://reviewboard.asterisk.org/r/1610/
+
+ * Fix a change in behavior in 'database show' from 1.8.
+
+ In 1.8 and previous versions, one could use any fullword portion of
+ the key name, including the full key, to obtain the record. Until this
+ patch, this did not work for the full key.
+
+ (closes issue ASTERISK-18886)
+
+ * Default to nat=yes; warn when nat in general and peer differ
+
+ AST-2011-013. It is possible to enumerate SIP usernames when the general and
+ user/peer nat settings differ in whether to respond to the port a request is
+ sent from or the port listed for responses in the Via header. In 1.4 and
+ 1.6.2, this would mean if one setting was nat=yes or nat=route and the other
+ was either nat=no or nat=never. In 1.8 and 10, this would mean when one
+ was nat=force_rport and the other was nat=no.
+
+ In order to address this problem, it was decided to switch the default
+ behavior to nat=yes/force_rport as it is the most commonly used option
+ and to strongly discourage setting nat per-peer/user when at all
+ possible.
+
+ For more discussion of the issue, please see:
+ http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
+
+ (closes issue ASTERISK-18862)
+ Review: https://reviewboard.asterisk.org/r/1591/
+
+ * Fixed SendMessage stripping extension from To: header in SIP MESSAGE
+
+ When using the MessageSend application to send a SIP MESSAGE to a
+ non-peer, chan_sip attempted to validate the hostname or IP Address. In the
+ process, it stripped off the extension and failed to add it back to the sip_pvt
+ structure before transmitting. This patch adds the full URI passed in
+ from the message core to the sip_pvt structure.
+
+ (closes issue ASTERISK-18903)
+ Reported by: Shaun Clark
+ Tested by: Matt Jordan
+
+ Review: https://reviewboard.asterisk.org/r/1597/
+
+2011-11-15 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.0.0-rc2 Released.
+
+ * Test results: http://bamboo.asterisk.org/browse/TESTING-ASTERISK10RCS-3
+
+ * Ensure that a null vmexten does not cause a segfault
+
+ Ensure that a null vmexten does not cause a segfault. When
+ sip_send_mwi_to_peer was modified recently to avoid
+ deadlocks, vmexten was not expected to be null. This change handles
+ that situation to avoid a segfault
+
+ Merged revisions 345063 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * Fixes issue with ConfBridge participants hanging up during DTMF feature
+ menu usage getting stuck in conference forever.
+
+ When a conference user enters the DTMF menu they are suspended from the
+ bridge while the channel is handed off to the DTMF feature code. If a
+ user entered this state and hungup, there existed a race condition where
+ the channel could not exit the conference because it was waiting on a
+ signal that would never arrive. This patch fixes that, because it would
+ stupid for me to talk about the problem and commit a patch for something
+ else.
+
+ (closes issue ASTERISK-18829)
+ Reported by: zvision
+
+ * Fix app_macro.c MODULEINFO section termination.
+
+ (closes issue ASTERISK-18848)
+ Reported by: Tony Mountifield
+
+2011-11-08 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.0.0-rc1 Released.
+
+ * Test results: http://bamboo.asterisk.org/browse/AST10-LUCID-317
+
+2011-11-08 19:27 +0000 [r343944] wdoekes <wdoekes@localhost>:
+
+ * /, pbx/pbx_config.c: Fix crash when dialplan remove include is
+ called with too few arguments. "dialplan remove include x from y"
+ crashed when the amount of arguments was less than 6. (closes
+ issue ASTERISK-18762) Reported by: Andrey Solovyev Tested by:
+ Andrey Solovyev ........ Merged revisions 343936 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-08 18:29 +0000 [r343900] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes regression caused by r343635 There was
+ a missing unlock for a function return that is only present in
+ Asterisk 10 and Asterisk Trunk. (closes issue ASTERISK-18839)
+ Reported by: Michael L. Young Patches:
+ asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch
+ uploaded by Michael L. Young
+
+2011-11-08 18:01 +0000 [r343852] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c, main/acl.c: Fixed reference to incorrect
+ variable if unknown host configured crash. * Fixed a LOG_ERROR
+ message referencing the config variable list v that had
+ previously been processed and became NULL. * Added error return
+ value set that was missing in an ast_append_ha() error return
+ path. (closes issue ASTERISK-18743) Reported by: Michele Patches:
+ issueA18743-fix_dynamic_exclude_static_bad_host_log.patch
+ (license #5674) patch uploaded by Walter Doekes Tested by:
+ Michele ........ Merged revisions 343851 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-08 13:26 +0000 [r343789-343792] Leif Madsen <lmadsen@digium.com>
+
+ * /: Recorded merge of revisions 343791 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Fix
+ boo-boo in prep_tarball script. A hardcoded a branch number was
+ in the prep_tarball which could not work. Changed it to the
+ variable.
+
+ * build_tools/prep_tarball: Fix boo-boo in prep_tarball script. A
+ hardcoded a branch number was in the prep_tarball which could not
+ work. Changed it to the variable.
+
+2011-11-07 22:37 +0000 [r343743] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Make "sip show settings" CLI command get
+ RPID flags from the right global page The "Trust RPID" and "Send
+ RPID" entries in the "sip show settings" CLI command pulled the
+ flags from the incorrect global flags page. These are now read
+ from sip global flags page 0. (closes issue AST-711)
+
+2011-11-07 21:42 +0000 [r343691] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: respect case changes in peer names on sip
+ reload ASTERISK-18669 ........ Merged revisions 343690 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-07 21:27 +0000 [r343677] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Fix __sip_subscribe_mwi_do() incorectly
+ changing dialogs hash key callid. Changing an object value used
+ as a container key requires removing the object from the
+ container and reinserting it. * Created change_callid_pvt() to
+ call instead of build_callid_pvt(). The change_callid_pvt() will
+ correctly change the dialog callid so the ao2 conainter can
+ explicitly unlink it. ........ Merged revisions 343637 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-07 20:31 +0000 [r343635] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Prevent BLF subscriptions from causing
+ deadlocks Fix a locking inversion in sip_send_mwi_to_peer that
+ was causing deadlocks. This function now requires that both the
+ peer and associated pvt be unlocked before it is called for cases
+ where peer and peer->mwipvt form a circular reference. (closes
+ issue ASTERISK-18663) Review:
+ https://reviewboard.asterisk.org/r/1563/ ........ Merged
+ revisions 343621 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-07 19:55 +0000 [r343580] wdoekes <wdoekes@localhost>:
+
+ * main/udptl.c, UPGRADE.txt: Correct the default udptl port range.
+ The udptl port range was defined as 4000-4999 in the
+ udptl.conf.sample, as 4500-4599 if you didn't have a config and
+ 4500-4999 if your config was broken. Default is now 4000-4999.
+ (closes issue ASTERISK-16250) Reviewed by: Tilghman Lesher
+ Review: https://reviewboard.asterisk.org/r/1565
+
+2011-11-07 19:51 +0000 [r343578] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Fix deadlock if peer is destroyed while
+ sending MWI notice. A dialog cannot be destroyed by the
+ ao2_callback dialog_needdestroy because of a deadlock between the
+ dialogs container lock and the RWLOCK of the events subscription
+ list. * Create dialogs_to_destroy container to hold dialogs that
+ will be destroyed. * Ensure that the event subscription callback
+ will never happen with an invalid peer pointer by making the
+ event callback removal the first thing in the peer destructor
+ callback. NOTE: This particular deadlock will not happen with
+ Asterisk 10, but some of the changes still apply. (closes issue
+ ASTERISK-18747) Reported by: Gregory Hinton Nietsky Review:
+ https://reviewboard.asterisk.org/r/1564/ ........ Merged
+ revisions 343577 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-07 18:39 +0000 [r343533] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/format.c: list all of the codecs associated with a
+ particular format id for CLI command "core show codec" AST-699
+
+2011-11-04 15:11 +0000 [r343445] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooGkClient.c,
+ addons/ooh323c/src/ooTimer.c, addons/ooh323c/src/dlist.c, /,
+ addons/ooh323c/src/dlist.h, addons/ooh323c/src/printHandler.c:
+ Final fix memleaks in GkClient codes, same for Timer codes.
+ (these memleaks stop development of gk codes, now i can continue)
+ Fix printHandler 'Unbalanced Structure' issues with locking
+ printHandler data for single thread. ........ Merged revisions
+ 343281 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-03 20:31 +0000 [r343393] wdoekes <wdoekes@localhost>:
+
+ * /, res/res_config_sqlite.c: Fix sqlite config driver segfault and
+ broken queries The sqlite realtime handler assumed you had a
+ static config configured as well. The realtime multientry handler
+ assumed that you weren't using dynamic realtime. (closes issue
+ ASTERISK-18354) (closes issue ASTERISK-18355) Review:
+ https://reviewboard.asterisk.org/r/1561 ........ Merged revisions
+ 343375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-03 19:57 +0000 [r343337] Richard Mudgett <rmudgett@digium.com>
+
+ * /, funcs/func_dialgroup.c: Remove invalid flag given to iterator
+ in func_dialgroup.c ........ Merged revisions 343336 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-03 15:39 +0000 [r343221-343277] Terry Wilson <twilson@digium.com>
+
+ * /, channels/sip/include/sip.h: Make room for the fax detect flags
+ The original REGISTERTRYING flag, in addition to being impossible
+ to check, also encroached on the space for the flag above it.
+ This patch moves the flags that were below REGISTERTRYING back to
+ where they were as though we had just removed the REGISTERTRYING
+ option. ........ Merged revisions 343276 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * contrib/realtime/mysql/sippeers.sql, /, channels/chan_sip.c,
+ channels/sip/include/sip.h: Remove registertrying option in
+ chan_sip This option is not only useless, but has been broken
+ since inception since the flag was never copied from the peer
+ where it is set to the pvt where it was checked. RFC 3261
+ specificially states that you should not send a provisional
+ response to a non-INVITE request, and if we did fix the code so
+ that it worked, it would cause the same kind of user enumeration
+ vulnerability that we've discussed with the nat= setting. This
+ patch removes registertrying option and any code that would have
+ sent a 100 response to a register. Review:
+ https://reviewboard.asterisk.org/r/1562/ ........ Merged
+ revisions 343220 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-02 22:24 +0000 [r343158-343192] wdoekes <wdoekes@localhost>:
+
+ * /, channels/chan_sip.c: Fix improper warning introduced by
+ r342927 and more tweaks Changeset r342927 introduced a warning
+ which was only supposed to be emitted when a found realtime peer
+ had an empty (or no) name. It turned out that there were some
+ inconsistencies left. Now found peers with an empty name are
+ explicitly ignored like before r342927 but better. Reviewed by:
+ Stefan Schmidts, Terry Wilson Review:
+ https://reviewboard.asterisk.org/r/1560 ........ Merged revisions
+ 343181 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * include/asterisk/stringfields.h, include/asterisk/utils.h, /,
+ main/utils.c: Ensure that string field lengths are properly
+ aligned Integers should always be aligned. For some platforms
+ (ARM, SPARC) this is more important than for others. This
+ changeset ensures that the string field string lengths are
+ aligned on *all* platforms, not just on the SPARC for which there
+ was a workaround. It also fixes that the length integer can be
+ resized to 32 bits without problems if needed. (closes issue
+ ASTERISK-17310) Reported by: radael, S Adrian Reviewed by:
+ Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review:
+ https://reviewboard.asterisk.org/r/1549 ........ Merged revisions
+ 343157 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-02 19:33 +0000 [r343048-343103] Leif Madsen <lmadsen@digium.com>
+
+ * /, apps/app_authenticate.c: Add note about how Authenticate()
+ application with option 'd' works. (closes issue ASTERISK-17422)
+ Reported by: Leif Madsen ........ Merged revisions 343102 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, configs/queues.conf.sample: Update documentation for
+ leastrecent strategy. In queues.conf.sample the leastrecent
+ strategy was incorrectly described. Now updated to reflect how
+ the strategy actually checks peers. (closes issue ASTERISK-17854)
+ Reported by: Sebastian Denz Patches: queues.conf-doc_issue.patch
+ (License #6139) ........ Merged revisions 343047 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-02 13:45 +0000 [r342991] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_meetme.c: Modify comments in MeetMe application
+ documentation about DAHDI. The MeetMe application documentation
+ has some comments about usage of DAHDI, and they were a bit
+ outdated relative to modern DAHDI releases. This patch changes
+ the comment to just tell the user that a functional DAHDI timing
+ source is required, and no longer mention 'dahdi_dummy', since
+ that module does not exist in current DAHDI releases. ........
+ Merged revisions 342990 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-01 20:58 +0000 [r342870-342929] wdoekes <wdoekes@localhost>:
+
+ * /, channels/chan_sip.c, configs/extconfig.conf.sample,
+ include/asterisk/config.h, main/config.c: Several fixes to the
+ chan_sip dynamic realtime peer/user lookup There were several
+ problems with the dynamic realtime peer/user lookup code. The
+ lookup logic had become rather hard to read due to lots of
+ incremental changes to the realtime_peer function. And, during
+ the addition of the sipregs functionality, several possibilities
+ for memory leaks had been introduced. The insecure=port matching
+ has always been broken for anyone using the sipregs family. And,
+ related, the broken implementation forced those using sipregs to
+ *still* have an ipaddr column on their sippeers table. Thanks
+ Terry Wilson for comprehensive testing and finding and fixing
+ unexpected behaviour from the multientry realtime call which
+ caused the realtime_peer to have a completely unused code path.
+ This changeset fixes the leaks, the lookup inconsistenties and
+ that you won't need an ipaddr column on your sippeers table
+ anymore (when you're using sipregs). Beware that when you're
+ using sipregs, peers with insecure=port will now start matching!
+ (closes issue ASTERISK-17792) (closes issue ASTERISK-18356)
+ Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry
+ Wilson Review: https://reviewboard.asterisk.org/r/1395 ........
+ Merged revisions 342927 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * contrib/realtime/mysql/sipfriends.sql (removed),
+ contrib/realtime/mysql/sippeers.sql (added),
+ configs/res_config_mysql.conf.sample, /,
+ configs/extconfig.conf.sample, configs/res_ldap.conf.sample,
+ res/res_realtime.c, UPGRADE-1.8.txt, configs/dbsep.conf.sample,
+ main/config.c: Cleanup references to sipusers and sipfriends
+ dynamic realtime families Somewhere between 1.4 and 1.8 the
+ sipusers family has become completely unused. Before that, the
+ sipfriends family had been obsoleted in favor of separate
+ sipusers and sippeers families. Apparently, they have been merged
+ back again into a single family which is now called "sippeers".
+ Reviewed by: irroot, oej, pabelanger Review:
+ https://reviewboard.asterisk.org/r/1523 ........ Merged revisions
+ 342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-31 17:46 +0000 [r342824] Richard Mudgett <rmudgett@digium.com>
+
+ * main/format.c, main/format_cap.c: Misc format capability fixes. *
+ Fixed typo in format_cap.c:joint_copy_helper() using the wrong
+ variable. * Fix potential race between checking if an interface
+ exists and adding it to the container in
+ format.c:ast_format_attr_reg_interface(). * Fixed double rwlock
+ destroy in format.c:ast_format_attr_init() error exit path. *
+ Simplified format.c:find_interface() and
+ format.c:has_interface().
+
+2011-10-31 16:04 +0000 [r342770] Matthew Jordan <mjordan@digium.com>
+
+ * main/pbx.c, /, channels/chan_iax2.c: Fixed invalid memory access
+ when adding extension to pattern match tree When an extension is
+ removed from a context, its entry in the pattern match tree is
+ not deleted. Instead, the extension is marked as deleted. When an
+ extension is removed and re-added, if that extension is also a
+ prefix of another extension, several log messages would report an
+ error and did not check whether or not the extension was deleted
+ before accessing the memory. Additionally, if the extension was
+ already in the tree but previously deleted, and the pattern was
+ at the end of a match, the findonly flag was not honored and the
+ extension would be erroneously undeleted. Additionaly, it was
+ discovered that an IAX2 peer could be unregistered via the CLI,
+ while at the same time it could be scheduled for unregistration
+ by Asterisk. The unregistration method now checks to see if the
+ peer was already unregistered before continuing with an
+ unregistration. (closes issue ASTERISK-18135) Reported by: Jaco
+ Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/1526 ........ Merged
+ revisions 342769 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-30 02:21 +0000 [r342715] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar.c: Don't crash on empty notify channel
+
+2011-10-29 04:26 +0000 [r342662] Richard Mudgett <rmudgett@digium.com>
+
+ * /, include/asterisk/linkedlists.h, tests/test_linkedlists.c: Fix
+ AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable.
+ AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an
+ iteration or before AST_LIST_REMOVE_CURRENT() without corrupting
+ the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the
+ list if AST_LIST_INSERT_BEFORE_CURRENT() or
+ AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed
+ cut and paste error using the wrong variable in
+ AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests
+ for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and
+ AST_LIST_INSERT_LIST_AFTER(). ........ Merged revisions 342661
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-27 20:10 +0000 [r342605] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/dsp.c: tweak the v21 detector to detect an additional
+ pattern of hits and misses
+
+2011-10-27 19:41 +0000 [r342546-342603] Jonathan Rose <jrose@digium.com>
+
+ * res/res_rtp_multicast.c, /: Fix sequence number overflow over 16
+ bits causing codec change in RTP packets. Sequence number was
+ handled as an unsigned integer (usually 32 bits I think, more
+ depending on the architecture) and was put into the rtp packet
+ which is basically just a bunch of bits using an or operation.
+ Sequence number only has 16 bits allocated to it in an RTP packet
+ anyway, so it would add to the next field which just happened to
+ be the codec. This makes sure the sequence number is set to be a
+ 16 bit integer regardless of architecture (hopefully) and also
+ makes it so the incrementing of the sequence number does bitwise
+ or at the peak of a 16 bit number so that the value will be set
+ back to 0 when going beyond 65535 anyway. (closes issue
+ ASTERISK-18291) Reported by: Will Schick Review:
+ https://reviewboard.asterisk.org/r/1542/ ........ Merged
+ revisions 342602 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, res/res_jabber.c: Cleanup reference leaks in res_jabber
+ res_jabber.c had a number of places where astobjs would be
+ referenced and have their reference counts bumped without having
+ a dereference made before the object lost scope. This patch adds
+ a number of ASTOBJ_UNREFs to resolve that. Review:
+ https://reviewboard.asterisk.org/r/1478/ ........ Merged
+ revisions 342545 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-25 22:05 +0000 [r342485-342488] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/astobj2.c: Check fopen return value for ao2 reference
+ debug output. Reported by: wdoekes Patched by: wdoekes Review:
+ https://reviewboard.asterisk.org/r/1539/ ........ Merged
+ revisions 342487 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/sig_pri.c: Change D-channel warning to be less
+ confusing on non-NFAS setups. The "No D-channels available! Using
+ Primary channel as D-channel anyway!" WARNING message has been
+ confusing on non-NFAS setups. The message refers to things that
+ are NFAS specific. * Changed the warning to several different
+ warnings to be more accurate for the situation and less confusing
+ as a result: "No D-channels up! Switching selected D-channel from
+ X to Y.", "No D-channels up!", and "D-channel is down!". ........
+ Merged revisions 342484 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-25 21:10 +0000 [r342381-342436] Terry Wilson <twilson@digium.com>
+
+ * /, apps/app_queue.c: Use int for storing ao2_container_count
+ instad of size_t AST-676 ........ Merged revisions 342435 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_queue.c: Simplify queue membercount code Despite an
+ ominous sounding comment stating that membercount was for "logged
+ in" members only and thus we couldn't use ao2_container_count(),
+ I could not find a single place in the code where that seemed to
+ be accurate. The only time we decremented membercount was when we
+ were marking something dead or actually removing it. The only
+ places we incremented it were either after ao2_link(), or trying
+ to correct for having set it to 0 during a reload. In every case
+ where we were correcting the value, it seemed that we were trying
+ to make the count actually match what ao2_container_count() would
+ return. The only place I could find where we made a determination
+ about something being "logged in" or not, we didn't trust the
+ membercount, but instead looked at devicestate, paused, etc. This
+ patch removes membercount, replaces its use with
+ ao2_container_count, and manually adds the results of
+ ao2_container_count to a "membercount" field for ast_data queue
+ query results. This patch also would fix AST-676, but as it is
+ slightly riskier than the previously committed fix, the two
+ commits have been made separately. Reivew:
+ https://reviewboard.asterisk.org/r/1541/ ........ Merged
+ revisions 342383 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_queue.c: Properly update membercount for reloaded
+ members Since q->membercount is set to 0 before reloading, it is
+ important to increment it again for reloaded members as well as
+ added. (closes issue AST-676) Review:
+ https://reviewboard.asterisk.org/r/1541/ ........ Merged
+ revisions 342380 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-25 19:08 +0000 [r342277-342329] Kinsey Moore <kmoore@digium.com>
+
+ * pbx/pbx_spool.c, /: Fix compilation on Snow Leopard/FreeBSD for
+ pbx_spool.c One of the changes in the recent spool handling of
+ hardlinks patch was just outside a HAVE_INOTIFY block and caused
+ compilation to fail in some build environments. This has been
+ corrected. ........ Merged revisions 342328 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * pbx/pbx_spool.c, /: Merged revisions 342276 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r342276 | kmoore | 2011-10-25 11:06:57 -0500 (Tue, 25 Oct 2011) |
+ 18 lines Fix spool handling to allow call files to be hardlinked
+ into place This fixes the inotify code to handle call files being
+ hardlinked into the spool directory. The smsq utility does this,
+ instead of rename(), to ensure that it cannot accidentally
+ overwrite an existing spool file. A rename() might do that, but
+ link() will definitely not. The inotify code had broken this,
+ because it would wait for an IN_CLOSE_WRITE event on the file...
+ which was never forthcoming, since it was never opened. Now we
+ look for IN_OPEN events following the IN_CREATE event, and only
+ wait for an IN_CLOSE_WRITE if the file was actually opened.
+ Patch-by: dwmw2 (closes issue ASTERISK-18331) Review:
+ https://reviewboard.asterisk.org/r/1391/ ........
+
+2011-10-25 01:25 +0000 [r342224] Terry Wilson <twilson@digium.com>
+
+ * /, include/asterisk/config.h, main/config.c: Return NULL when no
+ results returned for realtime_multientry It was not documented
+ what the return value should be when no entries were returned
+ with the multientry realtime callback. This change forces
+ consistent behavior even if the backends return an empty
+ ast_config. Review: https://reviewboard.asterisk.org/r/1521/
+ ........ Merged revisions 342223 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-24 22:32 +0000 [r342183] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/astobj2.h: Fix ao2obj.h comment typos and add
+ missing link/unlink nolock debug defines.
+
+2011-10-24 19:51 +0000 [r342062] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Outbound SIP OPTIONS messages will now
+ include fromuser of related peer. This behavior matches up more
+ closely with the way invite/register/etc are handled. This patch
+ also modifies some adjacent code for code style compliance.
+ Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy
+ Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded
+ by Jeremy Kister (license #6232) ........ Merged revisions 342061
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-24 07:31 +0000 [r341920-342017] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * apps/app_queue.c: queues container needs locking when using the
+ OBJ_NOLOCK flag
+
+ * apps/app_queue.c: Remove some ref leaks and a return without
+ unlock. There some resource leaks introduced in asterisk 10 make
+ sure that locks are not held on return and we release ref's held.
+
+ * /, apps/app_queue.c: Revert Janitor patch 341920 For now
+
+ * /, apps/app_queue.c: Whitespace Fixups / Add Braces This
+ janitorial patch is related to work on RB1538 ........ Merged
+ revisions 341906 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-21 16:42 +0000 [r341807-341810] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, pbx/pbx_lua.c: only process args that exist ASTERISK-18395
+ ........ Merged revisions 341809 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, pbx/pbx_lua.c: don't limit the length of app and function
+ arguments ASTERISK-18395 ........ Merged revisions 341806 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-20 21:58 +0000 [r341718] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/features.h, /, main/features.c, res/res_agi.c:
+ Fix AGI exec Park to honor the Park application parameters. The
+ fix for ASTERISK-12715 and ASTERISK-12685 added a check for the
+ Park application because the channel needed to be masqueraded to
+ prevent a crash. Since the Park application now always
+ masquerades the channel into the parking lot, the special check
+ is no longer needed. The fix also resulted in AGI exec Park
+ attempting to double park the call and not honor the Park
+ application parameters. * Removed no longer necessary call to
+ ast_masq_park_call() by AGI exec for the Park application.
+ (Reverts -r146923) * Fix Park application to only return 0 or -1.
+ The AGI exec Park was causing broken pipe error messages because
+ the Park application returned 1 on successful park. (closes issue
+ ASTERISK-18737) ........ Merged revisions 341717 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-20 21:27 +0000 [r341665-341707] Paul Belanger <pabelanger@digium.com>
+
+ * /, funcs/func_callerid.c: Fixed typo from previous commit
+ ........ Merged revisions 341704 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, funcs/func_callerid.c: Updated documentation for the optional
+ CID parameter with CALLERID ........ Merged revisions 341664 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-20 18:20 +0000 [r341580-341599] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * configs/queues.conf.sample: add documentation for
+ check_state_unknown in configs/queues.conf.sample app_queue
+ allows calls to members in a "Unknown" state to be treated as
+ available setting check_state_unknown = yes will cause app_queue
+ to query the channel driver to better determine the state this
+ only applies to queues with ringinuse or ignorebusy set
+ appropriately.
+
+ * CHANGES, apps/app_queue.c: Add option to check state when state
+ is unknown r341486 reverts r325483 this is a rework of the patch.
+ optimize to minimize load. add option check_state_unknown to
+ control whether a member with unknown device state is checked
+ there is a small % chance that calls will be sent to the member
+ when they on a call. app_queue will see a device with unknown
+ state as available and does not try verify the state without this
+ option enabled. Review: https://reviewboard.asterisk.org/r/1535/
+
+2011-10-20 15:14 +0000 [r341530] Terry Wilson <twilson@digium.com>
+
+ * /, include/asterisk/strings.h: Clean up ast_check_digits The code
+ was originally copied from the is_int() function in the AEL code.
+ wdoekes pointed out that the function should take a const char*
+ and that their was an unneeded variable. This is now fixed.
+ ........ Merged revisions 341529 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-19 21:23 +0000 [r341486] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c: Fix a performance regression introduced in
+ r325483. The regression was caused by a call to
+ ast_parse_device_state() in app_queue's ring_entry() function.
+ The ast_parse_device_state() function eventually calls
+ ast_channel_get_full() with a channel name prefix which causes it
+ to walk the channel list causing massive lock contention and slow
+ downs. This patch fixes the regression by removing the call to
+ ast_parase_device_state() which should be unnecessary. Queue
+ member device state should be maintained by device state events.
+ Some users have seen instances where busy agents were called when
+ they shouldn't have, which is the reason the call to
+ ast_parse_device_state() was added. That change appears to have
+ resolved that issue but also causes this performance regression.
+ There may still be issues with queue member status, and if so,
+ alternative methods should be investigated to resolve them.
+ AST-695
+
+2011-10-19 19:01 +0000 [r341436] Paul Belanger <pabelanger@digium.com>
+
+ * /, channels/chan_gtalk.c: Outgoing calls with Google Voice Google
+ has recently make some changes (again) to their protocol. Rather
+ then patching asterisk to flip between the two different methods,
+ we now allow both. Lets hope this keeps Google Voice happy for a
+ while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov
+ Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses
+ 6311) ........ Merged revisions 341435 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-19 07:42 +0000 [r341380] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c, include/asterisk/strings.h: Don't use
+ is_int() since it doesn't link well on all platforms Just create
+ an normal API function in strings.h that does the same thing just
+ to be safe. ASTERISK-17146 ........ Merged revisions 341379 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-19 07:23 +0000 [r341377] Stefan Schmidt <sst@sil.at>
+
+ * /, channels/chan_sip.c: Don't sent in-dialog requests like UPDATE
+ when Asterisk has not yet received a Contact URI from a UAS
+ ........ Merged revisions 341366 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-18 23:42 +0000 [r341315] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Don't resolve numeric hosts or contact
+ unresolved hosts If a SIP dial string contains a numeric hostname
+ that is not a peer name, don't try to resolve it as it is
+ unlikely that someone really means Dial(SIP/0.0.4.26) when
+ Dial(SIP/1050) is called. Also, make sure that create_addr
+ returns -1 if an address isn't resolved so that we don't attempt
+ to send SIP requests to an address that doesn't resolve. (closes
+ issue ASTERISK-17146, ASTERISK-17716) Review:
+ https://reviewboard.asterisk.org/r/1532/ ........ Merged
+ revisions 341314 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-18 23:33 +0000 [r341313] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: Merged revisions 341312 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r341312 | may | 2011-10-19 03:20:53 +0400 (Wed, 19 Oct
+ 2011) | 3 lines fix issue on channel numbering (calls could have
+ same channel number on heavy loaded system) ........
+
+2011-10-18 21:11 +0000 [r341255] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_mgcp.c, include/asterisk/features.h,
+ channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/chan_sip.c, main/features.c, channels/chan_iax2.c,
+ channels/sip/include/sip.h: More parking issues. * Fix potential
+ deadlocks in SIP and IAX blind transfer to parking. * Fix SIP,
+ IAX, DAHDI analog, and MGCP channel drivers to respect the
+ parkext_exclusive option with transfers (Park(,,,,,exclusive_lot)
+ parameter). Created ast_park_call_exten() and
+ ast_masq_park_call_exten() to maintian API compatibility. * Made
+ masq_park_call() handle a failed ast_channel_masquerade() setup.
+ * Reduced excessive struct parkeduser.peername[] size. ........
+ Merged revisions 341254 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-17 17:36 +0000 [r341190] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Initialize variables before calling
+ parse_uri If parse_uri was called with an empty URI, some
+ pointers would be modified and an invalid read could result. This
+ patch avoids calling parse_uri with an empty contact uri when
+ parsing REGISTER requests. AST-2011-012 (closes issue
+ ASTERISK-18668) ........ Merged revisions 341189 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-17 16:53 +0000 [r341148] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, pbx/pbx_realtime.c: Remove an unused include of md5.h Unused
+ include of asterisk/md5.h in pbx_realtime.c . A commit needed to
+ test the commit message. Merged-From:
+ http://svn.asterisk.org/svn/asterisk/branches/1.8@341074
+
+2011-10-17 16:38 +0000 [r341122-341146] Paul Belanger <pabelanger@digium.com>
+
+ * tests/test_format_api.c: Set 'core' support level for
+ test_format_api.c
+
+ * apps/app_voicemail.c, /: Multiple revisions 341108,341112
+ ........ r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon,
+ 17 Oct 2011) | 2 lines Voicemail compiler flags are 'core'
+ support ........ r341112 | pabelanger | 2011-10-17 12:23:33 -0400
+ (Mon, 17 Oct 2011) | 2 lines Fix previous commit ........ Merged
+ revisions 341108,341112 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-17 16:18 +0000 [r341094] Jason Parker <jparker@digium.com>
+
+ * CHANGES: Add information about limitations of new codec support
+ in channel drivers. (issue ASTERISK-18680)
+
+2011-10-17 15:39 +0000 [r341089] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Don't try to remove peers without IPs
+ from peers_by_ip (closes issue ASTERISK-18696) ........ Merged
+ revisions 341088 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-14 21:36 +0000 [r341023] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, build_tools/embed_modules.xml, Makefile.moddir_rules: Change
+ the internal name of the menuselect options that are used to
+ control whether modules are embedded or not; using just the bare
+ category name led to accidentally enabling these options when
+ users used the wrong "--enable" operation on the menuselect
+ command line. Now the internal option names are prefixed with
+ "EMBED_", so they won't be the same as the name of the category
+ containing the modules they control the embedding of. ........
+ Merged revisions 341022 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-14 20:50 +0000 [r340971] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions
+ 340970 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) |
+ 8 lines Quiet RTCP Receiver Reports during fax transmission RTCP
+ is now disabled for "inactive" RTP audio streams during SIP T.38
+ sessions. The ability to disable RTCP streams in res_rtp_asterisk
+ was missing, so this code was added to support the bug fix.
+ (closes issue ASTERISK-18400) ........
+
+2011-10-14 18:23 +0000 [r340931] Jonathan Rose <jrose@digium.com>
+
+ * utils/utils.xml, funcs/func_jitterbuffer.c: Some additional
+ module documentation changes for 10 for the menuselect change.
+ (issue ASTERISK-18268)
+
+2011-10-14 16:39 +0000 [r340879] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Avoid unnecessary WARNING message Add
+ AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid
+ displaying a WARNING message. (closes issue ASTERISK-18610) Patch
+ by: Kristijan_Vrban ........ Merged revisions 340878 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-14 16:18 +0000 [r340868] Jonathan Rose <jrose@digium.com>
+
+ * funcs/func_realtime.c, build_tools/cflags.xml, utils/utils.xml,
+ /, res/res_fax.c, apps/app_celgenuserevent.c,
+ codecs/codec_dahdi.c, apps/app_system.c, res/res_curl.c: Fixes
+ some support level info so that it can be read by menuselect.
+ (issue ASTERISK-18268) Review:
+ https://reviewboard.asterisk.org/r/1525/ ........ Merged
+ revisions 340863 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-13 22:54 +0000 [r340810] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Fix DTMF blind transfer continuing to execute
+ dialplan after transfer. Party A calls Party B. Party A DTMF
+ blind transfers Party B to Party C. Party A channel continues to
+ execute dialplan. * Fixed the return value of
+ builtin_blindtransfer() to return the correct value after a
+ transfer so the dialplan will not keep executing. * Removed
+ unnecessary connected line update that did not really do
+ anything. * Made access to GOTO_ON_BLINDXFR thread safe in
+ check_goto_on_transfer(). * Fixed leak of xferchan for failure
+ cases in check_goto_on_transfer(). * Updated debug messages in
+ builtin_blindtransfer() and check_goto_on_transfer(). (closes
+ issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett
+ ........ Merged revisions 340809 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-13 08:46 +0000 [r340770] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * channels/chan_sip.c: Only send MWI Notify on register if the
+ registration is successful. lastmsgssent was removed from
+ chan_sip and the old behavior of sending a mwi notify on register
+ [except when subscribemwi is set] was restored but this must only
+ happen when registration succeeds. leaking information for
+ unsuccessful registrations is not secure.
+
+2011-10-13 06:59 +0000 [r340718] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: Merged revisions 340717 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13 Oct 2011)
+ | 3 lines storing the route-set also on a 181 response not only
+ on 180,182 or 183. ........
+
+2011-10-13 06:56 +0000 [r340578-340716] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Initialize ast_sockaddr before calling
+ ast_sockaddr_resolve Avoid possible jump based on unitialized
+ value ........ Merged revisions 340715 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, res/res_config_sqlite.c: Don't skip the query field on a
+ realtime multi query There is no documented reason to not add the
+ query field to the varlist returned by a realtime multi query,
+ despite the config category being set to its value. Of course,
+ there is no documentation that the category should be set to the
+ value either. There is lots of no documentation when it comes to
+ realtime. But, other engines do not skip this field so I am
+ forcing this backend to follow the convention, because not doing
+ so is very silly. ........ Merged revisions 340662 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Merged revisions 340534 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011)
+ | 9 lines Update SIP realtime fullcontact regardless of caching
+ We should update the fullcontact field in the realtime table
+ whether or not rtcachefriends is set. There is no reason to treat
+ a non-cached realtime entity differently than a cached in this
+ regard. (closes issue ASTERISK-18446) Reported by: wdoekes
+ ........
+
+2011-10-12 20:33 +0000 [r340577] Stefan Schmidt <sst@sil.at>
+
+ * /, channels/chan_sip.c: Merged revisions 340576 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r340576 | schmidts | 2011-10-12 20:30:37 +0000 (Mit, 12 Okt 2011)
+ | 3 lines Store route-set from provisional SIP responses so
+ early-dialog requests can be routed properly ........
+
+2011-10-12 20:08 +0000 [r340471-340523] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Initialize the PRI channel alarms
+ properly on startup. The PRI channel alarms were initialized with
+ an inverted sense. (closes issue ASTERISK-18710) Reported by:
+ Tzafrir Cohen ........ Merged revisions 340522 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_meetme.c: Update MeetMe p and X option documentation
+ when interacting with the s option. ASTERISK-12175 changed the p
+ and X options to not interfere with the s option when they are
+ used together. It makes more sense for the s option to have
+ priority for the DTMF '*' key since it cannot change its
+ activation code. Otherwise, you could not use option s with the p
+ or X options. JIRA AST-671 ........ Merged revisions 340470 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-12 16:28 +0000 [r340419] Paul Belanger <pabelanger@digium.com>
+
+ * /, channels/chan_sip.c: Fix verbose messages when IPv6 logic was
+ added (closes issue ASTERISK-18612) Reported by: Tim Osman
+ ........ Merged revisions 340418 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-11 21:05 +0000 [r340281-340366] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c:
+ Add protection for SS7 channel allocation and better glare
+ handling. * Added a CLI "ss7 show channels" command that might
+ prove useful for future debugging. * Made the incoming SS7
+ channel event check and gripe message uniform. * Made sure that
+ the DNID string for an incoming call is always initialized.
+ (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven
+ Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621)
+ patch uploaded by rmudgett ........ Merged revisions 340365 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/sip/include/dialog.h, /, channels/chan_sip.c: Fix some
+ potential deadlocks pointed out by helgrind. * Fixed deadlock
+ potential calling dialog_unlink_all() in __sip_autodestruct().
+ Found by helgrind. * Fixed deadlock potential in
+ handle_request_invite() after calling sip_new(). Found by
+ helgrind. * The sip_new() function now returns with the created
+ channel already locked. * Removed the dead code that starts a PBX
+ in in sip_new(). No sip_new() callers caused that code to be
+ executed and it was a bad thing to do anyway. * Removed unused
+ parameters and return value from dialog_unlink_all(). * Made
+ dialog_unlink_all() and __sip_autodestruct() safely obtain the
+ owner and private channel locks without a deadlock avoidance
+ loop. ........ Merged revisions 340284 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/manager.c, /, include/asterisk/manager.h: Convert registered
+ AMI actions to ao2 objects. * Fixed race between calling an AMI
+ action callback and unregistering that action. Refixes
+ ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential
+ memory leak if an AMI action failed to get registered because is
+ already was registered. Part of the ao2 conversion. * Fixed AMI
+ ListCommands action not walking the actions list with a lock
+ held. * Fix usage of ast_strdupa() and alloca() in loops. Excess
+ stack usage. * Fix AMI Originate action Variable header requiring
+ a space after the header colon. Reported by Yaroslav Panych on
+ the asterisk-dev list. * Increased the number of listed variables
+ allowed per AMI Originate action Variable header to 64. * Fixed
+ AMI GetConfigJSON action output format. * Fixed usage of res
+ contents outside of scope in append_channel_vars(). * Fixed
+ inconsistency of config file channelvars option. The values no
+ longer accumulate with every channelvars option in the config
+ file. Only the last value is kept to be consistent with the CLI
+ "manager show settings" command. (closes issue ASTERISK-18479)
+ Reported by: Jaco Kroon ........ Merged revisions 340279 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-11 18:41 +0000 [r340280] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * main/channel.c, /, main/sha1.c, include/asterisk/sha1.h: Update
+ SHA1 code to RFC 6234 RFC 6234 is an update to RFC 3174 from
+ which the code was originally taken. It has a slightly better
+ code, and a better phrased license (simple 3-clause BSD). *
+ main/sha1.c is sha1.c from RFC 6234 with formatting changes only.
+ * include/asterisk/sha1.h merges sha.h and sha-private.h from RFC
+ 6234. * Removed unused include of asterisk/sha1.h from
+ main/channels.c Review: https://reviewboard.asterisk.org/r/1503/
+ Merge-From:
+ http://svn.asterisk.org/svn/asterisk/branches/1.8@340263
+
+2011-10-10 22:55 +0000 [r340219-340222] Terry Wilson <twilson@digium.com>
+
+ * main/db.c: On astdb conversion, also warn about permissions
+ requirements The user running Asterisk must have permission to
+ the directory the Asterisk database resides in since SQLite 3
+ needs to be able to create a journal file. (closes issue
+ ASTERISK-18174)
+
+ * utils/astdb2bdb.c (added): Add a missing file for the astdb2bdb
+ conversion utility
+
+ * utils/Makefile, utils/utils.xml, UPGRADE.txt: Add astdb
+ conversion utility for Berkeley to SQLite 3 If someone wants to
+ backtrack from Asterisk 1.8 to 10 they can use the astdb2bdb
+ utility to convert the database back to the Berkeley format that
+ Asterisk 1.8 uses. Review:
+ https://reviewboard.asterisk.org/r/1502/
+
+2011-10-10 20:30 +0000 [r340165] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 340164 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r340164 | mjordan | 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011)
+ | 13 lines Updated chan_sip to place calls on hold if SDP address
+ in INVITE is ANY This patch fixes the case where an INVITE is
+ received with c=0.0.0.0 or ::. In this case, the call should be
+ placed on hold. Previously, we checked for the address being
+ null; this patch keeps that behavior but also checks for the ANY
+ IP addresses. Review: https://reviewboard.asterisk.org/r/1504/
+ (closes issue ASTERISK-18086) Reported by: James Bottomley Tested
+ by: Matt Jordan ........
+
+2011-10-10 14:15 +0000 [r340109] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/loader.c, main/xmldoc.c, main/pbx.c, main/manager.c, /,
+ res/res_fax.c, apps/app_fax.c, include/asterisk/module.h,
+ res/res_agi.c, include/asterisk/xmldoc.h, doc/appdocsxml.dtd:
+ Merged revisions 340108 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct
+ 2011) | 11 lines Load the proper XML documentation when multiple
+ modules document the same application. This patch adds an
+ optional "module" attribute to the XML documentation spec that
+ allows the documentation processor to match apps with identical
+ names from different modules to their documentation. This patch
+ also fixes a number of bugs with the documentation processor and
+ should make it a little more efficient. Support for multiple
+ languages has also been properly implemented. ASTERISK-18130
+ Review: https://reviewboard.asterisk.org/r/1485/ ........
+
+2011-10-09 22:18 +0000 [r339992-340031] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Return -1 to skinny_session if register
+ rejected. If device registration is rejected, return -1 so that
+ the session is destroyed immediately. Previously, a segfault
+ would occur on a graceful shutdown if a register is rejected and
+ the skinny_session has not yet timed out.
+
+ * channels/chan_skinny.c: Remove log message on traverse session
+ list. On destroying a session, a list of sessions is traversed to
+ find the matching session. For each session not matching, skinny
+ erroneously logged that the session was not matched. While
+ technically correct the message was misleading, and tended to
+ indicate errors that were not there.
+
+2011-10-09 01:18 +0000 [r339831-339942] igorg <igorg@localhost>:
+
+ * channels/chan_unistim.c, /: Merged revisions 339938 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r339938 | igorg | 2011-10-09 08:16:09 +0700 (Вск, 09 Окт
+ 2011) | 6 lines Fix compilation issue, caused by missed session
+ structure (closes issue ASTERISK-18694) Reported by: alex70
+ ........
+
+ * channels/chan_unistim.c, /: Merged revisions 339884 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт
+ 2011) | 7 lines Fix segfault in Unistim channel (closes issue
+ ASTERISK-18638) Reported by: jonnt ........
+
+ * channels/chan_unistim.c, /: Merged revisions 339830 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт
+ 2011) | 8 lines Fix char array cast as short array in
+ send_client() function (for ARM platform) (closes issue
+ ASTERISK-17314) Reported by: jjoshua ........
+
+2011-10-07 19:36 +0000 [r339777] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_url.c: Merged revisions 339776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339776 | rmudgett | 2011-10-07 14:34:55 -0500 (Fri, 07 Oct 2011)
+ | 5 lines Initialize option flags for SendURL application.
+ (closes issue ASTERISK-18574) Reported by: marcelloceschia
+ ........
+
+2011-10-06 23:08 +0000 [r339722] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Reject v17 skinny devices in Asterisk10
+ Small fix for Asterisk10 to reject skinny devices with skinny
+ firmware version17 and above. Review:
+ https://reviewboard.asterisk.org/r/1497/
+
+2011-10-06 22:58 +0000 [r339720] Richard Mudgett <rmudgett@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ autoconf/ast_ext_lib.m4: Merged revisions 339719 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06
+ Oct 2011) | 20 lines Fix regression in configure script for
+ libpri capability checks. JIRA AST-598 added the
+ PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer 2 persistence
+ issues with some telcos. ASTERISK-18535 attempted to fix the
+ unexpected requirement that libpri *must* have that feature to
+ work with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made
+ the PRI optional features required. Unfortunately, I thought
+ AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri
+ and deleted those lines for libpri. The result was the
+ HAVE_PRI_xxx defines that control the ability to use optional
+ libpri features were also deleted. * Created
+ AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional
+ features in a library that the source code could take advantage
+ of if the code supports the feature. (closes issue
+ ASTERISK-18687) Reported by: Norbert Tested by: rmudgett ........
+
+2011-10-06 20:47 +0000 [r339681] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fixed segfault on core stop gracefully.
+ There was an issue that the cap and confcap pointers for each
+ line and device were being memcpy'd so they all pointed to the
+ same ast_format_cap. On destroying, a segfault occured on the
+ second call to the same struct. skinny reload now works again as
+ well. Tested by snuff (in trunk) and myself.
+
+2011-10-06 17:53 +0000 [r339626] Richard Mudgett <rmudgett@digium.com>
+
+ * main/udptl.c, /, channels/chan_sip.c: Merged revisions 339625 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011)
+ | 18 lines Fix debugging messages generated by 'udptl debug'. *
+ Makes chan_sip set the tag to the channel name. * Fixes received
+ debug message sequence number. * Removed tx/rx debug message type
+ since it was hard coded to 0. * Made udptl.c logged message
+ header consistent if possible: "UDPTL (%s): ". * Removed unused
+ rx_expected_seq_no from struct ast_udptl. (closes issue
+ ASTERISK-18401) Reported by: Kevin P. Fleming Patches:
+ jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Matthew Nicholson ........
+
+2011-10-06 13:43 +0000 [r339586] Leif Madsen <lmadsen@digium.com>
+
+ * /, build_tools/prep_tarball: Merged revisions 339566 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r339566 | lmadsen | 2011-10-05 16:30:11 -0500 (Wed, 05
+ Oct 2011) | 8 lines Update prep_tarball script to download
+ pre-exported documentation. I've updated the prep_tarball script
+ to now download the pre-exported documentation from the Asterisk
+ wiki. This will give us more control over what is being included
+ in the tarball releases, and will make both the PDF and HTML
+ exported documentation look much better (especially when viewing
+ from a console). (Closes issue ASTERISK-18677) ........
+
+2011-10-05 17:01 +0000 [r339508-339512] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 339511 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339511 | rmudgett | 2011-10-05 12:01:01 -0500 (Wed, 05 Oct 2011)
+ | 1 line Fix Dial F option notes formatting. ........
+
+ * main/manager.c, /: Merged revisions 339504,339506 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r339504 | rmudgett | 2011-10-05 11:26:45 -0500 (Wed, 05
+ Oct 2011) | 7 lines Add missing documentation of required AMI
+ action Challenge AuthType header. (closes issue ASTERISK-18554)
+ Reported by: Vlad Povorozniuc Patches:
+ __20110919-manager-challenge-docs.patch.txt (license #4999) patch
+ uploaded by Leif Madsen ........ r339506 | rmudgett | 2011-10-05
+ 11:32:03 -0500 (Wed, 05 Oct 2011) | 1 line Fix XML error in AMI
+ action Challenge. ........
+
+2011-10-05 16:32 +0000 [r339507] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 339505 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339505 | mnicholson | 2011-10-05 11:31:21 -0500 (Wed, 05 Oct
+ 2011) | 3 lines The app name in the documentation must match what
+ we register the application as. ........
+
+2011-10-05 06:28 +0000 [r339463] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * res/res_fax.c: Only change the capabilities on the gateway when
+ the session is been destroyed there is still a race condition
+ that ends in a segfault. if the caps are changed the logic in
+ res_fax_spandsp will run T30 code not gateway code to end the
+ session. this has been experienced on a "slower" under spec
+ system.
+
+2011-10-04 22:56 +0000 [r339407] Richard Mudgett <rmudgett@digium.com>
+
+ * Makefile, /: Merged revisions 339406 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339406 | rmudgett | 2011-10-04 17:54:15 -0500 (Tue, 04 Oct 2011)
+ | 8 lines Make always create the MOH directory
+ (/var/lib/asterisk/moh). (closes issue ASTERISK-18409) Reported
+ by: abelbeck Patches: asterisk-1.8-makefile-moh.patch (license
+ #5903) patch uploaded by abelbeck Tested by: abelbeck, Michael
+ Keuter ........
+
+2011-10-04 19:44 +0000 [r339298-339353] Jonathan Rose <jrose@digium.com>
+
+ * /, main/say.c: Merged revisions 339352 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339352 | jrose | 2011-10-04 14:33:12 -0500 (Tue, 04 Oct 2011) |
+ 12 lines Removes improper use of sound 'and' in German language
+ mode from application saynumber Asterisk would say 'Five hundert
+ und sechs und zwanzig' instead of 'Five hundert sechs und
+ zwanzig'... which is both weird sounding and wrong. This patch
+ makes sure Asterisk will only say the 'and' word between the
+ single digit and double digit places. (closes issue
+ ASTERISK-18212) Reported By: Lionel Elie Mamane Patches:
+ upstream_germand_no_and.diff (License #5402) uploaded by Lionel
+ Elie Mamane ........
+
+ * /, res/res_jabber.c: Merged revisions 339297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339297 | jrose | 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) |
+ 13 lines Reverting revision 333265 due to component connection
+ problems it introduces. I'm going to attempt some generic
+ res_jabber cleanup and come up with a new fix for this problem,
+ but first it seems prudent to remove this rather broad attempt to
+ fix it and instead approach this problem either from the same
+ angle but looking only at canceling (or possibly rescheduling)
+ the send when we absolutely know it will cause a segfault or, if
+ that can't be easily accomplished, strictly from the devstate
+ side of things. Also, I'm pretty sure a lot of the code in
+ res_jabber isn't thread safe. (issue ASTERISK-18626) (issue
+ ASTERISK-18078) ........
+
+2011-10-04 11:49 +0000 [r339245] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/memheap.c, /: Merged revisions 339244 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339244 | may | 2011-10-04 15:44:55 +0400 (Tue, 04 Oct 2011) | 2
+ lines fix forget declaration in previous change ........
+
+2011-10-03 20:13 +0000 [r339145-339148] Leif Madsen <lmadsen@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 339147 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339147 | lmadsen | 2011-10-03 15:12:43 -0500 (Mon, 03 Oct 2011)
+ | 6 lines Remove duplicated Maxforwards line in AMI output.
+ (Closes issue ASTERISK-18637) Reported by: Jacek Konieczny
+ Patches: asterisk-sipshowpeer.patch (License #6298) uploaded by
+ Jacek Konieczny ........
+
+ * apps/app_dial.c, /: Merged revisions 339144 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339144 | lmadsen | 2011-10-03 14:54:52 -0500 (Mon, 03 Oct 2011)
+ | 6 lines Make documentation for Dial() options 'F' and 'F()'
+ more clear. (Closes issue ASTERISK-18646) Reported by: Physis
+ Heckman Tested by: Richard Mudgett ........
+
+2011-10-03 18:52 +0000 [r339089] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/memheap.c, /: Merged revisions 339087 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339087 | may | 2011-10-03 22:42:49 +0400 (Mon, 03 Oct 2011) | 4
+ lines destroy memheap mutex properly before memheap deleted (fix
+ memory leak occured after r304950 changes with DEBUG_THREAD
+ compile option) ........
+
+2011-10-03 18:44 +0000 [r339088] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c, main/file.c: Merged revisions 339086 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339086 | twilson | 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011)
+ | 10 lines Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more
+ places After the change in r336294, the new
+ AST_CONTROL_UPDATE_RTP_PEER frame is sent when a re-invite
+ happens. If we receive a re-invite from a device the
+ waitstream_core was not aware of the new control frame and would
+ drop the call. (closes issue ASTERISK-18610) Reported by:
+ Kristijan_Vrban ........
+
+2011-10-03 15:54 +0000 [r339011-339045] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Ported ast_fax_caps_to_str() to 10, not sure why
+ it wasn't already here. This function prints a list of caps
+ instead of a hex bitfield.
+
+ * res/res_fax.c: Don't clear the AST_FAX_TECH_MULTI_DOC flag right
+ after we set it.
+
+ * res/res_fax.c: properly remove the AST_FAX_TECH_GATEWAY flag
+ (instead of setting all of the other flags)
+
+2011-10-03 14:38 +0000 [r338904-338997] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * CHANGES: Documentation noting the extension of CHANNEL() for
+ chan_ooh323
+
+ * addons/chan_ooh323.c, funcs/func_channel.c: Remove the channel
+ function OOH323() and place its options into CHANNEL() channel
+ drivers should not have there own dialplan functions.
+
+ * res/res_fax.c: Fixup a race condition in res_fax.c where
+ FAXOPT(gateway)=no will turn off the gateway but the framehook is
+ not destroyed. this problem happens when a gateway is attempted
+ in the dialplan and the device is not available i may want to do
+ fax to mail in the server it will not be allowed. instead of
+ checking only AST_FAX_TECH_GATEWAY also check gateway_id Reverts
+ 338904 Fix some white space.
+
+ * res/res_fax.c: Remove T38 Gateway capability when detaching
+ framehook. SET(FAXOPT(gateway)=no) does not remove the capability
+ when detaching the framehook. small patch to fix this problem.
+
+2011-09-30 22:06 +0000 [r338801] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 338800 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r338800 | rmudgett | 2011-09-30 17:05:10 -0500 (Fri, 30
+ Sep 2011) | 12 lines Fix segfault in analog_ss_thread() not
+ checking ast_read() for NULL. NOTE: The problem was reported
+ against v1.6.2. It is unlikely to ever happen on v1.8 and above
+ since chan_dahdi.c:analog_ss_thread() is unlikely to be used. The
+ version in sig_analog.c has largely replaced it. (closes issue
+ ASTERISK-18648) Reported by: Stephan Bosch Patches:
+ jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Stephan Bosch ........
+
+2011-09-30 18:55 +0000 [r338719] Jonathan Rose <jrose@digium.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 338718 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338718 | jrose | 2011-09-30 13:54:30 -0500 (Fri, 30 Sep 2011) |
+ 1 line Adds documentation for QueueMemberStatus event generation
+ ........
+
+2011-09-30 16:35 +0000 [r338664] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Fix formatting of AMI header for SIP show
+ peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes
+ issue ASTERISK-18649) Reported by: Jacek Konieczny Patches:
+ asterisk-sipshowpeer_response_end.patch (license #6298) patch
+ uploaded by Jacek Konieczny ........ Merged revisions 338663 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-09-29 21:14 +0000 [r338556] Paul Belanger <pabelanger@digium.com>
+
+ * tests/test_amihooks.c, tests/test_security_events.c, /,
+ tests/test_locale.c, tests/test_logger.c,
+ tests/test_dlinklists.c, tests/test_linkedlists.c: Merged
+ revisions 338555 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338555 | pabelanger | 2011-09-29 17:12:21 -0400 (Thu, 29 Sep
+ 2011) | 2 lines Test modules should depend on the TEST_FRAMEWORK
+ flag ........
+
+2011-09-29 20:54 +0000 [r338552] Jason Parker <jparker@digium.com>
+
+ * /, tests/test_db.c, tests/test_netsock2.c: Merged revisions
+ 338551 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338551 | qwell | 2011-09-29 15:54:13 -0500 (Thu, 29 Sep 2011) |
+ 1 line Test modules have a support level of core. ........
+
+2011-09-29 18:32 +0000 [r338493] Leif Madsen <lmadsen@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 338492 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338492 | lmadsen | 2011-09-29 13:31:33 -0500 (Thu, 29 Sep 2011)
+ | 6 lines Update documentation for SIP_HEADER. The SIP_HEADER
+ function only works on the the initial SIP INVITE. The
+ documentation was updated in trunk, but not in 1.8 or 10, so I'm
+ making them match. (Closes issue ASTERISK-18640) ........
+
+2011-09-29 12:16 +0000 [r338417] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
+ revisions 338416 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) |
+ 12 lines The rtptimeout setting is ignored on a per peer basis.
+ Not only is the rtptimeout ignored in some cases but rtpkeepalive
+ and rtpholdtimeout is affected. this commit also removes
+ rtptimeout/rtpholdtimeout on text rtp. (closes issue
+ ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452
+ ........
+
+2011-09-28 22:36 +0000 [r338253-338323] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 338322 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338322 | rmudgett | 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011)
+ | 5 lines Make duplicate call ptr warning message more helpful. *
+ Adds the value of the call ptr to the duplicate call ptr message
+ to help trace why there is a duplicate call ptr. ........
+
+ * include/asterisk/logger.h, /: Merged revisions 338235 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338235 | rmudgett | 2011-09-28 16:17:45 -0500 (Wed, 28 Sep 2011)
+ | 7 lines Fix inconsistency in LOG_VERBOSE/AST_LOG_VERBOSE
+ declaration. (closes issue ASTERISK-17973) Reported by: Luke H
+ Patches: logger_h.patch (license #6278) patch uploaded by Luke H
+ ........
+
+2011-09-28 20:54 +0000 [r338228] Jason Parker <jparker@digium.com>
+
+ * build_tools/cflags.xml, channels/chan_usbradio.c,
+ build_tools/cflags-devmode.xml, agi/agi.xml, utils/utils.xml, /,
+ build_tools/embed_modules.xml, tests/test_db.c,
+ tests/test_netsock2.c: Merged revisions 338227 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338227 | qwell | 2011-09-28 15:52:47 -0500 (Wed, 28 Sep 2011) |
+ 1 line Add support levels to non-module sections of menuselect
+ (cflags, utils, etc). ........
+
+2011-09-28 20:26 +0000 [r338225] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 338224 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r338224 | rmudgett | 2011-09-28 15:24:41 -0500 (Wed, 28
+ Sep 2011) | 5 lines Fix chan_dahd compiling with gcc 4.6 when PRI
+ and SS7 not present. (closes issue ASTERISK-18357) Reported by:
+ Matthew Nicholson ........
+
+2011-09-27 20:13 +0000 [r338085] Paul Belanger <pabelanger@digium.com>
+
+ * /, apps/app_macro.c: Merged revisions 338084 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338084 | pabelanger | 2011-09-27 16:10:13 -0400 (Tue, 27 Sep
+ 2011) | 2 lines Upgrade app_macro to core ........
+
+2011-09-26 19:35 +0000 [r337974] Richard Mudgett <rmudgett@digium.com>
+
+ * cdr/cdr_manager.c, cdr/cdr_custom.c, apps/app_voicemail.c,
+ apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, /,
+ include/asterisk/cel.h, cdr/cdr_syslog.c, tests/test_gosub.c,
+ include/asterisk/channel.h, main/cel.c, main/manager.c,
+ funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c,
+ main/logger.c, cel/cel_sqlite3_custom.c: Merged revisions 337973
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011)
+ | 30 lines Fix deadlock when using dummy channels. Dummy channels
+ created by ast_dummy_channel_alloc() should be destoyed by
+ ast_channel_unref(). Using ast_channel_release() needlessly grabs
+ the channel container lock and can cause a deadlock as a result.
+ * Analyzed use of ast_dummy_channel_alloc() and made use
+ ast_channel_unref() when done with the dummy channel. (Primary
+ reason for the reported deadlock.) * Made
+ app_dial.c:dial_exec_full() not call ast_call() holding any
+ channel locks. Chan_local could not perform deadlock avoidance
+ correctly. (Potential deadlock exposed by this issue. Secondary
+ reason for the reported deadlock since the held lock was part of
+ the deadlock chain.) * Fixed some uses of
+ ast_dummy_channel_alloc() not checking the returned channel
+ pointer for failure. * Fixed some potential chan=NULL pointer
+ usage in func_odbc.c. Protected by testing the bogus_chan value.
+ * Fixed needlessly clearing a 1024 char auto array when setting
+ the first char to zero is enough in manager.c:action_getvar().
+ (closes issue ASTERISK-18613) Reported by: Thomas Arimont
+ Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Tested by: Thomas Arimont ........
+
+2011-09-27 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 10.0.0-beta2 Released.
+
+ * Based on revision that passed automated testing
+ (http://bamboo.asterisk.org/browse/AST10-LUCID-178)
+
+2011-09-26 19:35 +0000 [r337974] Richard Mudgett <rmudgett@digium.com>
+
+ * cdr/cdr_manager.c, cdr/cdr_custom.c, apps/app_voicemail.c,
+ apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, /,
+ include/asterisk/cel.h, cdr/cdr_syslog.c, tests/test_gosub.c,
+ include/asterisk/channel.h, main/cel.c, main/manager.c,
+ funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c,
+ main/logger.c, cel/cel_sqlite3_custom.c: Merged revisions 337973
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011)
+ | 30 lines Fix deadlock when using dummy channels. Dummy channels
+ created by ast_dummy_channel_alloc() should be destoyed by
+ ast_channel_unref(). Using ast_channel_release() needlessly grabs
+ the channel container lock and can cause a deadlock as a result.
+ * Analyzed use of ast_dummy_channel_alloc() and made use
+ ast_channel_unref() when done with the dummy channel. (Primary
+ reason for the reported deadlock.) * Made
+ app_dial.c:dial_exec_full() not call ast_call() holding any
+ channel locks. Chan_local could not perform deadlock avoidance
+ correctly. (Potential deadlock exposed by this issue. Secondary
+ reason for the reported deadlock since the held lock was part of
+ the deadlock chain.) * Fixed some uses of
+ ast_dummy_channel_alloc() not checking the returned channel
+ pointer for failure. * Fixed some potential chan=NULL pointer
+ usage in func_odbc.c. Protected by testing the bogus_chan value.
+ * Fixed needlessly clearing a 1024 char auto array when setting
+ the first char to zero is enough in manager.c:action_getvar().
+ (closes issue ASTERISK-18613) Reported by: Thomas Arimont
+ Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Tested by: Thomas Arimont ........
+
+2011-09-23 19:18 +0000 [r337840-337902] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, contrib/init.d/rc.archlinux.asterisk: Merged revisions 337898
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337898 | irroot | 2011-09-23 21:14:30 +0200 (Fri, 23 Sep 2011) |
+ 4 lines Spelling fix ........
+
+ * /, apps/app_queue.c: Merged revisions 337839 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337839 | irroot | 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) |
+ 11 lines Make sure a CDR is on the stack for call in the Queue.
+ Only let update_cdr act on the last CDR in the stack. In some
+ circumstances [Attended transfer to queue] a CDR record is not
+ inserted for this call where it should. (closes issue
+ ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266
+ ........
+
+2011-09-23 00:45 +0000 [r337775] Russell Bryant <russell@digium.com>
+
+ * configs/res_pktccops.conf.sample, /: Merged revisions 337774 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337774 | russell | 2011-09-22 19:44:19 -0500 (Thu, 22 Sep 2011)
+ | 11 lines Comment out entries in sample res_pktccops.conf. With
+ these options enabled, they can cause Asterisk to freak out by
+ SYN flooding a network and eating the CPU. Obviously it would be
+ good to fix the code so that this can't happen, but we can at
+ least change the default configuration so it doesn't happen. This
+ was reported downstream to the Fedora issue tracker:
+ https://bugzilla.redhat.com/show_bug.cgi?id=658431 ........
+
+2011-09-22 21:37 +0000 [r337721] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 337720 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011)
+ | 18 lines Made ISDN not add numbering plan prefix strings to
+ empty numbers. When the Caller-ID is restricted, the expected
+ behavior is for the Caller-ID to be blank. In chan_dahdi, the
+ national prefix is placed onto the Caller-ID number even if it is
+ restricted (empty) causing the Caller-ID to be the national
+ prefix rather than blank. This behavior was lost when sig_pri was
+ extracted from chan_dahdi. * Made not add prefix strings to empty
+ connected line, calling, and ANI number strings. (closes issue
+ ASTERISK-18577) Reported by: Kris Shaw Patches:
+ jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Kris Shaw ........
+
+2011-09-22 18:43 +0000 [r337640] Paul Belanger <pabelanger@digium.com>
+
+ * CREDITS, apps/app_meetme.c, CHANGES: Revert previous commit New
+ feature should be added into trunk, unfortunately it is too late
+ for the Asterisk 10 branch.
+
+2011-09-22 15:47 +0000 [r337595-337597] Jonathan Rose <jrose@digium.com>
+
+ * channels/sip/security_events.c (added),
+ channels/sip/include/security_events.h (added): Forgot to svn add
+ new files to r337595 Part of Generating security events for
+ chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young
+ Patches: security_events_chan_sip_v4.patch (License #5026) by
+ Michael L. Young Reviewboard:
+ https://reviewboard.asterisk.org/r/1362/
+
+ * configs/logger.conf.sample, channels/chan_sip.c,
+ include/asterisk/event_defs.h, main/security_events.c,
+ main/event.c, CHANGES, channels/sip/include/sip.h,
+ include/asterisk/security_events_defs.h: Generate Security events
+ in chan_sip using new Security Events Framework Security Events
+ Framework was added in 1.8 and support was added for AMI to
+ generate events at that time. This patch adds support for
+ chan_sip to generate security events. (closes issue
+ ASTERISK-18264) Reported by: Michael L. Young Patches:
+ security_events_chan_sip_v4.patch (license #5026) by Michael L.
+ Young Review: https://reviewboard.asterisk.org/r/1362/
+
+2011-09-22 11:44 +0000 [r337431-337542] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * res/res_srtp.c, /: Merged revisions 337541 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) |
+ 8 lines Add warned to ast_srtp to prevent errors on each frame
+ from libsrtp The first 9 frames are not reported as some devices
+ dont use srtp from first frame these are suppresed. the warning
+ is then output only once every 100 frames. ........
+
+ * /, channels/chan_h323.c: Merged revisions 337486 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22
+ Sep 2011) | 10 lines If IP address is used in chan_h323 host
+ parameter of peer configuration. module tries to resolve IP
+ address to IP address and fails. Simple fix to set family of
+ socket this is a hangover from ipv6 changes. (closes issue
+ ASTERISK-18237) (issue ASTERISK-17278) (issue ASTERISK-17500)
+ ........
+
+ * apps/app_originate.c, CHANGES: Revert commit r337261 This commit
+ is for trunk not version 10 ----- Adds a timeout argument to
+ app_originate the default is 30s this will be used if the timout
+ supplied is invalid or no timeout is supplied. -----
+
+ * main/channel.c, /: Merged revisions 337430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) |
+ 19 lines Its possible to loose audio on ast_write when the
+ channel is not transcoded correctly. in the case of DAHDI the
+ channel is hungup. This patch tries to "fix" the problem and make
+ the channel compatiable and warn the user of this problem. Please
+ note there is a underlying problem with codec negotion this does
+ not fix the problem it does try to rectify it and prevent loss of
+ service. Review: https://reviewboard.asterisk.org/r/1442/ (closes
+ issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue
+ ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325)
+ (issue ASTERISK-18422) ........
+
+2011-09-21 21:25 +0000 [r337342-337380] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * apps/app_voicemail.c, /: More silly spacing changes ..... Merged
+ revisions 337353 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_voicemail.c, /: ........ Dumb little spacing fix.
+ ........ Merged revisions 337344 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * funcs/func_curl.c, /: ........ Escape commas in keys and values,
+ when keys and values are enumerated by commas. Review:
+ https://reviewboard.asterisk.org/r/1433 ........ Merged revisions
+ 337325 from https://origsvn.digium.com/svn/asterisk/branches/1.8
+
+2011-09-21 11:15 +0000 [r337261-337263] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * configs/sip.conf.sample: Whitespace fixup from SRTP patch
+
+ * apps/app_originate.c, CHANGES: Adds a timeout argument to
+ app_originate the default is 30s this will be used if the timout
+ supplied is invalid or no timeout is supplied. Contributed by:
+ jacco (thank you for the work) Review:
+ https://reviewboard.asterisk.org/r/1310/
+
+2011-09-21 09:32 +0000 [r337178-337219] Olle Johansson <oej@edvina.net>
+
+ * configs/extensions.conf.sample, main/pbx.c, CHANGES: Make
+ ast_pbx_run() not default to s@default if extension is not found
+ Review: https://reviewboard.asterisk.org/r/1446/ This is a bug -
+ or architecture mistake - that has been in Asterisk for a very
+ long time. It was exposed by the AMI originate action and
+ possibly some other applications. Most channel drivers checks if
+ an extension exists BEFORE starting a pbx on an inbound call, so
+ most calls will not depend on this issue. Thanks everyone
+ involved in the review and on IRC and the mailing list for a
+ quick review and all the feedback. (closes issue ASTERISK-18578)
+
+ * res/res_rtp_asterisk.c, configs/rtp.conf.sample, CHANGES: Change
+ strictrtp option to default to yes in the RTP module Suggested by
+ Kapejod on Facebook Review:
+ https://reviewboard.asterisk.org/r/1448/ (closes issue
+ ASTERISK-18587) Thanks for quick feedback to kpfleming and
+ Tilghman --Denna och nedanstående rader kommer inte med i
+ loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M
+ res/res_rtp_asterisk.c
+
+2011-09-20 22:49 +0000 [r337120] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c, apps/app_dial.c, include/asterisk/app.h, /,
+ apps/app_meetme.c, apps/app_minivm.c, main/app.c,
+ apps/app_confbridge.c, apps/app_followme.c: Merged revisions
+ 337118 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011)
+ | 21 lines Fix for incorrect voicemail duration in external
+ notifications This patch fixes an issue where the voicemail
+ duration was being reported with a duration significantly less
+ than the actual sound file duration. Voicemails that contained
+ mostly silence were reporting the duration of only the sound in
+ the file, as opposed to the duration of the file with the
+ silence. This patch fixes this by having two durations reported
+ in the __ast_play_and_record family of functions - the
+ sound_duration and the actual duration of the file. The
+ sound_duration, which is optional, now reports the duration of
+ the sound in the file, while the actual full duration of the file
+ is reported in the duration parameter. This allows the voicemail
+ applications to use the sound_duration for minimum duration
+ checking, while reporting the full duration to external parties
+ if the voicemail is kept. (issue ASTERISK-2234) (closes issue
+ ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad
+ House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1443 ........
+
+2011-09-20 22:47 +0000 [r337119] Richard Mudgett <rmudgett@digium.com>
+
+ * funcs/func_strings.c: Fix crash with STRREPLACE function. The
+ ast_func_read() function calls the .read2 callback with the len
+ parameter set to zero indicating no size restrictions on the
+ supplied ast_str buffer. The value was used to dimension a local
+ starts[] array with the array subsequently used. * Reworked the
+ strreplace() function to perform the string replacement in a
+ straight forward manner. Eliminated the need for the starts[]
+ array. (closes issue ASTERISK-18545) Reported by: Federico Alves
+ Patches: jira_asterisk_18545_v10.patch (license #5621) patch
+ uploaded by rmudgett Tested by: rmudgett, Federico Alves
+
+2011-09-20 22:19 +0000 [r337116] Leif Madsen <lmadsen@digium.com>
+
+ * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 337115 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337115 | lmadsen | 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011)
+ | 7 lines Update RedHat Init script to work with Heartbeat. The
+ current RedHat init script was not LSB compatible. This change
+ will make it LSB compatible so that it can work correctly with
+ Heartbeat. (Closes issue ASTERISK-18253) Reported by: c0rnoTa
+ ........
+
+2011-09-20 21:05 +0000 [r337062] Kinsey Moore <kmoore@digium.com>
+
+ * tests/test_pbx.c, main/pbx.c, /: Merged revisions 337061 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337061 | kmoore | 2011-09-20 16:04:11 -0500 (Tue, 20 Sep 2011) |
+ 11 lines Make CANMATCH with the new pattern match engine behave
+ more like the old one When checking an extension for E_CANMATCH
+ using the new extension matching algorithm, an exact match was
+ not returned as a possible match resulting in the queue failing
+ to allow a caller to exit on DTMF. This removes the requirement
+ that an extension be longer than acquired digits for an
+ E_CANMATCH operation to succeed. (closes issue ASTERISK-18044)
+ Review: https://reviewboard.asterisk.org/r/1367/ ........
+
+2011-09-20 19:12 +0000 [r336978-337008] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_ss7.c: Merged revisions 337007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011)
+ | 15 lines Check if a channel was created before using the
+ pointer in sig_ss7_new_ast_channel(). Fixes the crash in
+ ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing
+ libss7 access lock protection. * Prevent cancelling the
+ ss7_linkset() thread at inoportune times just like the
+ pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M
+ Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621)
+ patch uploaded by rmudgett (attached to related ASTERISK-17966)
+ ........
+
+ * /, channels/sig_ss7.c: Merged revisions 336977 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011)
+ | 21 lines Fix deadlock from not releasing SS7 linkset lock.
+ sig_ss7_hangup() failed to release the SS7 linkset lock if the
+ call had the alreadyhungup flag set. * Made unlock the SS7
+ linkset lock in sig_ss7_hangup() if the alreadyhungup flag is
+ set. * Made ss7_start_call() not hold any locks while creating
+ the channel for an incoming call to prevent deadlock. * Made
+ ss7_grab() a void function, since it could never fail, to
+ simplify calling code. * Made obtain the channel lock to do
+ softhangup in some places. Patches: jira_ast_668_v1.8.patch
+ (license #5621) patch uploaded by rmudgett JIRA AST-668 ........
+
+2011-09-20 16:51 +0000 [r336936] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * channels/sip/sdp_crypto.c, channels/chan_sip.c,
+ channels/sip/include/sdp_crypto.h, channels/sip/include/srtp.h,
+ configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h:
+ Allow Setting Auth Tag Bit length Based on invite or config
+ option Update the SIP SRTP API to allow use of 32 or 80 bit
+ taglen. Curently only 80 bit is supported. The outgoing invite
+ will use the taglen of the incoming invite preventing one-way
+ audio. (Closes issue ASTERISK-17895) Review:
+ https://reviewboard.asterisk.org/r/1173/
+
+2011-09-20 01:03 +0000 [r336878] Russell Bryant <russell@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Merged revisions 336877 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19
+ Sep 2011) | 36 lines Fix crashes in ast_rtcp_write(). This patch
+ addresses crashes related to RTCP handling. The backtraces just
+ show a crash in ast_rtcp_write() where it appears that the RTP
+ instance is no longer valid. There is a race condition with
+ scheduled RTCP transmissions and the destruction of the RTP
+ instance. This patch utilizes the fact that ast_rtp_instance is a
+ reference counted object and ensures that it will not get
+ destroyed while a reference is still around due to scheduled RTCP
+ transmissions. RTCP transmissions are scheduled and executed from
+ the chan_sip scheduler context. This scheduler context is
+ processed in the SIP monitor thread. The destruction of an RTP
+ instance occurs when the associated sip_pvt gets destroyed (which
+ happens when the sip_pvt reference count reaches 0). However, the
+ SIP monitor thread is not the only thread that can cause a
+ sip_pvt to get destroyed. The sip_hangup function, executed from
+ a channel thread, also decrements the reference count on a
+ sip_pvt and could cause it to get destroyed. While this is being
+ changed anyway, the patch also removes calling ast_sched_del()
+ from within the RTCP scheduler callback. It's not helpful. Simply
+ returning 0 prevents the callback from being rescheduled. (closes
+ issue ASTERISK-18570) Related issues that look like they are the
+ same problem: (issue ASTERISK-17560) (issue ASTERISK-15406)
+ (issue ASTERISK-15257) (issue ASTERISK-13334) (issue
+ ASTERISK-9977) (issue ASTERISK-9716) Review:
+ https://reviewboard.asterisk.org/r/1444/ ........
+
+2011-09-19 22:13 +0000 [r336792] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 336791 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011)
+ | 2 lines Don't interfere with T.38 reinvites This is an update
+ to the fix for ASTERISK-18340 and ASTERISK-17725 ........
+
+2011-09-19 21:41 +0000 [r336734-336789] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * funcs/func_strings.c: Ensure substring will not be found in the
+ previous match.
+
+ * include/asterisk/optional_api.h, Makefile, /, configure,
+ include/asterisk/autoconfig.h.in, main/Makefile,
+ codecs/gsm/Makefile, configure.ac, Makefile.rules: Merged
+ revisions 336733 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011)
+ | 11 lines Various changes to allow 1.8 to compile on Mac OS X
+ Lion (10.7) * Makefile workaround for 10.6 extended to work on
+ 10.7 and later. * Now uses the 'weak' symbol for Lion systems,
+ which no longer support 'weak_import' Closes ASTERISK-17612.
+ Closes ASTERISK-18213. Tested by: tilghman, oej. ........
+
+2011-09-19 20:16 +0000 [r336717] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_echo.c, apps/app_saycounted.c, apps/app_mp3.c,
+ apps/app_morsecode.c, res/res_musiconhold.c, apps/app_queue.c,
+ apps/app_mixmonitor.c: Merged revisions 336716 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) |
+ 7 lines Document applications that play audio and do not answer
+ unanswered calls. This patch is part of an effort to document
+ early media and its usage. If you are interested in contributing
+ to this documentation effort, there are probably other
+ applications worth documenting as well as an Asterisk wiki
+ article at
+ https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
+ ........
+
+2011-09-19 18:51 +0000 [r336659] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, /, UPGRADE-1.8.txt: Merged revisions 336658 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011)
+ | 31 lines Made Dial d and H options no longer immediately
+ auto-answer the calling leg. The Dial d and H options break DTMF
+ attended transfer atxferdropcall option. 1) Party A calls party
+ B. 2) Party B does a DTMF attended transfer to Party C. If the
+ dialplan uses the Dial d or H options to call Party C then the
+ Dial application answers the call immediately before initiating
+ the call leg to Party C. The premature answer causes the transfer
+ code to not invoke the atxferdropcall=no behavior for a blonde
+ transfer since Party C has "answered". The transfer code thinks
+ that Party B has "consulted" with Party C when Party B hangs up
+ and completes the transfer to Party A. Party A now hears ringback
+ until Party C actually answers. ASTERISK-13294 Dial d option.
+ ASTERISK-11067 Dial H option to disconnect before answer. The
+ referenced issues made Dial answer with the d and H options
+ because many SIP and ISDN phones cannot send DTMF before the call
+ is connected. * Made require the dialplan to control when or if
+ the call needs to be answered to use the Dial application d and H
+ options. (The call is no longer surprise answered when using the
+ Dial d or H options.) Review:
+ https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA
+ AST-666 ........
+
+2011-09-19 15:42 +0000 [r336573] Leif Madsen <lmadsen@digium.com>
+
+ * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 336572
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011)
+ | 7 lines Update get_ilbc_source.sh script to work again.
+ Recently iLBC support in Asterisk has changed after the
+ acquisition of GIPS by Google. More information about how this
+ may affect you is available in a blog post at:
+ http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
+ ........
+
+2011-09-19 15:32 +0000 [r336570] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 336569 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011)
+ | 4 lines Rework sig_pri_hangup() to be simpler and clearer. JIRA
+ AST-675 ........
+
+2011-09-19 13:48 +0000 [r336502-336504] Olle Johansson <oej@edvina.net>
+
+ * Makefile: Revert accidental change
+
+ * Makefile, /, channels/chan_sip.c: Merged revisions 336501 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5
+ lines Add diversion header to a 302 redirect response if we have
+ diversion data (closes issue ASTERISK-18143) patch by oej
+ ........
+
+2011-09-19 13:31 +0000 [r336500] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, channels/chan_h323.c: Merged revisions 336499 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19
+ Sep 2011) | 13 lines A long time ago in a galaxy far far away a
+ IPv6 update was made, chan_h323 was not updated causeing all to
+ flee to chan_ooh323. the brave Jedi [asterisk developers]
+ pondered this miscarrige of justice and restored order to the
+ force for the sake of closing out 2 old issues. (closes issue
+ ASTERISK-17278) (closes issue ASTERISK-17500) Reported by: dread,
+ sybasesql Tested by: irroot Reviewed by: IRC (russellb,
+ kpfleming) ........
+
+2011-09-19 12:15 +0000 [r336381-336441] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c, /: Merged revisions 336440 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336440 | oej | 2011-09-19 14:06:48 +0200 (Mån, 19 Sep 2011) | 2
+ lines Make sure manager_debug option is reset at reload ........
+
+ * /, channels/chan_sip.c: Merged revisions 336378 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9
+ lines Add missing unlock at MWI message sending time (closes
+ issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041)
+ by Gregory Hinton Nietsky Thanks to irrot for the reminder, to
+ Gregory for the patch! ........
+
+2011-09-16 22:11 +0000 [r336313-336316] Terry Wilson <twilson@digium.com>
+
+ * /, funcs/func_frame_trace.c: Merged revisions 336314 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16
+ Sep 2011) | 2 lines Whitespace fix ........
+
+ * /, funcs/func_frame_trace.c: Merged revisions 336312 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16
+ Sep 2011) | 5 lines Add missing frame types to func_frame_trace
+ Also casts control frames to the proper enum so that the compile
+ will catch new additions. ........
+
+2011-09-16 21:09 +0000 [r336307] Jonathan Rose <jrose@digium.com>
+
+ * main/channel.c, main/rtp_engine.c, /, channels/chan_sip.c,
+ include/asterisk/frame.h: Merged revisions 336294 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep
+ 2011) | 13 lines Fix bad RTP media bridges in directmedia calls
+ on peers separated by multiple Asterisk nodes. In a situation
+ involving devices on separate Asterisk trunks, the remote RTP
+ bridge would break when starting a call with directmedia. This
+ patch queues a new type of control frame so that our RTP bridge
+ loop can properly detect when these situations occur and check to
+ see if peers need to be updated in order to send their media to
+ the proper location. (Closes issue ASTERISK-18340) Reported by:
+ Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk
+ Tested by: twilson, jrose ........
+
+2011-09-16 19:10 +0000 [r336235] Sean Bright <sean@malleable.com>
+
+ * /, UPGRADE-1.8.txt: Merged revisions 336234 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri, 16 Sep
+ 2011) | 2 lines Make a note that inotify won't work with an NFS
+ mounted spooler directory. ........
+
+2011-09-16 10:12 +0000 [r336094-336167] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * channels/chan_misdn.c, /: Merged revisions 336166 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16
+ Sep 2011) | 16 lines The round robin routing routine in
+ chan_misdn.c is broken. it rotates between ports but never checks
+ the channels in the ports. i have extensivly tested it and
+ verified it works on 1 upto 4 ports. before the patch only 1 out
+ of each port was used now all are used as expected. (closes issue
+ ASTERISK-18413) Reported by: irroot Tested by: irroot Reviewed
+ by: irroot Review: https://reviewboard.asterisk.org/r/1410/
+ ........
+
+ * /, apps/app_queue.c: Merged revisions 336093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) |
+ 20 lines Locking order in app_queue.c causes deadlocks. a channel
+ lock must never be held with the queues container lock held. the
+ deadlock occured on masquerade. the queues container lock is a
+ relic of the past the old queue module lock. with ao2 there is no
+ need to hold this lock when dealing with members this patch
+ removes unneeded locks. (closes issue ASTERISK-18101) (closes
+ issue ASTERISK-18487) Reported by: Paul Rolfe, Jason Legault
+ Tested by: irroot, Jason Legault, Paul Rolfe Reviewed by: Matthew
+ Nicholson Review: https://reviewboard.asterisk.org/r/1402/
+ ........
+
+2011-09-15 15:19 +0000 [r336091] David Vossel <dvossel@digium.com>
+
+ * main/format_cap.c: Removes some no-op code found in format_cap.c.
+
+2011-09-15 12:46 +0000 [r336042] Olle Johansson <oej@edvina.net>
+
+ * CREDITS, apps/app_meetme.c, CHANGES: Meetme: Introducing a new
+ option "k" to kill a conference if there's only a single member
+ left. When using Meetme as a modular call bridge from third party
+ applications, it's handy to make it behave like a normal call
+ bridge. When the second to last person exists, the last person
+ will be kicked out of the conference when this option is enabled.
+ (closes issue ASTERISK-18234) Review:
+ https://reviewboard.asterisk.org/r/1376/ Patch by oej, sponsored
+ by ClearIT, Solna, Sweden
+
+2011-09-15 08:29 +0000 [r335991] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, channels/chan_agent.c: Merged revisions 335978 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15
+ Sep 2011) | 11 lines lock the channel before calling
+ ast_bridged_channel() to prevent a seg fault. AMI agents list
+ called on shutdown causes a segfault, introducing proper locking
+ will prevent this. (closes issue ASTERISK-18092) Reported by:
+ agustina Patches: chan_agent.patch (License #5041) patch uploaded
+ by irroot ........
+
+2011-09-14 18:31 +0000 [r335852-335912] Richard Mudgett <rmudgett@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 335911 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011)
+ | 13 lines Remove unnecessary libpri dependency checks in the
+ configure script. Using the --with-pri option with the configure
+ script generated an error about not having PRI_L2_PERSISTENCE if
+ you did not have the absolute latest libpri SVN checkout
+ installed. The AST_EXT_LIB_SETUP_DEPENDENT macro in the
+ configure.ac script seems to be for libraries that are dependent
+ upon other libraries and not necessarily for optional/added
+ features within a library. (closes issue ASTERISK-18535) Reported
+ by: Michael Keuter ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 335851 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14
+ Sep 2011) | 11 lines Fixed cut-n-paste regression using the wrong
+ variable. Fixes the missing DAHDI channels when using the newer
+ chan_dahdi.conf sections for channel configuration. (closes issue
+ ASTERISK-18496) Reported by: Sean Darcy Patches:
+ jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Sean Darcy, rmudgett ........
+
+2011-09-14 13:28 +0000 [r335791] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /: Merged revisions 335790 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep
+ 2011) | 4 lines The tech and data members of
+ fast_originate_helper are not string fields. ASTERISK-17709
+ ........
+
+2011-09-13 22:10 +0000 [r335721] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 335720 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13 Sep 2011)
+ | 1 line Remove obsolete todo comment about PICKUPRESULT.
+ ........
+
+2011-09-13 21:37 +0000 [r335717] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * main/asterisk.c: do parse defaultlanguage from asterisk.conf Do
+ parse the option "defaultlanguage" from the [options] section of
+ asterisk.conf, as in the sample config file. Otherwise the
+ build-time default language (normally "en") is always the default
+ one. Review: https://reviewboard.asterisk.org/r/1342/
+ Signed-off-by: Tzafrir Cohen (License #5035)
+ <tzafrir.cohen@xorcom.com> Original-Commit:
+ http://svn.digium.com/svn/asterisk/branches/1.8@335716
+
+2011-09-13 18:55 +0000 [r335656] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, configure, configure.ac: Merged revisions 335655 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13
+ Sep 2011) | 4 lines Move mandatory checks closer to the beginning
+ of the file. If these are going to fail, they should fail as
+ quickly as possible. ........
+
+2011-09-13 18:47 +0000 [r335653] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, main/manager.c, /: Merged revisions 335618 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep
+ 2011) | 5 lines Don't limit the size of appdata for manager
+ originate actions. ASTERISK-17709 Patch by: tilghman (with
+ modifications) ........
+
+2011-09-13 07:24 +0000 [r335510] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/event.h, /, res/ais/evt.c, main/event.c: Merged
+ revisions 335497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011)
+ | 15 lines Fix a crash in res_ais. This patch resolves a crash
+ observed in a load testing environment that involved the use of
+ the res_ais module. I observed some crashes where the event
+ delivery callback would get called, but the length parameter
+ incidcating how much data there was to read was 0. The code
+ assumed (with good reason I would think) that if this callback
+ got called, there was an event available to read. However, if the
+ rare case that there's nothing there, catch it and return instead
+ of blowing up. More specifically, the change always ensure that
+ the size of the received event in the cluster is always big
+ enough to be a real ast_event. Review:
+ https://reviewboard.asterisk.org/r/1423/ ........
+
+2011-09-12 15:55 +0000 [r335434] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 335433 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep
+ 2011) | 6 lines Properly set caller_warning and callee_warning
+ before we try to use them. ASTERISK-18199 Patch by: elguero
+ Testing by: rtang ........
+
+2011-09-12 14:22 +0000 [r335346] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 335341 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) |
+ 10 lines Ensure frames are not written to dialed channel if
+ ringback is requested When a single channel was dialed and there
+ was media to be forwarded to the calling channel, the media was
+ written without regard for ringback causing silence to be heard
+ in some circumstances. This regression was introduced when the
+ meaning of "single" changed to mean only the number of channels
+ dialed. (closes issue ASTERISK-18083) ........
+
+2011-09-12 13:47 +0000 [r335323] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 335319 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12
+ lines Lock the peer->mvipvt to avoid crashes with SIP history
+ enabled After the launch of 1.6 event-based MWI we have two
+ threads handling the peer->mwipvt, which cause issues with SIP
+ history additions in combination with the max limit for number of
+ history entries. Review: https://reviewboard.asterisk.org/r/1373/
+ (closes issue ASTERISK-18288) Thanks to irrot for peer review.
+ Work with this bug funded by IPvision AS ........
+
+2011-09-12 13:27 +0000 [r335321] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 335320 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12
+ Sep 2011) | 9 lines Prevent IAX2 from getting IPv6 addresses via
+ DNS IAX2 does not support IPv6 and getting such addresses from
+ DNS can cause error messages on the remote end involving bad IPv4
+ address casts in the presence of IPv6/IPv4 tunnels. This patch
+ ensures that IAX2 will not encounter IPv6 addresses via DNS
+ queries. (closes issue ASTERISK-18090) ........
+
+2011-09-12 11:11 +0000 [r335260] Stefan Schmidt <sst@sil.at>
+
+ * /, channels/chan_sip.c: Merged revisions 335259 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011)
+ | 6 lines build_peer doesnt unlink a peer object from peers_by_ip
+ container which leads to a wrong refcounter value. adding an
+ ao2_unlink from the peers_by_ip container fix it. Review:
+ https://reviewboard.asterisk.org/r/1428/ ........
+
+2011-09-09 16:27 +0000 [r335078] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_mgcp.c, channels/chan_unistim.c, apps/app_dial.c,
+ main/pbx.c, addons/chan_ooh323.c, /, channels/chan_sip.c,
+ channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c,
+ main/channel.c, channels/chan_usbradio.c, main/dial.c,
+ channels/chan_dahdi.c, channels/chan_misdn.c,
+ channels/chan_skinny.c, funcs/func_frame_trace.c,
+ main/features.c, channels/chan_h323.c, channels/chan_alsa.c,
+ include/asterisk/frame.h, channels/sig_ss7.c: Merged revisions
+ 335064 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011)
+ | 23 lines Updated SIP 484 handling; added Incomplete control
+ frame When a SIP phone uses the dial application and receives a
+ 484 Address Incomplete response, if overlapped dialing is enabled
+ for SIP, then the 484 Address Incomplete is forwarded back to the
+ SIP phone and the HANGUPCAUSE channel variable is set to 28.
+ Previously, the Incomplete application dialplan logic was
+ automatically triggered; now, explicit dialplan usage of the
+ application is required. Additionally, this patch adds a new
+ AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel
+ driver receives this control frame, it is an indication that the
+ dialplan expects more digits back from the device. If the device
+ supports overlap dialing it should attempt to notify the device
+ that the dialplan is waiting for more digits; otherwise, it can
+ handle the frame in a manner appropriate to the channel driver.
+ (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested
+ by: Matthew Jordan Review:
+ https://reviewboard.asterisk.org/r/1416/ ........
+
+2011-09-09 07:23 +0000 [r335014] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * funcs/func_dialplan.c, apps/app_readexten.c, CHANGES: Move code
+ for VALID_EXTEN from app_readexten to func_dialplan Mark
+ VALID_EXTEN deprecated. Review:
+ https://reviewboard.asterisk.org/r/1396/
+
+2011-09-08 22:28 +0000 [r334954] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/logger.c: Merged revisions 334953 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011)
+ | 10 lines Fix crash with res_fax when MALLOC_DEBUG and "core
+ stop gracefully" are used. Asterisk crashes if MALLOC_DEBUG is
+ enabled when res_fax tries to unregister its logger level. * Make
+ ast_logger_unregister_level() use ast_free() instead of free().
+ When MALLOC_DEBUG is enabled, ast_free() does not degenerate into
+ a call to free(). Therefore, if you allocated memory with a form
+ of ast_malloc you must free it with ast_free. ........
+
+2011-09-07 19:37 +0000 [r334844] Paul Belanger <pabelanger@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 334843 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed,
+ 07 Sep 2011) | 4 lines Cleanup chan_iax2.c log messages Review:
+ https://code.asterisk.org/code/cru/CR-AST-11 ........
+
+2011-09-07 19:33 +0000 [r334841] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 334840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011)
+ | 10 lines Fix AMI action Park crash. * Made AMI action Park not
+ say anything to the parker channel (AMI header Channel2) since
+ the AMI action is a third party parking the call. (This is a
+ change in behavior that cannot be preserved without a lot of
+ effort.) * Made not play pbx-parkingfailed if the Park 's' option
+ is used. JIRA AST-660 ........
+
+2011-09-07 15:10 +0000 [r334682-334747] Stefan Schmidt <sst@sil.at>
+
+ * /, main/features.c: Merged revisions 334682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011)
+ | 3 lines Adding the Feature to sent a Reason Header in a SIP
+ Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before
+ doing a masquerade in the pickup function. ........
+
+ * main/features.c: another clean up
+
+ * main/features.c: Adding the Feature to sent a Reason Header in a
+ SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE
+ before doing a masquerade in the pickup function.
+
+2011-09-07 08:14 +0000 [r334617-334621] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, CHANGES, apps/app_queue.c: Merged revisions 334620 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07 Sep
+ 2011) | 2 lines peroid typo ........
+
+ * main/pbx.c, /: Merged revisions 334616 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334616 | alecdavis | 2011-09-07 19:33:39 +1200 (Wed, 07 Sep
+ 2011) | 10 lines Prevent segfault if call arrives before Asterisk
+ is fully booted. Prevent ast_pbx_start and ast_run_start from
+ starting a new thread unless asterisk is fully booted. alecdavis
+ (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1407/ ........
+
+2011-09-06 15:47 +0000 [r334514] Paul Belanger <pabelanger@digium.com>
+
+ * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: authdebug
+ is now disabled by default To enable this functionaility again
+ set authdebug = yes in iax.conf Review:
+ https://reviewboard.asterisk.org/r/1414/
+
+2011-09-06 13:58 +0000 [r334455] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * apps/app_voicemail.c, /: Merged revisions 334453 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06
+ Sep 2011) | 13 lines Make SQL query in app_voicemail.c portable
+ LIMIT is not portable. Regression from r312212 (closes issue
+ ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen
+ Review: https://reviewboard.asterisk.org/r/1415/ ........
+
+2011-09-02 21:08 +0000 [r334297-334357] Richard Mudgett <rmudgett@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 334355 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02
+ Sep 2011) | 19 lines MusicOnHold has extra unref which may lead
+ to memory corruption and crash. The problem happens when a call
+ is disconnected and you had started a MOH class that does not use
+ the files mode. If you define REF_DEBUG and recreate the problem,
+ it will announce itself with the following warning: Attempt to
+ unref mohclass 0xb70722e0 (default) when only 1 ref remained, and
+ class is still in a container! * Fixed moh_alloc() and
+ moh_release() functions not handling the state->class reference
+ consistently. (closes issue ASTERISK-18346) Reported by: Mark
+ Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621)
+ patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski
+ Review: https://reviewboard.asterisk.org/r/1404/ ........
+
+ * /, include/asterisk/config.h, main/config.c: Merged revisions
+ 334296 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334296 | rmudgett | 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011)
+ | 39 lines Fix potential memory allocation failure crashes in
+ config.c. * Added required checks to the returned memory
+ allocation pointers to prevent crashes. * Made
+ ast_include_rename() create a replacement ast_variable list node
+ if the new filename is longer than the available space. Fixes
+ potential crash and memory leak. * Factored out
+ ast_variable_move() from ast_variable_update() so
+ ast_include_rename() can also use it when creating a replacement
+ ast_variable list node. * Made the filename stuffed at the end of
+ the struct a minimum allocated size in ast_variable_new() in case
+ ast_include_rename() changes the stored filename. * Constify
+ struct char pointers pointing to strings stuffed at the end of
+ the struct for: ast_variable, cache_file_mtime, and
+ ast_config_map. * Factored out cfmtime_new() to remove inlined
+ code and allow some struct pointers to become const. * Removed
+ the list lock from struct cache_file_mtime that was never used. *
+ Added doxygen comments to several structure elements and better
+ documented what strings are stuffed at the struct end char array.
+ * Reworked ast_config_text_file_save() and set_fn() to handle
+ allocation failure of the include file scratch pad object
+ tracking blank lines. * Made ast_config_text_file_save() fn[]
+ declared with PATH_MAX to ensure it is long enough for any
+ filename with path. Also reduced the number of container fileset
+ buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review:
+ https://reviewboard.asterisk.org/r/1378/ ........
+
+2011-09-01 17:39 +0000 [r334235] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/pbx.c, /: Merged revisions 334234 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334234 | tilghman | 2011-09-01 12:38:33 -0500 (Thu, 01 Sep 2011)
+ | 2 lines Remove 1.6 compatibility documentation from 1.8, as it
+ no longer applies. ........
+
+2011-09-01 17:36 +0000 [r334233] Matthew Nicholson <mnicholson@digium.com>
+
+ * CHANGES: fixed a typo
+
+2011-09-01 17:30 +0000 [r334230] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * res/res_config_odbc.c, /: Merged revisions 334229 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r334229 | tilghman | 2011-09-01 12:28:09 -0500 (Thu, 01
+ Sep 2011) | 18 lines Create a local alias for
+ ast_odbc_clear_cache. As a function pointer, the reference has to
+ be resolved at load time irrespective of the RTLD_LAZY flag.
+ Creating a local alias solves this problem, because the structure
+ is initialized with that local function pointer, while the actual
+ function can remain lazily linked until runtime. The reason why
+ this is important is because we lazily load function references
+ during the module loading process, in order to obtain priority
+ values for each module, ensuring that modules are loaded in the
+ correct order. Previous to this change, when this module was
+ initially loaded, the module loader would emit a symbol
+ resolution error, because of the above requirement. Closes
+ ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by
+ Walter Doekes, patch by me) ........
+
+2011-08-31 18:53 +0000 [r334064-334157] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 334156 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334156 | mnicholson | 2011-08-31 13:50:33 -0500 (Wed, 31 Aug
+ 2011) | 4 lines Disable T.38 when we get a invite with image
+ media port set to 0 ASTERISK-17678 ........
+
+ * res/res_fax.c: only alter the gateway_timeout when attching the
+ gateway to a channel ASTERISK-18219
+
+2011-08-31 16:00 +0000 [r334010-334013] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 334012 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31
+ Aug 2011) | 23 lines No DAHDI channel available for conference,
+ user introduction disabled. The following error will consistently
+ occur when trying to dial into a MeetMe conference when the
+ server does not have DAHDI hardware installed: app_meetme.c: No
+ DAHDI channel available for conference, user introduction
+ disabled (is chan_dahdi loaded?) While chan_dahdi is loaded
+ correctly during compilation and install of Asterisk/Dahdi,
+ including associated modules, etc., a chan_dahdi.conf
+ configuration file in /etc/asterisk is not created by FreePBX if
+ hardware does not exist, causing MeetMe to be unable to open a
+ DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo
+ channel when there is no chan_dahdi.conf file to load. (closes
+ issue ASTERISK-17398) Reported by: Preston Edwards Patches:
+ jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett ........
+
+ * main/channel.c, /, channels/chan_agent.c: Merged revisions 334009
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011)
+ | 43 lines Call pickup race leaves orphaned channels or crashes.
+ Multiple users attempting to pickup a call that has been forked
+ to multiple extensions either crashes or fails a masquerade with
+ a "bad things may happen" message. This is the scenario that is
+ causing all the grief: 1) Pickup target is selected 2) target is
+ marked as being picked up in ast_do_pickup() 3) target is
+ unlocked by ast_do_pickup() 4) app dial or queue gets a chance to
+ hang up losing calls and calls ast_hangup() on target 5) SINCE A
+ MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with
+ ast_channel_masquerade(), ast_hangup() completes successfully and
+ the channel is no longer in the channels container. 6)
+ ast_do_pickup() then calls ast_channel_masquerade() to schedule
+ the masquerade on the dead channel. 7) ast_do_pickup() then calls
+ ast_do_masquerade() on the dead channel 8) bad things happen
+ while doing the masquerade and in the process ast_do_masquerade()
+ puts the dead channel back into the channels container 9) The
+ "orphaned" channel is visible in the channels list if a crash
+ does not happen. This patch does the following: * Made
+ ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up
+ channel and not release the channel lock until that has happened.
+ * Made __ast_channel_masquerade() not setup a masquerade if
+ either channel has AST_FLAG_ZOMBIE set. * Fix chan_agent misuse
+ of AST_FLAG_ZOMBIE since it would no longer work. (closes issue
+ ASTERISK-18222) Reported by: Alec Davis Tested by: rmudgett, Alec
+ Davis, irroot, Karsten Wemheuer (closes issue ASTERISK-18273)
+ Reported by: Karsten Wemheuer Tested by: rmudgett, Alec Davis,
+ irroot, Karsten Wemheuer Review:
+ https://reviewboard.asterisk.org/r/1400/ ........
+
+2011-08-31 15:19 +0000 [r334007] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 334006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) |
+ 7 lines Correct an AMI protocol violation with SIPshowpeer The
+ response of SIPshowpeer ends with "\r\n\r\n". Since other
+ commands are ended by using \r\n this confuses any interfacing
+ script. (closes issue ASTERISK-17486) ........
+
+2011-08-30 21:53 +0000 [r333961-333962] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c: security fix. really drop call if
+ signalling addr is not same as socket addr
+
+ * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c,
+ addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, /,
+ addons/ooh323c/src/ooCalls.h, addons/ooh323c/src/oochannels.c:
+ Merged revisions 333947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5
+ lines cleanups in ACF/ARJ GK replies processing fixed long (24
+ sec) pause if acf/arj proccessed before ast_cond_wait called to
+ wait this ........
+
+2011-08-30 14:01 +0000 [r333895] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Replaced FAXOPT(gwtimeout) with a second parameter
+ to FAXOPT(gateway). Patch by: irroot Review:
+ https://reviewboard.asterisk.org/r/1385/ ASTERISK-18219
+
+2011-08-29 21:41 +0000 [r333837] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 333836 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011)
+ | 15 lines Refresh peer address if DNS unavailable at peer
+ creation If Asterisk starts and no DNS is available, outbound
+ registrations will fail indefinitely. This patch copies the
+ address from the sip_registry struct, which will be updated, to
+ the peer->addr when necessary. If dnsmgr is enabled, the
+ registration fails without the patch because even though the
+ address on the registry is updated via dnsmgr, the address is
+ just copied on the first try. Since we use ast_sockaddr_copy,
+ dnsmgr can't update the address that is copied to the sip_pvt or
+ peers. Closes issue ASTERISK-18000 Review:
+ https://reviewboard.asterisk.org/r/1335/ ........
+
+2011-08-29 21:12 +0000 [r333786] Richard Mudgett <rmudgett@digium.com>
+
+ * /, include/asterisk/channel.h, addons/chan_mobile.c: Merged
+ revisions 333784-333785 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333784 | rmudgett | 2011-08-29 16:05:43 -0500 (Mon, 29 Aug 2011)
+ | 2 lines Fix deadlock potential of
+ chan_mobile.c:mbl_ast_hangup(). ........ r333785 | rmudgett |
+ 2011-08-29 16:06:16 -0500 (Mon, 29 Aug 2011) | 1 line Add some do
+ not hold locks notes to channel.h ........
+
+2011-08-29 18:22 +0000 [r333716] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax_spandsp.c: It is possible for the gateway to be
+ attached when the channel is still negotiating T.38. This change
+ handles that case. ASTERISK-18329
+
+2011-08-29 17:28 +0000 [r333681] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, CHANGES: Use realtime text when it is negotiated
+ This patch make use of wirte_text() realtime text instead of
+ send_text() if T.140 is in native formats. ASTERISK-17937 Review:
+ https://reviewboard.asterisk.org/r/1356/
+
+2011-08-29 17:12 +0000 [r333631] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 333630 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r333630 | mjordan | 2011-08-29 12:11:15 -0500 (Mon, 29
+ Aug 2011) | 1 line Fixed improperly formatted TestEvent AMI
+ message in app_voicemail ........
+
+2011-08-29 15:56 +0000 [r333570] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_jabber.c: Merged revisions 333569 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333569 | jrose | 2011-08-29 10:55:34 -0500 (Mon, 29 Aug 2011) |
+ 4 lines Accidental use of variable client->status instead of
+ client->state in from ASTERISK-18078 (issue ASTERISK-18078)
+ ........
+
+2011-08-28 09:55 +0000 [r333508] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_vpb.cc: chan_vpb: remove unused variables (gcc4.6)
+ GCC 4.6 detects variables that get assined to, but never used
+ later. Also removes some remmed-out lines that become invalid.
+ (closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen
+ (License #5035) <tzafrir.cohen@xorcom.com>,
+
+2011-08-26 16:28 +0000 [r333410] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_jabber.c: Merged revisions 333378 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333378 | jrose | 2011-08-26 11:19:07 -0500 (Fri, 26 Aug 2011) |
+ 13 lines [patch] Buddies are always auto-registered when
+ processing the roster Reporter said autoregister flag was ignored
+ for registering 'buddies' which had a subscription to us.
+ Verified that this was the case and observed how the patch
+ addressed this and made sure it didn't break anything. (closes
+ issue ASTERISK-14233) Reported by: Simon Arlott Patches:
+ asterisk-0015229.patch (license #5756) patch uploaded by Simon
+ Arlott Tested by: Jonathan Rose ........
+
+2011-08-26 15:58 +0000 [r333370] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 333339 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r333339 | mjordan | 2011-08-26 08:36:36 -0500 (Fri, 26
+ Aug 2011) | 20 lines Bug fixes for voicemail user emailsubject /
+ emailbody. This code change fixes a few issues with the voicemail
+ user override of emailbody and emailsubject, including escaping
+ the strings, potential memory leaks, and not overriding the
+ voicemail defaults. Revision 325877 fixed this for
+ ASTERISK-16795, but did not fix it for ASTERISK-16781. A
+ subsequent check-in prevented 325877 from being applied to 10.
+ This check-in resolves both issues, and applies the changes to
+ 1.8, 10, and trunk. (closes issue ASTERISK-16781) Reported by:
+ Sebastien Couture Tested by: mjordan (closes issue
+ ASTERISK-16795) Reported by: mdeneen Tested by: mjordan Review:
+ https://reviewboard.asterisk.org/r/1374 ........
+
+2011-08-25 19:01 +0000 [r333268] Jason Parker <jparker@digium.com>
+
+ * Makefile, /: Merged revisions 333267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333267 | qwell | 2011-08-25 14:00:55 -0500 (Thu, 25 Aug 2011) |
+ 2 lines Fix for DESTDIR spaces patch. ........
+
+2011-08-25 19:00 +0000 [r333266] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_jabber.c: Merged revisions 333265 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333265 | jrose | 2011-08-25 13:47:42 -0500 (Thu, 25 Aug 2011) |
+ 14 lines Segfault when publishing device states via XMPP and not
+ connected When using publishing device state with res_jabber,
+ Asterisk will attempt to send a device state using the
+ unconnected client using iks_send_raw and crash. This patch
+ checks the validity of the connection before attempting to send
+ the device state. (closes issue ASTERISK-18078) Reported by:
+ Michael L. Young Patches:
+ res_jabber-segfault-pubsub-not-connected2.patch (license #5026)
+ patch uploaded by Michael L. Young Tested by: Jonathan Rose
+ ........
+
+2011-08-25 15:29 +0000 [r333203] Jason Parker <jparker@digium.com>
+
+ * Makefile, build_tools/mkpkgconfig, /, configure, configure.ac,
+ makeopts.in, sounds/Makefile: Merged revisions 333201 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) |
+ 8 lines Fix installation into directories containing spaces. This
+ also vastly simplifies the logic in sounds/Makefile (Closes issue
+ ASTERISK-18290) Reported by: Paul Belanger Review:
+ https://reviewboard.asterisk.org/r/1379/ ........
+
+2011-08-24 16:51 +0000 [r333115] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Changed the "timeout" option to "gwtimeout".
+ ASTERISK-18219
+
+2011-08-23 18:15 +0000 [r332878-333011] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 333010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333010 | rmudgett | 2011-08-23 13:14:01 -0500 (Tue, 23 Aug 2011)
+ | 12 lines Memory Leak in app_queue The patch that was committed
+ in the 1.6.x versions of Asterisk for ASTERISK-15862 actually
+ fixed two issues. One was not applicable to 1.8 but the other is.
+ queue_leak.patch fixes the portion applicable to 1.8. (closes
+ issue ASTERISK-18265) Reported by: Fred Schroeder Patches:
+ queue_leak.patch (license #5049) patch uploaded by mmichelson
+ Tested by: Thomas Arimont ........
+
+ * /, main/config.c: Merged revisions 332939 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332939 | rmudgett | 2011-08-22 16:22:24 -0500 (Mon, 22 Aug 2011)
+ | 7 lines Minor code optimizations. * Simplify
+ ast_category_browse() logic for easier understanding. * Remove
+ dead code in ast_variable_delete() and simplify some of its
+ logic. ........
+
+ * /, apps/app_queue.c: Merged revisions 332874 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332874 | rmudgett | 2011-08-22 14:32:19 -0500 (Mon, 22 Aug 2011)
+ | 18 lines Reference leaks in app_queue. * Fixed
+ load_realtime_queue() leaking a queue reference when it
+ overwrites q when processing a realtime queue. (issue
+ ASTERISK-18265) * Make join_queue() unreference the queue
+ returned by load_realtime_queue() when it is done with the
+ pointer. The load_realtime_queue() returns a reference to the
+ just loaded realtime queue. * Fixed queues container reference
+ leak in queues_data_provider_get(). * queue_unref() should not
+ return q that was just unreferenced. * Made logic in
+ __queues_show() and queues_data_provider_get() when calling
+ load_realtime_queue() easier to understand. ........
+
+2011-08-22 19:43 +0000 [r332877] Paul Belanger <pabelanger@digium.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 332876 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r332876 | pabelanger | 2011-08-22 15:41:24 -0400 (Mon,
+ 22 Aug 2011) | 6 lines Revert previous commit It seems google is
+ still making changes to the protocol. (issue ASTERISK-18301)
+ ........
+
+2011-08-22 19:41 +0000 [r332875] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Fix merge property.
+
+2011-08-22 18:40 +0000 [r332832] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c, include/asterisk/test.h, main/manager.c, /,
+ main/file.c, main/test.c, main/app.c,
+ configs/manager.conf.sample, include/asterisk/manager.h: Merged
+ revisions 332817 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332817 | mjordan | 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011)
+ | 4 lines Review: https://reviewboard.asterisk.org/r/1364/ This
+ update adds a new AMI event, TestEvent, which is enabled when the
+ TEST_FRAMEWORK compiler flag is defined. It also adds initial
+ usage of this event to app_voicemail. The TestEvent AMI event is
+ used extensively by the voicemail tests in the Asterisk Test
+ Suite. ........
+
+2011-08-22 18:32 +0000 [r332761-332830] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
+ revisions 332816 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332816 | rmudgett | 2011-08-22 13:14:59 -0500 (Mon, 22 Aug 2011)
+ | 8 lines Memory leaks in realtime_multi_xxx() when database
+ access returns error. * Fix realtime_multi_pgsql() configuration
+ memory leak when the database access returns an error. * Fix
+ realtime_multi_odbc() configuration category use after free when
+ the database access returns an error. ........
+
+ * /, main/config.c: Merged revisions 332759 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332759 | rmudgett | 2011-08-22 12:00:03 -0500 (Mon, 22 Aug 2011)
+ | 15 lines Memory leak reading realtime database variable list.
+ Calling ast_load_realtime() can leak the last list node if the
+ read list only contains empty variable value items. * Fixed list
+ filter loop in ast_load_realtime() to delete the list node
+ immediately instead of the next time through the loop. The next
+ time through the loop may not happen if the node to delete is the
+ last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265)
+ Patches: jira_asterisk_18265_v1.8_config.patch (license #5621)
+ patch uploaded by rmudgett ........
+
+2011-08-22 16:29 +0000 [r332756] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c, include/asterisk/res_fax.h: add a way to disable
+ and/or modify the gateway timeout ASTERISK-18219
+
+2011-08-21 14:33 +0000 [r332700] Paul Belanger <pabelanger@digium.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 332699 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r332699 | pabelanger | 2011-08-21 10:31:31 -0400 (Sun,
+ 21 Aug 2011) | 5 lines Fix outgoing calls in chan_gtalk (closes
+ issue ASTERISK-18301) Reported by: az1324 ........
+
+2011-08-19 19:59 +0000 [r332654] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_confbridge.c: Make CONFBRIDGE_INFO behave more nicely
+ CONFBRIDGE_INFO doesn't behave as well in edge cases as
+ MEETME_INFO. With this patch, CONFBRIDGE_INFO should behave in a
+ much more reasonable manner when presented with invalid
+ conferences and keywords. Review:
+ https://reviewboard.asterisk.org/r/1359/
+
+2011-08-18 21:34 +0000 [r332560] Terry Wilson <twilson@digium.com>
+
+ * main/netsock2.c, /: Merged revisions 332559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011)
+ | 5 lines Fix possible error on stringification of IPv4-mapped
+ addrs The FreeBSD netsock2 test has been failing for a while. We
+ were pasing sa->len to getnameinfo instead of sa_tmp->len.
+ ASTERISK-18289 ........
+
+2011-08-18 19:29 +0000 [r332504] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 332503 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r332503 | kmoore | 2011-08-18 14:28:00 -0500 (Thu, 18
+ Aug 2011) | 8 lines CRC4 in "dahdi show status" gives wrong
+ impression to T1 users Change CRC4 to CRC in the output of "dahdi
+ show status" so that it can apply in more situations without
+ confusing users, especially since T1 lines use CRC6 instead of
+ CRC4. (closes issue AST-471) ........
+
+2011-08-18 14:48 +0000 [r332369-332447] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * build_tools/cflags.xml, build_tools/cflags-devmode.xml, /: Merged
+ revisions 332446 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332446 | tilghman | 2011-08-18 09:46:54 -0500 (Thu, 18 Aug 2011)
+ | 2 lines Move BETTER_BACKTRACES out of development mode, as it's
+ useful when DEBUG_THREADS is enabled. ........
+
+ * Makefile, agi/Makefile, utils/Makefile, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ Makefile.moddir_rules, makeopts.in, sounds/Makefile: Merged
+ revisions 332355 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011)
+ | 10 lines Re-add support for spaces in pathnames, including now
+ spaces in DESTDIR. This was initially added to 1.8 prior to
+ release, primarily to support the standard paths on Mac OS X, but
+ was partially reverted recently in Subversion, due to the lack of
+ support for spaces in DESTDIR. This commit restores support for
+ the standard paths on Mac OS X, and also includes support for
+ spaces in DESTDIR. (closes issue ASTERISK-18290) Reported by:
+ pabelanger Review: https://reviewboard.asterisk.org/r/1326/
+ ........
+
+2011-08-17 18:09 +0000 [r332321] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_timing_timerfd.c: Merged revisions 332320 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r332320 | twilson | 2011-08-17 12:35:27 -0500 (Wed, 17
+ Aug 2011) | 10 lines Don't read from a disarmed or invalid
+ timerfd Numerous isues have been reported for deadlocks that are
+ caused by a blocking read in res_timing_timerfd on a file
+ descriptor that will never be written to. This patch adds some
+ checks to make sure that the timerfd is both valid and armed
+ before calling read(). Should fix: ASTERISK-18142,
+ ASTERISK-18166, ASTERISK-18197, AST-486, AST-495, AST-507 and
+ possibly others. Review: https://reviewboard.asterisk.org/r/1361/
+ ........
+
+2011-08-17 16:01 +0000 [r332265] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sig_pri.c: Merged revisions 332264 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011)
+ | 26 lines Outgoing BRI calls fail when using Asterisk 1.8 with
+ HA8, HB8, and B410P cards. France Telecom brings layer 2 and
+ layer 1 down on BRI lines when the line is idle. When layer 1
+ goes down Asterisk cannot make outgoing calls and the HA8 and HB8
+ cards also get IRQ misses. The inability to make outgoing calls
+ is because the line is in red alarm and Asterisk will not make
+ calls over a line it considers unavailable. The IRQ misses for
+ the HA8 and HB8 card are because the hardware is switching clock
+ sources from the line which just brought layer 1 down to internal
+ timing. There is a DAHDI option for the B410P card to not tell
+ Asterisk that layer 1 went down so Asterisk will allow outgoing
+ calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI
+ option for the HA8 and HB8 cards: "modprobe wctdm24xxp
+ bri_teignored=1". Unfortunately that will not clear up the IRQ
+ misses when the telco brings layer 1 down. * Add layer 2
+ persistence option to customize the layer 2 behavior on BRI PTMP
+ lines. The new option has three settings: 1) Use libpri default
+ layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when
+ the peer brings it down. 3) Leave layer 2 down when the peer
+ brings it down. Layer 2 will be brought up as needed for outgoing
+ calls. JIRA AST-598 ........
+
+2011-08-16 20:11 +0000 [r332177] Paul Belanger <pabelanger@digium.com>
+
+ * tests/test_amihooks.c, tests/test_substitution.c,
+ tests/test_heap.c, /, tests/test_expr.c,
+ tests/test_ast_format_str_reduce.c, tests/test_logger.c,
+ tests/test_gosub.c, tests/test_app.c, tests/test_dlinklists.c,
+ tests/test_event.c, tests/test_db.c, tests/test_linkedlists.c,
+ tests/test_sched.c, tests/test_netsock2.c, tests/test_pbx.c,
+ tests/test_strings.c, tests/test_func_file.c,
+ tests/test_security_events.c, tests/test_stringfields.c,
+ tests/test_time.c, tests/test_skel.c, tests/test_acl.c,
+ tests/test_locale.c, tests/test_utils.c,
+ tests/test_devicestate.c, tests/test_aoc.c, tests/test_astobj2.c,
+ tests/test_poll.c: Merged revisions 332176 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332176 | pabelanger | 2011-08-16 16:10:13 -0400 (Tue, 16 Aug
+ 2011) | 4 lines Flag test modules as 'core' Review:
+ https://reviewboard.asterisk.org/r/1369/ ........
+
+2011-08-16 17:45 +0000 [r332119] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 332118 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332118 | jrose | 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) |
+ 16 lines ASTERISK-18067 ASTERISK-15479 - White Space affects
+ mailbox value, multiple MWI subs Before, having multiple
+ subscriptions to mailboxes on a sip peer set via the mailbox
+ setting in sip.conf would only result in updates being sent on
+ whichever mailbox triggered the mwi event. Now all of them get
+ counted regardless. Also fixes a bug involving parsing of the
+ mailbox option in sip.conf so that trailing and leading spaces
+ before/after commas are trimmed. (closes issue ASTERISK-18067)
+ Reported by: aragon (closes issue ASTERISK-15479) Reported by:
+ Ben Winslow Patches:
+ chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288)
+ patch uploaded by Ben Winslow ........
+
+2011-08-16 17:17 +0000 [r332101] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c, CHANGES, configs/features.conf.sample,
+ main/asterisk.c: Merged revisions 332100 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011)
+ | 133 lines Fix multiple parking issues. JIRA ASTERISK-17183
+ Multi-parkinglot directs calls to wrong parkinglot. JIRA
+ ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430
+ ParkedCall() with no extension should pickup first available call
+ and does not. JIRA AST-576 Issues with parking lots * Removed
+ searching for parking lots by extension. Parking lots can only be
+ found by the parking lot name since parking lot access extensions
+ and spaces are not guaranteed to be unique. * Added
+ parking_lot_name option to the Park and ParkedCall applications.
+ Updated documentation for Park and ParkedCall applications. * Add
+ parkext_exclusive configuration option to make parking entry
+ extensions specify which parking lot they access. (closes issue
+ ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett,
+ David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi
+ Quezada (closes issue ASTERISK-17430) Reported by: Philippe
+ Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA
+ AST-624 'next' setting for findslot does nothing * Reimplemented
+ since findslot feature option broken by -r114655. (closes issue
+ ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett
+ JIRA ASTERISK-15792 Dialplan continues execution after transfer
+ to park. This happens for DTMF attended transfer, DTMF blind
+ transfer, and DTMF one-touch-parking if the party initiating
+ these features also initiated the call. * Fixed the return code
+ from the affected builtin features when parking a call. (closes
+ issue ASTERISK-15792) Reported by: Mat Murdock Tested by:
+ rmudgett, twilson JIRA AST-607 The courtesytone is not playing to
+ the expected call when picking up a parked call. This is mostly a
+ documentation problem. However, the option is not reset to the
+ default when features.conf is reloaded. * Updated
+ features.conf.sample documentation for courtesytone and
+ parkedplay options. * Reset the parkedplay option to default when
+ features.conf is reloaded. JIRA AST-615 AMI Park action followed
+ by features reload results in orphaned channels in parking lot. *
+ Reloading features.conf will not touch parking lots that have
+ calls still parked in them. Reload again at a later time. Misc
+ additional fixes: * Added unit test for parking lot dialplan
+ usage checking. * Made update connected line when a parked call
+ is retrieved from a parking lot. * Made retrieved parked call
+ stop ringing or MOH depending upon how the call was waiting in
+ the parking lot. * Made CLI "features show" indicate if the
+ parking lot is enabled for use. * Added PARKINGDYNEXTEN channel
+ variable to allow dynamic parking lots to specify the parking lot
+ access extension. * Made AMI ParkedCalls action ParkedCall events
+ have a Parkinglot header. * Made AMI ParkedCalls action
+ ParkedCallsComplete event have a Total header. * Fixed potential
+ deadlock from AMI Park action holding channel locks while calling
+ masq_park_call(). * Fixed several places where ast_strdupa() were
+ used inside of loops. (Mostly fixed by refactoring the loop body
+ into its own function.) * Fixed copy_parkinglot() copying too
+ much from the source parking lot. Extracted the parking lot
+ configuration settings into struct parkinglot_cfg. * Refactored
+ courtesytone playing code to put the channel not playing the tone
+ in autoservice. * Fix when pbx-parkingfailed is played that the
+ other channel is put in autoservice if it exists. * Fixed
+ parkinglot reference leak in parked_call_exec() error paths. *
+ Fixed parkinglot_unref() use of parkinglot after it was unreffed.
+ * Made destroy the struct ast_parkinglot parkings lock when done.
+ * Refactored the features.conf parking lot configuration code to
+ eliminate redundancy. * Fixed feature reload to better protect
+ parking lots. * Fixed parking lot container reference leak in
+ handle_parkedcalls(). * Fixed the total count in
+ handle_parkedcalls(). Review:
+ https://reviewboard.asterisk.org/r/1358/ ........
+
+2011-08-16 15:20 +0000 [r332022-332042] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/sip/include/sip.h: fix a code comment AST-580
+
+ * UPGRADE.txt, CHANGES: Moved notes about 'storesipcause' to
+ UPGRADE.txt from CHANGES AST-580
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
+ revisions 332026 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332026 | mnicholson | 2011-08-16 10:06:31 -0500 (Tue, 16 Aug
+ 2011) | 2 lines use DEFAULT_STORE_SIP_CAUSE to set the default
+ value for the 'storesipcause' option AST-580 ........
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: In 10
+ and trunk this option is disabled by default. Merged revisions
+ 332021 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug
+ 2011) | 7 lines Added the 'storesipcause' option to sip.conf to
+ allow the user to disable the setting of HASH(SIP_CAUSE,<chan
+ name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan
+ name>) on the channel carries a significant performance penalty
+ because of the usage of the MASTER_CHANNEL() dialplan function.
+ AST-580 ........
+
+2011-08-15 17:35 +0000 [r331956] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 331955 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r331955 | rmudgett | 2011-08-15 12:24:08 -0500 (Mon, 15
+ Aug 2011) | 13 lines Fix some minor chan_dahdi config load
+ issues. * Address chan_dahdi.conf dahdichan option todo item
+ about needing line number. * Make ignore_failed_channels option
+ also apply to dahdichan option. * Don't attempt to create a
+ default pseudo channel if the chan_dahdi.conf channel/channels
+ option is not allowed. * Add a similar check for dahdichan in
+ normal chan_dahdi.conf sections as is done in users.conf.
+ ........
+
+2011-08-15 15:22 +0000 [r331894] Paul Belanger <pabelanger@digium.com>
+
+ * main/rtp_engine.c, /: Merged revisions 331886 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331886 | pabelanger | 2011-08-15 11:21:16 -0400 (Mon, 15 Aug
+ 2011) | 5 lines Fix noisy message when briding channels (closes
+ issue ASTERISK-18270) Reported by: Federico Alves ........
+
+2011-08-15 15:14 +0000 [r331868] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 331867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331867 | dvossel | 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011)
+ | 6 lines Fixes locking inversion issues present in the handling
+ of the sip REFER method. (closes issue ASTERISK-18082) Reported
+ by: James Van Vleet ........
+
+2011-08-12 19:03 +0000 [r331775] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 331774 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331774 | mnicholson | 2011-08-12 14:01:27 -0500 (Fri, 12 Aug
+ 2011) | 11 lines Unlock the channel before calling update_queue.
+ Holding the channel lock when calling update_queue which attempts
+ to lock the queue lock can cause a deadlock. This deadlock
+ involves the following chain: 1. hold chan lock -> wait queue
+ lock 2. hold queue lock -> wait agent list lock 3. hold agent
+ list lock -> wait chan list lock 4. hold chan list lock -> wait
+ chan lock ........
+
+2011-08-12 18:59 +0000 [r331715-331772] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 331771 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r331771 | rmudgett | 2011-08-12 13:58:40 -0500 (Fri, 12
+ Aug 2011) | 8 lines Suppress warning message when using
+ DAHDITransfer or DAHDIHangup. * The fake event should only be
+ processed by the channel that currently owns the private and not
+ the associated call waiting or 3-way channel. JIRA AST-620 JIRA
+ SWP-3616 ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 331714 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r331714 | rmudgett | 2011-08-12 12:47:57 -0500 (Fri, 12
+ Aug 2011) | 22 lines AMI actions DAHDIHangup and DAHDITransfer
+ have no effect. The AMI actions DAHDIHangup and DAHDITransfer
+ have no effect on a DAHDI channel. These two AMI actions are
+ highly specialized to analog channels and appear to make the
+ channel behave like a jack port for headsets. * Made the faked
+ DAHDI event get processed before a normal media stream read in
+ dahdi_read() instead of trying to trigger an exception read by
+ setting the AST_FLAG_EXCEPTION flag. Apparently a change was made
+ long ago that changed how AST_FLAG_EXCEPTION is processed in the
+ core. Unfortunately, the faked DAHDI events no longer worked when
+ that happened. * Updated the DAHDI AMI action documentation for
+ the following actions: DAHDITransfer, DAHDIHangup,
+ DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and
+ DAHDIRestart. * Made use sscanf() instead of atoi() for better
+ error checking of the DAHDIChannel header string. JIRA AST-620
+ JIRA SWP-3616 ........
+
+2011-08-12 16:31 +0000 [r331659] Terry Wilson <twilson@digium.com>
+
+ * /, tests/test_netsock2.c: Merged revisions 331658 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r331658 | twilson | 2011-08-12 11:30:26 -0500 (Fri, 12
+ Aug 2011) | 4 lines Fix netsock2 multiple zero-expansion test
+ Remove erroneous single bracket. ........
+
+2011-08-12 16:21 +0000 [r331654] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/logger.c: Merged revisions 331649 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331649 | kmoore | 2011-08-12 11:20:25 -0500 (Fri, 12 Aug 2011) |
+ 12 lines Logger does not warn of failure to open logging channels
+ Currently, logger only prints an error message to stderr when it
+ fails to open a logger channel where many users will not see it
+ because the logger lock is held. The alternative provided by this
+ patch is to log the error to all attached consoles in the hopes
+ that it will be easier to see. Additionally, this patch prevents
+ the failed logger channel from being added to the list where it
+ would silently fail on each call to the Asterisk logger. (closes
+ issue ASTERISK-16231) Review:
+ https://reviewboard.asterisk.org/r/1338 ........
+
+2011-08-12 16:18 +0000 [r331644] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331635
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331635 | jrose | 2011-08-12 10:49:17 -0500 (Fri, 12 Aug 2011) |
+ 1 line Fixes 32bit compilation warnings brought on by 331634 in
+ app_dial and app_meetme ........
+
+2011-08-11 21:54 +0000 [r331579] Jason Parker <jparker@digium.com>
+
+ * apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331578
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331578 | qwell | 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) |
+ 6 lines Use proper values for 64-bit option flags. Also, reusing
+ bits es no bueno, so change the value of a duplicate. (issue
+ ASTERISK-18239) ........
+
+2011-08-11 21:42 +0000 [r331576] Richard Mudgett <rmudgett@digium.com>
+
+ * /, funcs/func_shell.c: Merged revisions 331575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011)
+ | 9 lines Segfault in shell_helper in func_shell.c. The return
+ value of popen() was not checked for failure to open. (closes
+ issue ASTERISK-18109) JIRA SWP-3633 Reported by: Michael Myles
+ Tested by: rmudgett ........
+
+2011-08-10 22:23 +0000 [r331518] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 331517 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) |
+ 10 lines SIP Notify via AMI or CLI leaks SIP PVTs Any SIP notify
+ sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing
+ the additional ref just before the invite and adding an unref
+ following it corrects the issue as seen via REF_DEBUG. The unref
+ existed in a distant revision and it appears as though the wrong
+ ref operation was removed. (closes issue ASTERISK-18091) Review:
+ https://reviewboard.asterisk.org/r/1332/ ........
+
+2011-08-10 20:41 +0000 [r331418-331462] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/logger.c: Merged revisions 331461 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331461 | rmudgett | 2011-08-10 15:29:59 -0500 (Wed, 10 Aug 2011)
+ | 30 lines Output of queue log not started until logger reloaded.
+ ASTERISK-15863 caused a regression with queue logging. The output
+ of the queue log is not started until the logger configuration is
+ reloaded. * Queue log initialization is completely delayed until
+ the first message is posted to the queue log system. Including
+ the initial opening of the queue log file. * Fixed rotate_file()
+ ROTATE strategy to give the file just rotated out to the
+ configured exec function after rotate. Just like the other
+ strategies. * Fixed logger reload to always post the queue reload
+ entry instead of just if there is a queue log file. * Refactored
+ some code to eliminate some redundancy and to reduce stack
+ utilization. (closes issue ASTERISK-17036) JIRA SWP-2952 Reported
+ by: Juan Carlos Valero Patches: jira_asterisk_17036_v1.8.patch
+ (license #5621) patch uploaded by rmudgett Tested by: rmudgett
+ (closes issue ASTERISK-18208) Reported by: Christian Pinedo
+ Review: https://reviewboard.asterisk.org/r/1333/ ........
+
+ * main/features.c: Make sure feature_request_and_dial() initializes
+ outstate if passed in.
+
+ * main/features.c, CHANGES: Revert -r318141. It was a band-aid that
+ only partially fixed parking. A better fix is on reviewboard
+ review 1358. (issue ASTERISK-17374)
+
+2011-08-10 13:48 +0000 [r331316] Kinsey Moore <kmoore@digium.com>
+
+ * main/manager.c, /: Merged revisions 331315 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331315 | kmoore | 2011-08-10 08:47:46 -0500 (Wed, 10 Aug 2011) |
+ 8 lines AMI action ModuleReload returns Error if Module: missing
+ or empty An empty string was not being checked for properly
+ causing identification of the module to be reloaded to fail and
+ return an Error with message "No such module." (closes issue
+ AST-616) ........
+
+2011-08-09 23:12 +0000 [r331265] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_parkandannounce.c, main/pbx.c, /, channels/chan_sip.c,
+ main/features.c, channels/chan_iax2.c: Merged revisions 331248
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011)
+ | 15 lines Misc minor items found in code. * Add some reentrancy
+ protection in pbx.c when creating the contexts_table hash table.
+ * Fix inverted test in chan_sip.c conditional code. * Fix
+ uninitialized variable and use of the wrong variable in
+ chan_iax2.c. * Fix test of return value in app_parkandannounce.c.
+ Explicitly testing for -1 is bad if the function does not
+ actually return that value when it fails. * Fixup some comments
+ and add some curly braces in features.c. ........
+
+2011-08-09 16:36 +0000 [r331147-331200] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooGkClient.c:
+ Setup IP proto version for call in GK mode Added additional check
+ for IP semantics before parse destination by ast_parse_args due
+ to it can parse numeric as IP. (closes issue ASTERISK-18218)
+ Reported by: slesru Patch: ASTERISK-18218.patch
+
+ * addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c, /,
+ addons/ooh323c/src/ooLogChan.c: Merged revisions 331146 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331146 | may | 2011-08-09 20:13:09 +0400 (Tue, 09 Aug 2011) | 4
+ lines move ast_cond_signal for admitted call after all data
+ filled/freed clear all log channels by pointed number not only
+ first free allocated callToken in ooh323_answer ........
+
+2011-08-09 15:59 +0000 [r331138-331143] Jason Parker <jparker@digium.com>
+
+ * /, doc/asterisk.8: Merged revisions 331142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331142 | qwell | 2011-08-09 10:58:16 -0500 (Tue, 09 Aug 2011) |
+ 1 line Regenerate asterisk man page from sgml. ........
+
+ * doc/asterisk.sgml, /, doc/asterisk.8,
+ configs/asterisk.conf.sample, configs/voicemail.conf.sample:
+ Merged revisions 306999 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r306999 | lathama | 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011)
+ | 12 lines Documentation Updates Note default polling setting in
+ voicemail.conf Add missing config to asterisk.conf Update manpage
+ (issue #16505) Reported by: tzafrir Patches:
+ asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46)
+ Tested by: lathama, tzafrir ........
+
+ * doc/asterisk.sgml, /, doc/asterisk.8,
+ configs/asterisk.conf.sample, configs/voicemail.conf.sample:
+ Revert merge of r306999, due to merge conflict.
+
+2011-08-08 22:59 +0000 [r331041-331097] Terry Wilson <twilson@digium.com>
+
+ * UPGRADE.txt, CHANGES, include/asterisk/manager.h: Bump the AMI
+ protocol version to 1.2 As a result of converting Unlink events
+ that were missed in the AMI 1.1 update to Bridge events, the AMI
+ protocol version is being incremented.
+
+ * main/channel.c, CHANGES: Replace AMI Unlink events with Bridge
+ events A previous update converted some of the Link and Unlink
+ events to Bridge events, but a couple of Unlink events were
+ missed. This patch rectifies the situation. (closes issue
+ ASTERISK-17455)
+
+2011-08-08 20:53 +0000 [r331039] Kinsey Moore <kmoore@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 331038 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r331038 | kmoore | 2011-08-08 15:52:45 -0500 (Mon, 08
+ Aug 2011) | 11 lines In-queue MOH stops after a periodic
+ announcement If the seek value is past the end of file when
+ resuming G.722 MOH, MOH will cease to function for the duration
+ of the MOH session through all starts and stops until saved state
+ is cleared. Adjusting the code to guarantee a single valid read
+ (which is already assumed) fixes the bug. (closes issue
+ ASTERISK-18077) Review: https://reviewboard.asterisk.org/r/1328/
+ Tested-by: Jonathan Rose <jrose@digium.com> ........
+
+2011-08-05 15:53 +0000 [r330940] David Vossel <dvossel@digium.com>
+
+ * codecs/codec_resample.c: The slin resampler is no longer
+ dependent on an external library, but the dependency was not
+ removed correctly.
+
+2011-08-05 07:38 +0000 [r330899] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c, /,
+ addons/ooh323c/src/ooCmdChannel.c: Merged revisions 330827 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330827 | may | 2011-08-04 23:37:16 +0400 (Thu, 04 Aug 2011) | 4
+ lines change gk client behaivour on rrq/grq failures to setup
+ timers and next tries after timeout instead of complete failure
+ in the ooh323 stack ........
+
+2011-08-04 20:51 +0000 [r330844] Terry Wilson <twilson@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 330843 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r330843 | twilson | 2011-08-04 15:29:19 -0500 (Thu, 04
+ Aug 2011) | 4 lines Make libsrtp instructions more explicit when
+ linking fails (closes issue ASTERISK-18139) ........
+
+2011-08-03 15:15 +0000 [r330706-330763] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/Makefile: Merged revisions 330762 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330762 | kmoore | 2011-08-03 10:14:36 -0500 (Wed, 03 Aug 2011) |
+ 9 lines editing files in main/editline does not ensure rebuild of
+ libedit.a When editing a source file in main/editline, the build
+ system does not rebuild libedit.a and uses the already existing
+ one instead. Adding a PHONY to CHECK_SUBDIR fixes this problem.
+ (closes issue ASTERISK-16221) Patch-by: Walter Doekes ........
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 330705 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330705 | kmoore | 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) |
+ 10 lines Call pickup broken for DAHDI channels when beginning
+ with # The call pickup feature did not work on DAHDI devices for
+ anything other than feature codes beginning with * since all
+ feature codes in chan_dahdi were originally hard-coded to begin
+ with *. This patch is also applied to chan_dahdi.c to fix this
+ bug with radio modes. (closes issue AST-621) Review:
+ https://reviewboard.asterisk.org/r/1336/ ........
+
+2011-08-02 20:52 +0000 [r330649] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, res/res_jabber.c: Merged revisions 330648 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330648 | kpfleming | 2011-08-02 15:51:56 -0500 (Tue, 02 Aug
+ 2011) | 2 lines Convert an error message to actually be helpful.
+ ........
+
+2011-08-02 16:17 +0000 [r330576-330586] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 330581 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r330581 | dvossel | 2011-08-02 11:15:08 -0500 (Tue, 02
+ Aug 2011) | 8 lines Fixes crash in chan_iax2. Fixes crash in
+ chan_iax2 resulting from an edge case in the way control frames
+ are queued during calltoken negotiation is complete. (closes
+ issue ASTERISK-17610) Reported by: mgrobecker ........
+
+ * /, channels/chan_sip.c: Merged revisions 330578 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330578 | dvossel | 2011-08-02 11:07:02 -0500 (Tue, 02 Aug 2011)
+ | 2 lines Optimization to buffer initialization fix. ........
+
+ * /, channels/chan_sip.c: Merged revisions 330575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330575 | dvossel | 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011)
+ | 5 lines Fixes uninitialized string buffer in log message.
+ (closes issue ASTERISK-17200) Reported by: lmadsen ........
+
+2011-08-01 15:23 +0000 [r330434] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/say.c: Merged revisions 330433 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330433 | kmoore | 2011-08-01 10:22:10 -0500 (Mon, 01 Aug 2011) |
+ 9 lines Incorrect playback for Spanish in some circumstances When
+ you say the time in spanish and it is 01:00 - 01:59 or 13:00 -
+ 13:59 you must use female pronunciation "1F". The function
+ "say_date_with_format_es" does not take this in account. (closes
+ ASTERISK-15016) Patch-by: Luis Jimenez ........
+
+2011-07-30 23:57 +0000 [r330369] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Merged revisions 330368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330368 | rmudgett | 2011-07-30 18:56:29 -0500 (Sat, 30 Jul 2011)
+ | 4 lines Remove some redundant locking code in
+ ast_do_masquerade(). Also updated some comments. ........
+
+2011-07-30 15:34 +0000 [r330312] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * main/channel.c, /: Merged revisions 330311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330311 | irroot | 2011-07-30 17:25:16 +0200 (Sat, 30 Jul 2011) |
+ 9 lines prevent double masqurading channels when one is been hung
+ up and deadlock avoidance is used. There is a race condition in
+ ast_do_masquerade / ast_hangup (at least) Reported by me signed
+ off by schmidts with input from David Vossel Review:
+ https://reviewboard.asterisk.org/r/1323/ ........
+
+2011-07-29 17:19 +0000 [r330204-330217] Sean Bright <sean@malleable.com>
+
+ * /, formats/format_wav.c: Merged revisions 330213 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r330213 | seanbright | 2011-07-29 13:18:56 -0400 (Fri,
+ 29 Jul 2011) | 2 lines Correct the check for O_RDONLY. ........
+
+ * /, formats/format_wav.c: Merged revisions 330203 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r330203 | seanbright | 2011-07-29 12:58:08 -0400 (Fri,
+ 29 Jul 2011) | 2 lines Only write to wav files that were opened
+ to be written to. ........
+
+2011-07-29 05:25 +0000 [r330162] Paul Belanger <pabelanger@digium.com>
+
+ * apps/app_confbridge.c: Fix typo pointed out on #asterisk Thanks
+ notten
+
+2011-07-28 21:44 +0000 [r330108] Terry Wilson <twilson@digium.com>
+
+ * main/term.c, /: Merged revisions 330107 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330107 | twilson | 2011-07-28 16:42:41 -0500 (Thu, 28 Jul 2011)
+ | 2 lines Make console colors work for TERM=xterm-256color
+ ........
+
+2011-07-28 17:10 +0000 [r330051] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 330050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r330050 | rmudgett | 2011-07-28 12:04:24 -0500
+ (Thu, 28 Jul 2011) | 22 lines Merged revisions 330033 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu,
+ 28 Jul 2011) | 15 lines Datacalls with B410P fail. Incoming and
+ outgoing call legs of a data call are using different formats:
+ a-law, u-law. When the call is bridged, the media stream is run
+ through translation to convert the media formats. The translation
+ is bad for data calls. * Make incoming call that does not
+ explicitly specify u-law or a-law use the DAHDI channel's default
+ law. The outgoing call always uses the default law from the DAHDI
+ channel. (closes issue ABE-2800) Patches:
+ jira_abe_2800_companding.patch (license #5621) patch uploaded by
+ rmudgett .......... ................
+
+2011-07-28 15:45 +0000 [r329995] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 329994 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329994 | qwell | 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) |
+ 6 lines Fix a SIP transfer deadlock. The locking in this function
+ is very scary. There are like 6 structs involved. (closes issue
+ AST-470) ........
+
+2011-07-28 15:28 +0000 [r329992] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 329991 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329991 | mnicholson | 2011-07-28 10:26:56 -0500 (Thu, 28 Jul
+ 2011) | 6 lines check for CONFIG_STATUS_FILE_INVALID when loading
+ the res_fax config file Patch by: tzafrir Reported by: tzafrir
+ (closes issue ASTERISK-18161) ........
+
+2011-07-28 13:03 +0000 [r329771-329952] Sean Bright <sean@malleable.com>
+
+ * configs/confbridge.conf.sample: The default conf-usermenu says
+ that '8' can be used to leave the conference, so put that in the
+ sample user menu. '5' is supposed to extend the conference, but
+ there doesn't appear to be a concept of that in the menu actions.
+
+ * apps/app_confbridge.c: Correct the spelling of 'conference.'
+
+ * /, channels/chan_sip.c: Merged revisions 329895 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329895 | seanbright | 2011-07-28 07:34:33 -0400 (Thu, 28 Jul
+ 2011) | 2 lines Make the output of Externhost in 'sip show
+ settings' more consistent. ........
+
+ * /, Makefile.moddir_rules: Merged revisions 329767 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r329767 | seanbright | 2011-07-27 15:17:46 -0400 (Wed,
+ 27 Jul 2011) | 8 lines Explicitly sort the module list so that
+ the menuselect lists are sorted. (closes issue ASTERISK-18141)
+ Reported by: Richard Miller Patches: sort-order.diff uploaded by
+ seanbright (License #5060) Tested by: leifmadsen ........
+
+2011-07-27 18:11 +0000 [r329710] Jonathan Rose <jrose@digium.com>
+
+ * /, configs/indications.conf.sample: Merged revisions 329709 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329709 | jrose | 2011-07-27 13:10:30 -0500 (Wed, 27 Jul 2011) |
+ 8 lines Fix New Zealand indications profile based on
+ http://www.telepermit.co.nz/TNA102.pdf (closes issue
+ ASTERISK-16263) Reported by: richardf Patches:
+ nz-indications.patch uploaded by richardf (License #6015)
+ ........
+
+2011-07-27 15:25 +0000 [r329670] Sean Bright <sean@malleable.com>
+
+ * main/loader.c: Sort the module list so that 'module show' is
+ alphabetical.
+
+2011-07-27 04:25 +0000 [r329614] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, cdr/cdr_odbc.c: Merged revisions 329613 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329613 | tilghman | 2011-07-26 23:23:46 -0500 (Tue, 26 Jul 2011)
+ | 6 lines Duration and billsec are swapped in high resolution
+ time. Closes ASTERISK-18024 Patches:
+ 20110726__ASTERISK-18024.diff by Tilghman Lesher (License 5003)
+ ........
+
+2011-07-26 14:19 +0000 [r329528-329538] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 329529 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r329529 | jrose | 2011-07-26 09:04:55 -0500 (Tue, 26 Jul
+ 2011) | 5 lines Changes sound file for prepend "then-press-pound"
+ to "vm-then-pound" which is the same prompt, only it turned out
+ "then-press-pound" was part of extra sounds. Also, vm is more
+ appropriate anyway. ........
+
+ * apps/app_voicemail.c, include/asterisk/app.h, /,
+ configs/voicemail.conf.sample, main/app.c: Merged revisions
+ 329527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) |
+ 17 lines Fixes some voicemail forwarding behavior based around
+ prepend mode. Formerly, prepend forwarding would have the user
+ record a message with no useful prompt and an expectation for the
+ user to push a button on the phone when finished recording. If a
+ length of silence was detected instead, the recording would be
+ canceled and the user would re-enter the voicemail forwarding
+ menu. Subsequent time-outs in prepend recording would also bug
+ out in the sense that they would write over the original message
+ and get sent to the recipient regardless of whether they timed
+ out or were accepted. This patch fixes this issue and adds a
+ prompt which will be played after a timeout informing the user
+ that they needed to press a button. Currently, the sound files
+ that we have are somewhat inadquate for this, so after the call
+ we simply have Allison say "Please try again. Then press pound."
+ which actually relies on two separate sound files. Just one would
+ be more appropriate. reporter: Vlad Povorozniuc Review:
+ https://reviewboard.asterisk.org/r/1327/ ........
+
+2011-07-25 19:55 +0000 [r329472] Paul Belanger <pabelanger@digium.com>
+
+ * /, main/enum.c: Merged revisions 329471 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329471 | pabelanger | 2011-07-25 15:49:40 -0400 (Mon, 25 Jul
+ 2011) | 2 lines Decrease verbose messages to debug, to help clean
+ up CLI. ........
+
+2011-07-25 14:06 +0000 [r329430-329431] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * main/dsp.c, include/asterisk/dsp.h: dsp_process was enhanced to
+ work with alaw and ulaw in addition to slin. noticed that some
+ functions could be refactored here it is. Reported by: irroot
+ Tested by: irroot, mnicholson Review:
+ https://reviewboard.asterisk.org/r/1304/
+
+ * channels/chan_sip.c, channels/sip/include/sip.h: Remove
+ lastmsgssent from sip it has not been working since 1.6 Clean up
+ the return values to be consistant not currently used Add doxygen
+ returns MWI Event is sent on Register (closes issue
+ ASTERISK-17866) Reported by: one47 Tested by: irroot, mvanbaak
+ Review: https://reviewboard.asterisk.org/r/1272/
+
+2011-07-22 21:14 +0000 [r329331-329334] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /: Make use less redundant loop construct for
+ iterating over hints.
+
+ * main/pbx.c, /: Merged revisions 329299 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329299 | rmudgett | 2011-07-22 10:44:58 -0500 (Fri, 22 Jul 2011)
+ | 48 lines Deadlocks dealing with dialplan hints during reload.
+ There are two remaining different deadlocks reported dealing with
+ dialplan hints. The deadlock in ASTERISK-17666 is caused by
+ invalid locking order in ast_remove_hint(). The hints container
+ must be locked before the hint object. The deadlock in
+ ASTERISK-17760 is caused by a catch-22 situation in
+ handle_statechange(). The deadlock is caused by not having the
+ conlock before calling the watcher callbacks. Unfortunately,
+ having that lock causes a different deadlock as reported in
+ ASTERISK-16961. * Fixed ast_remove_hint() locking order. * Made
+ handle_statechange() no longer call the watcher callbacks holding
+ any locks that matter. * Made hint ao2 destructor do the watcher
+ callbacks for extension deactivation to guarantee that they get
+ called. * Fixed hint reference leak in ast_add_hint() if the
+ callback container constructor failed. * Fixed hint reference
+ leak in complete_core_show_hint() for every hint it found for CLI
+ tab completion. * Adjusted locking in
+ ast_merge_contexts_and_delete() for safety. * Added
+ context_merge_lock to prevent ast_merge_contexts_and_delete() and
+ handle_statechange() from interfering with each other. * Fixed
+ ast_change_hint() not taking into account that the extension is
+ used for the hash key. (closes issue ASTERISK-17666) Reported by:
+ irroot Tested by: irroot JIRA SWP-3318 (closes issue
+ ASTERISK-17760) Reported by: Byron Clark Tested by: irroot JIRA
+ SWP-3393 Review: https://reviewboard.asterisk.org/r/1313/
+ ........
+
+2011-07-22 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 10.0.0-beta1 Released.
+
+2011-07-21 20:22 +0000 [r329257] Russell Bryant <russell@digium.com>
+
+ * channels/chan_dahdi.c, main/features.c,
+ include/asterisk/netsock2.h, CHANGES, channels/sig_pri.c,
+ include/asterisk/rtp_engine.h: s/1.10/10.0/
+
+2011-07-21 18:05 +0000 [r329200-329204] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
+ revisions 329203 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011)
+ | 6 lines Document parkinglot in chan_dahdi.conf.sample. *
+ Document existing feature in chan_dahdi.conf.sample. * Remove
+ some dead code related to the parkinglot option. ........
+
+ * /, apps/app_directed_pickup.c: Merged revisions 329199 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011)
+ | 17 lines Update PickupChan documentation. The PickupChan uses
+ the ampersand as the argument separator. Was documented as:
+ PickupChan(channel[,channel2[,...][,options]]) Fixed
+ documentation to:
+ PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
+ This is a continuation of ASTERISK-17494 for v1.8 and later.
+ (closes issue ASTERISK-18144) Reported by: Erik Smith Patches:
+ pickupchan_ducumentation-v2.patch (License #6263) patch uploaded
+ by Erik Smith Tested by: Erik Smith ........
+
+2011-07-21 17:27 +0000 [r329188] Jason Parker <jparker@digium.com>
+
+ * UPGRADE.txt: Fix version number in UPGRADE.txt.
+
+2011-07-21 16:52 +0000 [r329145] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 329144 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011)
+ | 9 lines Dialplan bridge() app mutex 'current_dest_chan' freed
+ more times than we've locked! This appears to be a leftover from
+ when ast_channel was converted to ao2 objects. Simply removed the
+ extraneous unlock. (closes issue ASTERISK-17772) ........
+
+2011-07-21 16:04 +0000 [r329104] Russell Bryant <russell@digium.com>
+
+ * / (added): Change Asterisk 2.0 to 2.0 in binary
+
+2011-07-20 21:31 +0000 [r329056] Paul Belanger <pabelanger@digium.com>
+
+ * /, UPGRADE-1.8.txt: Merged revisions 329055 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/2.0
+ ................ r329055 | pabelanger | 2011-07-20 17:27:50 -0400
+ (Wed, 20 Jul 2011) | 9 lines Merged revisions 329027 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r329027 | pabelanger | 2011-07-20 17:20:36 -0400 (Wed,
+ 20 Jul 2011) | 2 lines Asterisk now requires libpri 1.4.11+ for
+ PRI support. ........ ................
+
+2011-07-20 20:19 +0000 [r328996] Terry Wilson <twilson@digium.com>
+
+ * /, tests/test_netsock2.c: Merged revisions 328992 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/2.0
+ ................ r328992 | twilson | 2011-07-20 15:18:25 -0500
+ (Wed, 20 Jul 2011) | 12 lines Merged revisions 328987 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328987 | twilson | 2011-07-20 15:16:58 -0500 (Wed, 20 Jul 2011)
+ | 5 lines We can't guarantee an eth0 is present FreeBSD test
+ fails on this case presumably because there is no eth0 on the
+ test machine. Better to just remove this test for now. ........
+ ................
+
+2011-07-20 19:03 +0000 [r328937] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 328936 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/2.0
+ ................ r328936 | kmoore | 2011-07-20 14:01:37 -0500
+ (Wed, 20 Jul 2011) | 15 lines Merged revisions 328935 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) |
+ 8 lines Inband DTMF regression The functionality of inband DTMF
+ in chan_sip relied upon ast_rtp_instance_dtmf_mode_get/set not
+ working properly to avoid calling ast_rtp_instance_dtmf_begin/end
+ on RTP streams with inband DTMF. According to documentation,
+ ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
+ never inband. This fixes the regression introduced in revision
+ 328823. ........ ................
+
+2011-07-19 21:32 +0000 [r328880-328881] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /, Makefile.moddir_rules, sounds/Makefile: Merged
+ revisions 328879 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/2.0
+ ................ r328879 | kpfleming | 2011-07-19 16:31:16 -0500
+ (Tue, 19 Jul 2011) | 23 lines Merged revisions 328878 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul
+ 2011) | 17 lines Revert partial attempt at handling pathnames
+ with spaces. Revision 299794 attempted to improve the build
+ system to be able to handle pathnames (primarily DESTDIR) with
+ spaces in them, since this is common on some platforms (including
+ Mac OSX). Unfortunately, the changes were incomplete and did not
+ actually provide the desired behavior, and as a side effect the
+ functionality that ensured that stale headers in the Asterisk
+ 'include' directory were removed got broken. In addition, the
+ check for stale (and possibly incompatible) modules in the
+ Asterisk 'modules' directory also got broken, and would never
+ report any stale modules. Users upgrading to this version or
+ later versions would then see unexpected module load errors.
+ Since there are few users who actually want to install Asterisk
+ into paths that contain spaces, and a proper fix for the build
+ system would take many hours, the best solution for now is to
+ just revert the partial solution. ........ ................
+
+ * /: Edit the merge properties to match their names.
+
+2011-07-19 21:21 +0000 [r328877] Russell Bryant <russell@digium.com>
+
+ * /: Fix properties after twilson's change so merges still work
+
+2011-07-19 18:07 +0000 [r328772-328825] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, /,
+ channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged
+ revisions 328824 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328824 | kmoore | 2011-07-19 13:05:21 -0500
+ (Tue, 19 Jul 2011) | 18 lines Merged revisions 328823 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) |
+ 11 lines RTP bridge away with inband DTMF and feature detection
+ When deciding whether Asterisk was allowed to bridge the call
+ away from the core, chan_sip did not take into account the usage
+ of features on dialed channels that require monitoring of DTMF on
+ channels utilizing inband DTMF. This would cause Asterisk to
+ allow the call to be locally or remotely bridged, preventing
+ access to the data required to detect activations of such
+ features. (closes 17237) Review:
+ https://reviewboard.asterisk.org/r/1302/ ........
+ ................
+
+ * /, apps/app_meetme.c: Merged revisions 328771 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328771 | kmoore | 2011-07-19 10:46:54 -0500
+ (Tue, 19 Jul 2011) | 18 lines Merged revisions 328770 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) |
+ 11 lines MeetMe requests a PIN twice in some circumstances If a
+ call to MeetMe includes both the dynamic(D) and always request
+ PIN(P) options, MeetMe will ask for the PIN two times: once for
+ creating the conference and once for entering the conference.
+ This behavior was introduced in rev 311616 when adding the
+ CONFFLAG_ALWAYSPROMPT option to the logic branch controlling PIN
+ entry for joining a conference. (closes AST-601) Review:
+ https://reviewboard.asterisk.org/r/1305/ ........
+ ................
+
+2011-07-19 02:00 +0000 [r328718] Terry Wilson <twilson@digium.com>
+
+ * /, include/asterisk/linkedlists.h, tests/test_linkedlists.c
+ (added): Merged revisions 328717 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328717 | twilson | 2011-07-18 20:55:32 -0500
+ (Mon, 18 Jul 2011) | 14 lines Merged revisions 328716 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328716 | twilson | 2011-07-18 20:35:53 -0500 (Mon, 18 Jul 2011)
+ | 7 lines Make AST_LIST_REMOVE safer AST_LIST_REMOVE shouldn't
+ modify the element passed in if it isn't found. This commit also
+ adds linked list unit tests. Review:
+ https://reviewboard.asterisk.org/r/1321/ ........
+ ................
+
+2011-07-18 20:51 +0000 [r328610-328665] Mark Murawki <markm@intellasoft.net>
+
+ * apps/app_dial.c, /: Merged revisions 328664 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328664 | markm | 2011-07-18 16:50:13 -0400
+ (Mon, 18 Jul 2011) | 15 lines Merged revisions 328663 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) |
+ 9 lines app_dial may double free a channel datastore When
+ starting a call with originate, and having the callee channel run
+ Bridge() on pickup, we will double free the dialed_interface_info
+ datastore, causing a crash. Make sure to check if the datastore
+ still exists before trying to free it. (closes issue
+ ASTERISK-17917) Reported by: Mark Murawski Tested by: Mark
+ Murawski ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 328611 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328611 | markm | 2011-07-18 08:56:49 -0400
+ (Mon, 18 Jul 2011) | 15 lines Merged revisions 328608 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) |
+ 9 lines If the sip private structure is null, sip_setoption()
+ will defref the null pointer and crash. Ideally, sip_setoption
+ shouldn't be called if there is a lack of a sip private
+ structure. But this will fix a crash. (closes issue
+ ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark
+ Murawski ........ ................
+
+ * /, main/asterisk.c: Merged revisions 328609 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328609 | markm | 2011-07-18 08:37:53 -0400
+ (Mon, 18 Jul 2011) | 15 lines Merged revisions 328593 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328593 | markm | 2011-07-18 08:06:50 -0400 (Mon, 18 Jul 2011) |
+ 8 lines Fixed invalid read and null pointer deref on asterisk
+ shutdown. In some cases when starting asterisk with -c and
+ hitting control-c to shutdown, there will be an invalid read and
+ null pointer deref causing a crash. (closes issue ASTERISK-17927)
+ Reported by: Mark Murawski Tested by: Mark Murawski, Kinsey
+ Moore, Tilghman Lesher ........ ................
+
+2011-07-18 07:12 +0000 [r328542] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, funcs/func_odbc.c: Merged revisions 328541 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328541 | tilghman | 2011-07-18 02:11:26 -0500
+ (Mon, 18 Jul 2011) | 9 lines Merged revisions 328540 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r328540 | tilghman | 2011-07-18 02:10:15 -0500 (Mon, 18
+ Jul 2011) | 2 lines Typo ........ ................
+
+2011-07-15 21:41 +0000 [r328502] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c, /: Merged revisions
+ 328428-328429 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328428 | may | 2011-07-15 23:31:09 +0400 (Fri,
+ 15 Jul 2011) | 13 lines Merged revisions 328427 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328427 | may | 2011-07-15 23:22:24 +0400 (Fri, 15 Jul 2011) | 7
+ lines small gk processing fixes: - decrease for 1 second
+ registration ttl for very low expirations (some providers expire
+ few earlier than TTL) - delete rrq and registration expire timers
+ on URQ received as we make new registration. ........
+ ................ r328429 | may | 2011-07-15 23:35:50 +0400 (Fri,
+ 15 Jul 2011) | 2 lines delete unproperly changed svn props
+ ................
+
+2011-07-15 21:19 +0000 [r328449-328459] Leif Madsen <lmadsen@digium.com>
+
+ * /, apps/app_macro.c: Merged revisions 328451 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10 ........
+ r328451 | lmadsen | 2011-07-15 16:17:25 -0500 (Fri, 15 Jul 2011)
+ | 1 line Build app_macro by default because things depend on it.
+ ........
+
+ * /, UPGRADE-1.10.txt, UPGRADE.txt, CHANGES: Merged revisions
+ 328448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10 ........
+ r328448 | lmadsen | 2011-07-15 16:57:15 -0400 (Fri, 15 Jul 2011)
+ | 2 lines Update UPGRADE.txt and CHANGES files. Update
+ documentation files stating that deprecated modules are no longer
+ built by default. ........
+
+2011-07-15 08:19 +0000 [r328381] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Add SLA to skinny. Adds sublines to
+ skinny lines. Each subline can be attached to an SLA
+ station/trunk combo. Includes the following functionality: Callid
+ is persistent for both in/out calls on all skinny devices. Can
+ join, hold, resume. All sublines appear under a single line
+ button. See:
+ https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for
+ doc. (closes issue ASTERISK-17947) Review:
+ https://reviewboard.asterisk.org/r/1239/
+
+2011-07-15 00:23 +0000 [r328318-328344] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c,
+ include/asterisk/extconf.h, include/asterisk/pbx.h,
+ apps/app_queue.c: Merged revisions 328329 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10 ........
+ r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011)
+ | 2 lines Make hint watcher callback take const strings for
+ context and exten parameters. ........
+
+ * /, channels/chan_sip.c: Merged revisions 328317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328317 | rmudgett | 2011-07-14 18:28:49 -0500
+ (Thu, 14 Jul 2011) | 13 lines Merged revisions 328302 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328302 | rmudgett | 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011)
+ | 6 lines Missing SIP pvt and channel unlock in
+ sip_set_rtp_peer(). Regression introduced by -r326144. Add
+ missing SIP pvt and channel unlock in sip_set_rtp_peer().
+ ........ ................
+
+2011-07-14 20:28 +0000 [r328259] Leif Madsen <lmadsen@digium.com>
+
+ * funcs/func_speex.c, apps/app_playtones.c,
+ bridges/bridge_softmix.c, apps/app_alarmreceiver.c,
+ res/res_calendar_caldav.c, apps/app_ices.c, apps/app_exec.c,
+ channels/chan_iax2.c, res/res_pktccops.c, channels/chan_skinny.c,
+ pbx/pbx_ael.c, formats/format_h263.c, cdr/cdr_odbc.c,
+ cdr/cdr_manager.c, utils/refcounter.c, funcs/func_timeout.c,
+ formats/format_wav.c, apps/app_softhangup.c, codecs/codec_g726.c,
+ bridges/bridge_simple.c, funcs/func_cut.c, apps/app_talkdetect.c,
+ apps/app_db.c, funcs/func_callcompletion.c, funcs/func_channel.c,
+ funcs/func_iconv.c, pbx/pbx_config.c, res/res_odbc.c,
+ apps/app_voicemail.c, formats/format_sln.c,
+ apps/app_authenticate.c, apps/app_readexten.c,
+ res/res_phoneprov.c, apps/app_userevent.c, codecs/codec_gsm.c,
+ tests/test_func_file.c, apps/app_setcallerid.c,
+ res/res_config_odbc.c, funcs/func_audiohookinherit.c,
+ apps/app_osplookup.c, funcs/func_odbc.c, cel/cel_custom.c,
+ tests/test_utils.c, apps/app_mp3.c, res/res_timing_timerfd.c,
+ codecs/codec_resample.c, formats/format_h264.c,
+ apps/app_directory.c, formats/format_siren14.c,
+ tests/test_amihooks.c, res/res_config_pgsql.c,
+ funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c,
+ res/res_crypto.c, apps/app_originate.c, channels/chan_jingle.c,
+ apps/app_forkcdr.c, funcs/func_blacklist.c, apps/app_sms.c,
+ formats/format_g723.c, utils/extconf.c, tests/test_poll.c,
+ apps/app_stack.c, apps/app_verbose.c, utils/check_expr.c,
+ funcs/func_module.c, codecs/codec_adpcm.c, tests/test_event.c,
+ cdr/cdr_adaptive_odbc.c, apps/app_image.c,
+ formats/format_wav_gsm.c, utils/stereorize.c, pbx/pbx_loopback.c,
+ tests/test_time.c, funcs/func_shell.c, apps/app_skel.c,
+ channels/chan_alsa.c, apps/app_externalivr.c,
+ apps/app_milliwatt.c, formats/format_gsm.c, res/res_speech.c,
+ apps/app_dial.c, apps/app_page.c, apps/app_fax.c, utils/astman.c,
+ apps/app_disa.c, res/res_monitor.c, apps/app_waitforring.c,
+ addons/cdr_mysql.c, res/res_fax_spandsp.c,
+ res/res_timing_kqueue.c, apps/app_chanspy.c, apps/app_cdr.c,
+ channels/chan_unistim.c, funcs/func_base64.c,
+ channels/chan_multicast_rtp.c, funcs/func_md5.c,
+ apps/app_meetme.c, tests/test_gosub.c, funcs/func_sysinfo.c,
+ funcs/func_vmcount.c, res/res_musiconhold.c, cdr/cdr_radius.c,
+ apps/app_followme.c, res/res_config_sqlite.c,
+ apps/app_controlplayback.c, cdr/cdr_csv.c, formats/format_ilbc.c,
+ channels/chan_phone.c, funcs/func_enum.c, main/manager.c,
+ funcs/func_groupcount.c, tests/test_stringfields.c,
+ tests/test_locale.c, tests/test_devicestate.c,
+ funcs/func_frame_trace.c, funcs/func_aes.c, cdr/cdr_sqlite.c,
+ tests/test_astobj2.c, apps/app_ivrdemo.c, res/res_clioriginate.c,
+ apps/app_jack.c, apps/app_nbscat.c, res/res_calendar_icalendar.c,
+ codecs/codec_a_mu.c, tests/test_ast_format_str_reduce.c,
+ tests/test_dlinklists.c, res/res_convert.c, apps/app_waituntil.c,
+ pbx/pbx_lua.c, utils/astcanary.c, apps/app_queue.c,
+ channels/chan_oss.c, cdr/cdr_tds.c, channels/chan_usbradio.c,
+ apps/app_flash.c, apps/app_senddtmf.c, funcs/func_callerid.c,
+ addons/app_saycountpl.c, cel/cel_pgsql.c, apps/app_dahdibarge.c,
+ channels/chan_local.c, funcs/func_dialgroup.c,
+ tests/test_logger.c, apps/app_record.c, funcs/func_env.c,
+ funcs/func_strings.c, res/res_timing_dahdi.c,
+ apps/app_chanisavail.c, bridges/bridge_multiplexed.c,
+ res/res_rtp_multicast.c, cel/cel_odbc.c, channels/chan_dahdi.c,
+ pbx/pbx_spool.c, funcs/func_pitchshift.c, formats/format_pcm.c,
+ apps/app_dumpchan.c, main/http.c, res/res_clialiases.c,
+ res/res_calendar_exchange.c, res/res_ais.c, funcs/func_sprintf.c,
+ codecs/codec_g722.c, tests/test_expr.c, cel/cel_tds.c,
+ tests/test_app.c, utils/smsq.c, apps/app_morsecode.c,
+ formats/format_ogg_vorbis.c, tests/test_sched.c,
+ res/res_calendar_ews.c, apps/app_speech_utils.c,
+ tests/test_acl.c, apps/app_sendtext.c, funcs/func_cdr.c,
+ utils/hashtest2.c, utils/ael_main.c, apps/app_mixmonitor.c,
+ formats/format_g726.c, utils/streamplayer.c, res/res_calendar.c,
+ cel/cel_radius.c, channels/chan_vpb.cc, tests/test_heap.c,
+ addons/format_mp3.c, res/res_snmp.c, apps/app_dictate.c,
+ channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c,
+ res/res_jabber.c, funcs/func_uri.c, cel/cel_manager.c,
+ apps/app_minivm.c, res/res_realtime.c, res/res_config_ldap.c,
+ apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
+ codecs/codec_lpc10.c, apps/app_read.c, cdr/cdr_syslog.c,
+ codecs/codec_alaw.c, res/res_adsi.c, agi/eagi-test.c,
+ utils/conf2ael.c, tests/test_pbx.c, apps/app_channelredirect.c,
+ formats/format_vox.c, res/res_stun_monitor.c, tests/test_aoc.c,
+ pbx/pbx_dundi.c, funcs/func_devstate.c,
+ addons/res_config_mysql.c, funcs/func_rand.c,
+ apps/app_readfile.c, addons/chan_ooh323.c,
+ cdr/cdr_sqlite3_custom.c, /, apps/app_sayunixtime.c,
+ apps/app_test.c, res/res_http_post.c, res/res_smdi.c,
+ main/features.c, funcs/func_srv.c, apps/app_amd.c,
+ pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c,
+ formats/format_g719.c, channels/chan_bridge.c,
+ apps/app_privacy.c, apps/app_echo.c, codecs/codec_speex.c,
+ apps/app_saycounted.c, apps/app_dahdiras.c,
+ channels/chan_agent.c, funcs/func_math.c, res/res_ael_share.c,
+ apps/app_transfer.c, res/res_mutestream.c, apps/app_playback.c,
+ res/res_config_curl.c, channels/chan_misdn.c, funcs/func_curl.c,
+ tests/test_skel.c, apps/app_macro.c, apps/app_zapateller.c,
+ codecs/codec_ilbc.c, addons/app_mysql.c,
+ tests/test_substitution.c, utils/muted.c, utils/hashtest.c,
+ funcs/func_sha1.c, formats/format_siren7.c,
+ tests/test_security_events.c, funcs/func_config.c,
+ bridges/bridge_builtin_features.c, funcs/func_volume.c,
+ res/res_agi.c, apps/app_confbridge.c, addons/chan_mobile.c,
+ apps/app_parkandannounce.c, res/res_security_log.c,
+ cdr/cdr_custom.c, apps/app_while.c, res/res_rtp_asterisk.c,
+ funcs/func_dialplan.c, funcs/func_db.c, apps/app_festival.c,
+ res/res_limit.c, res/res_fax.c, apps/app_waitforsilence.c,
+ channels/chan_console.c, apps/app_getcpeid.c,
+ funcs/func_global.c, res/res_srtp.c, funcs/func_extstate.c,
+ tests/test_strings.c, res/res_timing_pthread.c,
+ apps/app_directed_pickup.c, channels/chan_h323.c,
+ cel/cel_sqlite3_custom.c, codecs/codec_ulaw.c,
+ channels/chan_nbs.c, formats/format_g729.c: Merged revisions
+ 328247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400
+ (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011)
+ | 6 lines Introduce <support_level> tags in MODULEINFO. This
+ change introduces MODULEINFO into many modules in Asterisk in
+ order to show the community support level for those modules. This
+ is used by changes committed to menuselect by Russell Bryant
+ recently (r917 in menuselect). More information about the support
+ level types and what they mean is available on the wiki at
+ https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
+ ........ ................
+
+2011-07-14 19:56 +0000 [r328208] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_monitor.c: Merged revisions 328207 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328207 | jrose | 2011-07-14 14:45:18 -0500
+ (Thu, 14 Jul 2011) | 13 lines Merged revisions 328205 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328205 | jrose | 2011-07-14 14:21:02 -0500 (Thu, 14 Jul 2011) |
+ 6 lines Monitor application arguments requirements fixed. Monitor
+ was requiring options in spite of no individual option on Monitor
+ being required. Review: https://reviewboard.asterisk.org/r/1320/
+ ........ ................
+
+2011-07-14 17:47 +0000 [r328163] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/dsp.c: Merged revisions 328162 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10 ........
+ r328162 | mnicholson | 2011-07-14 12:46:32 -0500 (Thu, 14 Jul
+ 2011) | 3 lines tune the v21 preamble detector to properly detect
+ the desired sequence of hits and misses ........
+
+2011-07-13 22:10 +0000 [r328121] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_mixmonitor.c: Merged revisions 328120 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ........ r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13
+ Jul 2011) | 15 lines Preserve sample rate quality of wideband
+ mixmonitor recordings. MixMonitor has the ability to record in
+ any file format Asterisk supports, but the quality of wideband
+ audio is not preserved. This is because regardless of the sample
+ rate the call is being recorded in, the audio is always
+ downsampled to 8khz and then upsampled to whatever wideband
+ format it is being written as. This patch resolves this by
+ requesting the audio from the audiohook in the signed linear
+ format closest to the sample rate of the format we are writing.
+ This fix is only possible for Asterisk 1.10 because audio hooks
+ in 1.8 are not capable of wideband audio. Review:
+ https://reviewboard.asterisk.org/r/1314/ ........
+
+2011-07-13 21:06 +0000 [r328079] Leif Madsen <lmadsen@digium.com>
+
+ * BUGS, UPGRADE-1.10.txt (added), UPGRADE.txt: Add UPGRADE-1.10.txt
+ file from UPGRADE.txt.
+
+2011-07-13 20:40 +0000 [r328075-328076] Russell Bryant <russell@digium.com>
+
+ * /: set 1.10 merge properties
+
+ * /: remove 1.8 merge properties
+
+2011-07-13 18:47 +0000 [r328016] Richard Mudgett <rmudgett@digium.com>
+
+ * /, configs/features.conf.sample: Merged revisions 328014 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328014 | rmudgett | 2011-07-13 13:46:38 -0500 (Wed, 13 Jul 2011)
+ | 1 line Add ATXFER_NULL_TECH note in features.conf.sample.
+ ........
+
+2011-07-12 23:02 +0000 [r327953] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/manager.c, /: Merged revisions 327950 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327950 | kpfleming | 2011-07-12 17:53:53 -0500 (Tue, 12 Jul
+ 2011) | 14 lines Correct double-free situation in manager output
+ processing. The process_output() function calls ast_str_append()
+ and xml_translate() on its 'out' parameter, which is a pointer to
+ an ast_str buffer. If either of these functions need to
+ reallocate the ast_str so it will have more space, they will free
+ the existing buffer and allocate a new one, returning the address
+ of the new one. However, because process_output only receives a
+ pointer to the ast_str, not a pointer to its caller's variable
+ holding the pointer, if the original ast_str is freed, the caller
+ will not know, and will continue to use it (and later attempt to
+ free it). (reported by jkroon on #asterisk-dev) ........
+
+2011-07-12 20:08 +0000 [r327891] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_directory.c: Merged revisions 327890 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r327890 | mnicholson | 2011-07-12 15:07:20 -0500 (Tue,
+ 12 Jul 2011) | 2 lines search in the current context for 'a' and
+ 'o' instead of 'default' ........
+
+2011-07-12 19:39 +0000 [r327889] Jason Parker <jparker@digium.com>
+
+ * Makefile, /: Merged revisions 327888 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327888 | qwell | 2011-07-12 14:38:44 -0500 (Tue, 12 Jul 2011) |
+ 1 line Fix uninstall target, so that modules dir gets cleared
+ again. ........
+
+2011-07-12 19:18 +0000 [r327856] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 327852 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r327852 | mjordan | 2011-07-12 14:10:34 -0500 (Tue, 12
+ Jul 2011) | 12 lines Added additional checks for mailbox /
+ password beginning with '*' character A bug existed such that if
+ a user entered a password with '*', and the extension 'a' did not
+ exist, an invalid mailbox would be created and the user
+ authenticated. The code was changed to prevent this from
+ occurring, and to prevent users from having mailboxes or
+ passwords defined that begin with the '*' character. (closes
+ issue ASTERISK-17443) Reported by: Kevin Scott Adams Tested by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/1316/
+ ........
+
+2011-07-12 15:38 +0000 [r327794] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * tests/test_substitution.c, /: Merged revisions 327793 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327793 | tilghman | 2011-07-12 10:35:46 -0500 (Tue, 12 Jul 2011)
+ | 14 lines Use 'printf' (POSIX issue 4) instead of 'echo -n', for
+ portability. The problem with using 'echo -n' is that it is not
+ portable. While BSD systems required that the '-n' option be
+ removed and interpreted, System V required that all strings
+ should be echoed with no interpretation of options. This
+ fundamental difference of behavior means that it is never
+ possible to use the '-n' flag to echo in tests which are meant to
+ be portable. In this case, on Mac OS X 10.6, the /bin/sh shell
+ builtin 'echo' uses the System V semantics of the command, and
+ thus the SHELL test failed on that platform.
+ http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16
+ ........
+
+2011-07-12 15:23 +0000 [r327769] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c, include/asterisk/dsp.h, main/dsp.c: do v21
+ detection instead of CED detection for the fax gateway
+
+2011-07-12 14:55 +0000 [r327749] David Vossel <dvossel@digium.com>
+
+ * main/bridging.c: Send video update frame to new video source in
+ follow_talker correctly.
+
+2011-07-12 14:40 +0000 [r327748] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_confbridge.c: Segfault on shutdown when confbridge is
+ active When undergoing a shutdown and channels are kicked out of
+ a bridge, a segfault occurs because ConfBridge tries to play
+ sounds on the bridge after the underlying channels have been
+ blown away due to the shutdown. (closes ASTERISK-18040) Review:
+ https://reviewboard.asterisk.org/r/1283/
+
+2011-07-11 20:06 +0000 [r327684] Matthew Nicholson <mnicholson@digium.com>
+
+ * tests/test_substitution.c: use printf instead of echo -n in
+ substitution test
+
+2011-07-11 19:49 +0000 [r327683] Terry Wilson <twilson@digium.com>
+
+ * /, include/asterisk/jingle.h, channels/chan_gtalk.c: Merged
+ revisions 327682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327682 | twilson | 2011-07-11 12:41:59 -0700 (Mon, 11 Jul 2011)
+ | 9 lines Update chan_gtalk to work with changed GMail-based
+ calls The messages sent by the GMail client have changed, but
+ include the old-style messages as well. This patch checks for
+ this case and uses the old-style offer. (closes issue
+ ASTERISK-18084) Review: https://reviewboard.asterisk.org/r/1312/
+ ........
+
+2011-07-11 18:44 +0000 [r327640] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/bridging.h, bridges/bridge_softmix.c,
+ main/bridging.c: Updates follow_talker video_mode in confbridge
+ application. follow_talker mode originally echoed the same video
+ stream to all participants. As the primary talker switched
+ around, the video stream would result in the talker seeing
+ themselves. Now the primary talker sees the last person who was
+ talking rather than themselves.
+
+2011-07-11 17:23 +0000 [r327469-327598] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: renamed fax_gateway_send_ced() to
+ fax_gateway_request_t38()
+
+ * res/res_fax.c: actually do something with the ced timeout, also
+ added more debug output
+
+ * res/res_fax.c: write silence on the channel during t.38
+ negotiation
+
+ * main/pbx.c, tests/test_substitution.c, /: Merged revisions 327512
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327512 | mnicholson | 2011-07-11 08:53:59 -0500 (Mon, 11 Jul
+ 2011) | 2 lines reset our buffer each iteration when doing
+ variable substitution ........
+
+ * res/res_fax.c: Delay sending an CED tone generated T.38 reinvite
+ to give the CED tone generating party time to send its own T.38
+ reinvite. Also don't forward frames through the gateway if we are
+ negotiating T.38.
+
+ * res/res_fax.c: fixed wording in a comment
+
+2011-07-11 10:57 +0000 [r327413] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, main/Makefile: Merged revisions 327411 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327411 | tzafrir | 2011-07-11 13:46:34 +0300 (ב', 11 יול 2011) |
+ 5 lines fix building the Debian armhf (HardFloat) port Fixes
+ http://buildd.debian-ports.org/status/fetch.php?pkg=asterisk&arch=armhf&ver=1%3A1.8.4.4~dfsg-2&stamp=1309935385
+ (Missing pthreads) ........
+
+2011-07-10 01:37 +0000 [r327359] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, configs/chan_ooh323.conf.sample: Full T.38
+ handshaking and fax detection Add full t.38 handshaking for
+ OOH323 that are required for newest T.38 gateway codes. Add fax
+ detection (cng tone, t38) and dialplan redirection to fax ext on
+ fax event detected. Add OOH323() function to set/get t38support
+ and faxdetect parameters. (closes issue ASTERISK-17754) Reported
+ by: irroot Patches: ooh323_faxdetect.patch uploaded by irroot
+ (license 52) issue19183-final.patch uploaded by may213 (license
+ 454) Tested by: may213, irroot Review:
+ https://reviewboard.asterisk.org/r/1174/
+
+2011-07-08 22:25 +0000 [r327246] Jason Parker <jparker@digium.com>
+
+ * main/stdtime, utils, codecs, utils/db1-ast/recno, apps, cel,
+ apps/confbridge, cdr, formats, codecs/gsm/src,
+ utils/db1-ast/hash, funcs, bridges, codecs/lpc10,
+ utils/db1-ast/db, codecs/g722, utils/db1-ast/mpool, main,
+ codecs/speex, channels/sip, pbx, res, res/ael, channels,
+ utils/db1-ast/btree: Add .o files to svn:ignore property, since
+ it's only ignored if locally configured to do so.
+
+2011-07-08 21:43 +0000 [r327212] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 327211 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011)
+ | 9 lines INVITE 403 Forbidden response always retransmits the
+ maximum times. Asterisk sends a 403 Forbidden response if
+ authentication fails for an INVITE as required. However, it
+ ignores the ACK and keeps retransmitting the response. * Made not
+ delete the to-tag in the dialog so the expected ACK can be
+ matched with the dialog and stop the retransmissions. ........
+
+2011-07-08 20:33 +0000 [r327116-327168] David Vossel <dvossel@digium.com>
+
+ * UPGRADE.txt, CHANGES: Adds entry in UPDATES.txt for removal of
+ formats/format_sln16.c. Fixes typo in CHANGES as well.
+
+ * CHANGES: Updates CHANGES log to reflect new slinear read/write
+ file interpreters.
+
+ * formats/format_sln.c, formats/format_sln16.c (removed): Support
+ for writing and reading raw slin files 8khz-192khz.
+
+ * formats/format_attr_silk.c (removed), formats/format_attr_celt.c
+ (removed), res/res_format_attr_silk.c (added),
+ res/res_format_attr_celt.c (added): Moves celt and silk format
+ attribute files into res folder. It was inconsistent to have the
+ silk and celt format attribute modules in the format file
+ interpreter folder.
+
+2011-07-08 19:54 +0000 [r327107] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, tests/test_substitution.c, /: Merged revisions 327106
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327106 | mnicholson | 2011-07-08 14:52:51 -0500 (Fri, 08 Jul
+ 2011) | 11 lines Reset our ast_str before passing it on to
+ dialplan function backends. It is possible for a dialplan backend
+ to not modify the given buffer or ast_str and still return
+ success. This causes any previous value stored in the buffer to
+ be used as if the new function call provided it. Some functions
+ also append to the given buffer assuming it is empty. The
+ test_substitution unit test has also been modified to detect this
+ problem. (closes issue ASTERISK-17878) ........
+
+2011-07-08 16:00 +0000 [r327045-327047] Russell Bryant <russell@digium.com>
+
+ * /, tests/test_netsock2.c: Merged revisions 327046 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r327046 | russell | 2011-07-08 11:00:05 -0500 (Fri, 08
+ Jul 2011) | 2 lines Fix an error and add more log message info to
+ help see why this fails on FreeBSD. ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 327044 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r327044 | russell | 2011-07-08 10:28:44 -0500 (Fri, 08
+ Jul 2011) | 2 lines Resolve some set-but-unused-variable
+ warnings. ........
+
+2011-07-08 01:26 +0000 [r327000] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /: Merged revisions 326985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r326985 | rmudgett | 2011-07-07 20:08:05 -0500 (Thu, 07 Jul 2011)
+ | 12 lines Some code cleanup in pbx.c * Mostly comment and format
+ changes. * ast_context_remove_extension_callerid() and
+ ast_add_extension_nolock() will write lock the found specific
+ context. * ast_context_find() will now tolerate a NULL name. *
+ Eliminated some inlined versions of find_context() and
+ find_context_locked(). ........
+
+2011-07-07 22:39 +0000 [r326943] Jason Parker <jparker@digium.com>
+
+ * include/asterisk/celt.h: I think reviewboard broke this. The
+ whole file was doubled.
+
+2011-07-07 22:17 +0000 [r326855-326904] David Vossel <dvossel@digium.com>
+
+ * formats/format_attr_celt.c (added): Adds the format_attr_celt
+ file which was also missing from the CELT pass through patch.
+
+ * include/asterisk/celt.h (added): Adds missing celt.h file from
+ celt pass-through support patch.
+
+ * CHANGES: Fixes spelling errors in CHANGES as well as adding a few
+ entries for CELT and confbridge.
+
+ * main/channel.c, main/format.c, res/res_rtp_asterisk.c,
+ main/frame.c, main/rtp_engine.c, channels/chan_sip.c,
+ include/asterisk/format.h, configs/codecs.conf.sample: Adds
+ pass-through support for codec CELT. This patch adds pass-through
+ support for CELT. CELT formats are defined in codecs.conf and can
+ be configured to any sample rate a CELT endpoint supports. This
+ patch also addresses a crash in channel.c resulting from a frame
+ list being freed incorrectly. This crash was discovered while
+ testing a CELT translator which had to split encoded audio into
+ multiple frames. The codec translator is not a part of this
+ patch, but may be contributed in the future. Review:
+ https://reviewboard.asterisk.org/r/1294/
+
+2011-07-07 19:20 +0000 [r326842] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, res/res_http_post.c: Merged revisions 326830 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r326830 | tilghman | 2011-07-07 14:17:19 -0500 (Thu, 07 Jul 2011)
+ | 1 line libgen.h is also needed on Darwin for basename(3)
+ ........
+
+2011-07-07 17:24 +0000 [r326782] David Vossel <dvossel@digium.com>
+
+ * configs/confbridge.conf.sample,
+ apps/confbridge/include/confbridge.h, apps/app_confbridge.c,
+ apps/confbridge/conf_config_parser.c: Updates confbridge.conf
+ video documentation and adds dtmf action for releasing video src.
+
+2011-07-07 16:50 +0000 [r326750] Terry Wilson <twilson@digium.com>
+
+ * utils/astdb2sqlite3.c, main/db.c: Use older functions out of
+ deference to older distros
+
+2011-07-07 16:18 +0000 [r326694] Jonathan Rose <jrose@digium.com>
+
+ * res/res_config_odbc.c, /: Merged revisions 326689 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r326689 | jrose | 2011-07-07 11:04:51 -0500 (Thu, 07 Jul
+ 2011) | 10 lines res_odbc patch by tilghman to fix integers with
+ null values Addresses some improper sql statements in res_odbc
+ that would cause an update to fail on realtime peers due to
+ trying to set as "(NULL)" rather than an actual NULL. (closes
+ issue #1922STERISK-17791) Reported by: marcelloceschia Patches:
+ 20110505__issue19223.diff.txt uploaded by tilghman (license 14)
+ ........
+
+2011-07-07 15:28 +0000 [r326682-326684] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 326683 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r326683 | mnicholson | 2011-07-07 10:28:25 -0500 (Thu, 07 Jul
+ 2011) | 3 lines use sips: or sip: depending on the transport in
+ use when building reply digest URIs ........
+
+ * /, channels/chan_sip.c: Merged revisions 326681 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r326681 | mnicholson | 2011-07-07 10:25:49 -0500 (Thu, 07 Jul
+ 2011) | 3 lines make the uri parameter used in reply digests more
+ standards compliant in certain cases by prepending "sip:" or
+ "sips:" to it ........
+
+2011-07-07 09:49 +0000 [r326636] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * contrib/scripts/live_ast: live_ast: valgrind: run asterisk under
+ valgrind Adds a new sub-command, "valgrind" to live_ast. It runs
+ asterisk under valgrind. The extra command-line parameters are
+ passed to Asterisk as usual, and parameters to valgrind are
+ passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review:
+ https://reviewboard.asterisk.org/r/1109/
+
+2011-07-06 20:58 +0000 [r326589] Terry Wilson <twilson@digium.com>
+
+ * utils/db1-ast/btree/bt_open.c, utils/db1-ast/hash/hash_log2.c,
+ utils/db1-ast/hash/hsearch.c, utils/db1-ast/btree/bt_page.c,
+ utils/db1-ast/hash/page.h, utils/db1-ast/mpool, configure,
+ utils/db1-ast/btree/extern.h, utils/db1-ast/include/db.h,
+ main/db.c, utils/db1-ast/btree/bt_seq.c,
+ utils/db1-ast/recno/recno.h, main/Makefile,
+ utils/db1-ast/btree/bt_utils.c, utils/db1-ast/recno/rec_seq.c,
+ configure.ac, utils/db1-ast/btree/bt_close.c, CHANGES,
+ utils/db1-ast/hash/search.h, utils/db1-ast/hash/README,
+ utils/db1-ast/recno/rec_open.c, utils/db1-ast/hash/hash_bigkey.c,
+ utils/db1-ast/recno/rec_delete.c, Makefile,
+ utils/db1-ast/include, utils/db1-ast/hash/hash_buf.c,
+ utils/db1-ast/db, utils/db1-ast/libdb.map,
+ utils/db1-ast/include/ndbm.h, utils/db1-ast/include/compat.h,
+ utils/db1-ast/mpool/mpool.c, utils/db1-ast/btree/bt_debug.c,
+ main/asterisk.c, utils/db1-ast (added),
+ utils/db1-ast/btree/bt_split.c, utils, utils/db1-ast/recno,
+ utils/db1-ast/btree/bt_delete.c,
+ utils/db1-ast/include/circ-queue.h, tests/test_db.c,
+ utils/db1-ast/Makefile, utils/db1-ast/hash/extern.h,
+ utils/db1-ast/recno/rec_search.c, utils/db1-ast/btree/bt_get.c,
+ utils/db1-ast/hash/hash.c, utils/db1-ast/btree/btree.h,
+ utils/db1-ast/db/db.c, utils/db1-ast/hash/hash.h,
+ utils/db1-ast/include/mpool.h, utils/db1-ast/recno/rec_get.c,
+ utils/db1-ast/hash/hash_func.c, utils/utils.xml,
+ utils/astdb2sqlite3.c (added), utils/db1-ast/btree/bt_overflow.c,
+ UPGRADE.txt, utils/db1-ast/btree/bt_conv.c,
+ utils/db1-ast/btree/bt_search.c, utils/db1-ast/btree/bt_put.c,
+ utils/db1-ast/recno/rec_utils.c, utils/Makefile,
+ utils/db1-ast/hash/hash_page.c, utils/db1-ast/hash,
+ utils/db1-ast/mpool/README, utils/db1-ast/hash/ndbm.c,
+ main/db1-ast (removed), utils/db1-ast/recno/rec_close.c,
+ utils/db1-ast/recno/rec_put.c, utils/db1-ast/recno/extern.h,
+ utils/db1-ast/btree: Replace Berkeley DB with SQLite 3 There were
+ some bugs in the very ancient version of Berkeley DB that
+ Asterisk used. Instead of spending the time tracking down the
+ bugs in the Berkeley code we move to the much better documented
+ SQLite 3. Conversion of the old astdb happens at runtime by
+ running the included astdb2sqlite3 utility. The ast_db API with
+ SQLite 3 backend should behave identically to the old Berkeley
+ backend, but in the future we could offer a much more robust
+ interface. We do not include the SQLite 3 library in the source
+ tree, but instead rely upon the distribution-provided libraries.
+ SQLite is so ubiquitous that this should not place undue burden
+ on administrators.
+
+2011-07-06 17:39 +0000 [r326485-326544] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes newlines from being stripped from out
+ of dialog sip MESSAGES.
+
+ * /, res/res_timing_timerfd.c: Merged revisions 326484 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r326484 | dvossel | 2011-07-06 10:26:49 -0500 (Wed, 06
+ Jul 2011) | 10 lines Reverts fix for timerfd locking issue. jrose
+ discovered a performance issue with this fix that prevents his
+ analog phones from working when using timerfd as a timing source.
+ Until it is understood what is causing this performance problem,
+ this patch is being reverted. ........
+
+2011-07-05 22:11 +0000 [r326412] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * channels/chan_jingle.c, channels/chan_dahdi.c,
+ funcs/func_speex.c, /, channels/chan_sip.c, codecs/codec_speex.c,
+ funcs/func_aes.c, pbx/pbx_dundi.c, channels/chan_gtalk.c,
+ apps/app_queue.c, channels/chan_iax2.c, res/res_jabber.c,
+ apps/app_stack.c, channels/chan_mgcp.c, apps/app_voicemail.c:
+ Merged revisions 326411 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011)
+ | 14 lines Add the attribute "type" to each "<use>" for
+ menuselect. This matters only when autoconf fails to detect that
+ weak linking is supported. External optional dependencies will
+ become optional in both cases, as they are removed at compile
+ time when not detected. However, runtime-optional modules are
+ made mandatory when weak linking is not found. This change
+ affects only the external optional dependencies; previously, they
+ were incorrectly required when weak linking support was not
+ detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt
+ by tilghman (License #5003) Tested by: iasgoscouk ........
+
+2011-07-05 20:25 +0000 [r326368] Kinsey Moore <kmoore@digium.com>
+
+ * contrib/scripts/file.convert.sh (added): Prompt conversion script
+ Several variables in the script control which files are converted
+ and the source and destination formats. Patch-by: Trey Blancher
+ <support@digium.com> (closes AST-560)
+
+2011-07-05 17:35 +0000 [r326321] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
+ revisions 326291 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r326291 | rmudgett | 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011)
+ | 23 lines Used auth= parameter freed during "sip reload" causes
+ crash. If you use the auth= parameter and do a "sip reload" while
+ there is an ongoing call. The peer->auth data points to free'd
+ memory. The patch does several things: 1) Puts the authentication
+ list into an ao2 object for reference counting to fix the
+ reported crash during a SIP reload. 2) Converts the
+ authentication list from open coding to AST list macros. 3) Adds
+ display of the global authentication list in "sip show settings".
+ (closes issue ASTERISK-17939) Reported by: wdoekes Patches:
+ jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Review: https://reviewboard.asterisk.org/r/1303/ JIRA
+ SWP-3526 ........
+
+2011-07-05 16:46 +0000 [r326267] Mark Murawki <markm@intellasoft.net>
+
+ * main/manager.c, CHANGES: New feature: AMI Action FilterAdd This
+ adds a new action, FilterAdd to the manager interface that allows
+ control over event filters for the current session (closes issue
+ ASTERISK-16795) Reported by: kobaz Tested by: kobaz,loloski
+
+2011-07-05 13:38 +0000 [r326210] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/file.c: Merged revisions 326209 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r326209 | mjordan | 2011-07-05 08:23:57 -0500 (Tue, 05 Jul 2011)
+ | 7 lines Updated filestream destructor to block until move is
+ complete when cache is used When a cache directory is used, the
+ process is forked and a mv command is executed to move the
+ temporary file to the permanent location. This caused issues with
+ voicemail, where a race condition occurred when the parent
+ expected the file to be in the permanent location prior to the mv
+ command completing. The parent process is now blocked until the
+ mv command completes. (closes issue ASTERISK-17724) Reported by:
+ Adiren P. Tested by: mjordan ........
+
+2011-07-01 21:11 +0000 [r326145] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 326144 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r326144 | rmudgett | 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011)
+ | 16 lines Better way to get chan and pvt lock for issue
+ ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431 deadlock
+ fix for sip_set_udptl_peer() and sip_set_rtp_peer(). * Lock the
+ channels in the defined order and avoid the need for a deadlock
+ avoidance loop. * Lock the channel before getting the pointer to
+ the private structure to be sure that the pointer will not change
+ due to a masquerade or channel hangup. * To preserve sanity,
+ check that chan and p->owner are the same. (Pointer rearangements
+ should not happen without the protection of locks because bad
+ things tend to happen otherwise.) ........
+
+2011-07-01 16:36 +0000 [r326056-326101] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * CHANGES: Change CHANGES move the commits to the right place
+ r296249 r318141 Application changes
+
+ * CHANGES: Change CHANGES move the commits to the right place in
+ the file missed in review
+
+2011-07-01 12:45 +0000 [r326006] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c, res/res_fax_spandsp.c: updated irroots info for
+ the authors section
+
+2011-06-30 21:05 +0000 [r325937] David Vossel <dvossel@digium.com>
+
+ * channels/chan_bridge.c: Fixes warning message caused by
+ confbridge playback chan not being answered.
+
+2011-06-30 20:47 +0000 [r325936] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 325935 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011)
+ | 11 lines Misc minor changes in chan_sip. * Add load failure
+ exit if primary SIP container(s) could not get created in
+ chan_sip.c:load_module(). * Removed a redundant static prototype.
+ * Some typos. * Some whitespace. ........
+
+2011-06-30 20:33 +0000 [r325931] David Vossel <dvossel@digium.com>
+
+ * configs/confbridge.conf.sample,
+ apps/confbridge/include/confbridge.h,
+ include/asterisk/bridging.h, include/asterisk/dsp.h,
+ bridges/bridge_softmix.c, apps/app_confbridge.c, CHANGES,
+ main/bridging.c, main/dsp.c, apps/app_voicemail.c,
+ apps/confbridge/conf_config_parser.c: Video support for
+ ConfBridge. Review: https://reviewboard.asterisk.org/r/1288/
+
+2011-06-30 20:24 +0000 [r325900] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 325877 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r325877 | mjordan | 2011-06-30 15:09:48 -0500 (Thu, 30
+ Jun 2011) | 9 lines Patched voicemail user option for emailbody /
+ emailsubject Incorporated changes per ASTERISK-16795; updated
+ unit tests to check for vmu->emailbody / vmu->emailsubject
+ (closes issue ASTERISK-16795) Reported by: mdeneen Tested by:
+ mjordan ........
+
+2011-06-30 19:31 +0000 [r325864] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 325821 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r325821 | jrose | 2011-06-30 14:17:32 -0500 (Thu, 30 Jun
+ 2011) | 10 lines Fixes an issue with Music on Hold classes losing
+ files in playlist when realtime is used. The bug occurs rather
+ intermittently and I relied on the reporters to test the patch.
+ After a sanity check and some testing, I'm giving it an OK.
+ (closes issue ASTERISK-17875) Reported by: David Cunningham
+ Patches: res_musiconhold.c.mohrt17875_v1 uploaded by Igor
+ Goncharovsky (license #5009) ........
+
+2011-06-30 18:22 +0000 [r325815-325816] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c, include/asterisk/res_fax.h, CHANGES,
+ res/res_fax_spandsp.c: Fax gateway functionality (i.e.
+ translating between a T.30 terminal and a T.38 terminal). Can be
+ enabled on a channel by setting FAXOPT(gateway)=yes in the
+ dialplan. Big thanks to irroot for porting this code to use the
+ framehooks api.
+
+ * main/frame.c: copy all flags on asterisk frames instead of just
+ the timing flag
+
+2011-06-29 21:50 +0000 [r325741] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
+ revisions 325740 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r325740 | kmoore | 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) |
+ 7 lines chan_sip: cleanup from the introduction of ast_str Remove
+ the length field from sip_req and sip_pkt in chan_sip since they
+ are redundant (ast_str holds its own length) and refactor the
+ necessary functions. Review:
+ https://reviewboard.asterisk.org/r/1281/ ........
+
+2011-06-29 19:02 +0000 [r325674] David Vossel <dvossel@digium.com>
+
+ * /, res/res_timing_timerfd.c: Merged revisions 325673 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r325673 | dvossel | 2011-06-29 13:59:33 -0500 (Wed, 29
+ Jun 2011) | 6 lines Fixes timerfd locking issue. (closes
+ ASTERISK-17867, ASTERISK-17415) Patches: fix uploaded by kobaz
+ Review: https://reviewboard.asterisk.org/r/1255/ ........
+
+2011-06-29 18:18 +0000 [r325611-325616] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 325614 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r325614 | rmudgett | 2011-06-29 13:16:45 -0500 (Wed, 29 Jun 2011)
+ | 5 lines Fixed some error exit cleanup in app_queue.c. * Fixed
+ error exit cleanup in app_queue.c copy_rules() and
+ reload_queue_rules(). ........
+
+ * /, apps/app_queue.c: Merged revisions 325610 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r325610 | rmudgett | 2011-06-29 13:05:15 -0500 (Wed, 29 Jun 2011)
+ | 18 lines Response to QueueRule manager command does not contain
+ ActionID if it was specified. * Add ActionID support as
+ documented for the QueueRule AMI action. * Remove documentation
+ for ActionID with the Queues AMI action. The output does not
+ follow normal AMI response output and there is no place to put an
+ ActionID header. (closes issue AST-602) Reported by: Vlad
+ Povorozniuc Patches: jira_ast_602_v1.8.patch (license #5621)
+ patch uploaded by rmudgett Tested by: Vlad Povorozniuc, rmudgett
+ Review: https://reviewboard.asterisk.org/r/1295/ JIRA SWP-3575
+ ........
+
+2011-06-29 16:19 +0000 [r325538-325547] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 325545 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r325545 | mnicholson | 2011-06-29 11:18:39 -0500 (Wed, 29 Jun
+ 2011) | 2 lines make framehooks prevent native bridging (for real
+ this time) ........
+
+ * apps/app_dial.c, main/rtp_engine.c, /: Merged revisions 325537
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r325537 | mnicholson | 2011-06-29 10:34:47 -0500 (Wed, 29 Jun
+ 2011) | 2 lines don't do native/remote bridging if a framehook is
+ active on the channel ........
+
+2011-06-29 06:39 +0000 [r325483] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * configs/queues.conf.sample, UPGRADE.txt, CHANGES,
+ apps/app_queue.c: Commit "distrotech" app_queue changes to Trunk
+ * Added general option negative_penalty_invalid default off. when
+ set members are seen as invalid/logged out when there penalty is
+ negative. for realtime members when set remove from queue will
+ set penalty to -1. * Added queue option autopausedelay when
+ autopause is enabled it will be delayed for this number of
+ seconds since last successful call if there was no prior call the
+ agent will be autopaused immediately. * Added member option
+ ignorebusy this when set and ringinuse is not will allow per
+ member control of multiple calls as ringinuse does for the Queue.
+ - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for
+ realtime members - QUEUE_MEMBER is now R/W supporting setting
+ paused, ignorebusy and penalty. (closes issue ASTERISK-17421)
+ (closes issue ASTERISK-17391) Reported by: irroot Tested by:
+ irroot, jrose Review: https://reviewboard.asterisk.org/r/1119/
+
+2011-06-28 21:51 +0000 [r325417] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 325416 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r325416 | kpfleming | 2011-06-28 16:50:43 -0500 (Tue, 28 Jun
+ 2011) | 3 lines Fix random misspelling noticed on asterisk-users.
+ ........
+
+2011-06-28 20:32 +0000 [r325345] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 325339 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r325339 | dvossel | 2011-06-28 15:31:00 -0500 (Tue, 28 Jun 2011)
+ | 4 lines Fixes locking inversion caused by holding sip pvt lock
+ during async_goto. (closes ASTERISK-17352) ........
+
+2011-06-28 17:38 +0000 [r325213] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 325212 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r325212 | rmudgett | 2011-06-28 12:30:16 -0500 (Tue, 28
+ Jun 2011) | 7 lines Use the device name and not the channel name
+ to initialize the device state. Correct ASTERISK-11323
+ implementation as I don't see how it ever worked as claimed when
+ it used the channel name and not the device name. (issue
+ ASTERISK-11323) ........
+
+2011-06-28 16:04 +0000 [r325153] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 325152 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r325152 | jrose | 2011-06-28 10:46:29 -0500 (Tue, 28 Jun
+ 2011) | 5 lines Fixes moh reload breaking custom mode moh classes
+ when the config file is untouched (closes issue ASTERISK-17730)
+ Reported by: sdolloff ........
+
+2011-06-28 15:34 +0000 [r325151] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes issue with video and text not being
+ reinvited correctly with directmedia If a SDP does not modify the
+ session, we ignore it. However, we were defaulting no text and
+ video support to true before checking to see if the sdp modified
+ anything or not. This would result in process_sdp ignoring an sdp
+ but removing video and text from the call during direct media
+ reinvites.
+
+2011-06-28 15:12 +0000 [r325092] Leif Madsen <lmadsen@digium.com>
+
+ * /, build_tools/prep_tarball: Merged revisions 325091 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r325091 | lmadsen | 2011-06-28 10:12:00 -0500 (Tue, 28
+ Jun 2011) | 1 line Remove line from prep_tarball that kills
+ mkrelease. ........
+
+2011-06-28 00:07 +0000 [r325046] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't forget to build the Via when sending
+ MESSAGE
+
+2011-06-27 16:32 +0000 [r324961] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/asterisk.c: Merged revisions 324955 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324955 | tilghman | 2011-06-27 11:30:50 -0500 (Mon, 27 Jun 2011)
+ | 5 lines Save and restore errno from within signal handlers.
+ This is recommended by the POSIX standard, as well as by the
+ sigaction(2) manpage for various platforms that we support (e.g.
+ Mac OS X). ........
+
+2011-06-27 15:38 +0000 [r324915] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 324914 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324914 | rmudgett | 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011)
+ | 21 lines When subscribing MWI to an unsolicited mailbox the
+ first notification is incorrect. A remote peer subscribed to MWI
+ with the unsolicited option and a local phone subscribed to the
+ remote mailbox. The notify message-summary events are sent
+ correctly except for the first one when subscribing, which will
+ always be 0. This means the phone MWI indicator will be wrong
+ until the mailbox read/unread count changes and the event is
+ fired. Looks like this is a regression from ASTERISK-16149. * Fix
+ the logic to check the cache and if allowed then fallback to
+ manually counting mailbox messages. (closes issue ASTERISK-17997)
+ Reported by: rsw686 Patches: jira_asterisk_17997_v1.8.patch
+ (license #5621) uploaded by rmudgett Tested by: rsw686 JIRA
+ SWP-3551 ........
+
+2011-06-24 20:50 +0000 [r324850] Richard Mudgett <rmudgett@digium.com>
+
+ * /, pbx/pbx_config.c: Merged revisions 324849 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324849 | rmudgett | 2011-06-24 15:46:01 -0500 (Fri, 24 Jun 2011)
+ | 15 lines Syntax errors in dialplan do not display the file
+ name. When issuing the CLI command "dialplan reload" syntax
+ errors and warnings are displayed on the console. The offending
+ line number is displayed on the console, but the file name is not
+ displayed. Errors caught in main/config.c do display the file
+ name. (closes issue ASTERISK-17985) Reported by: ulogic Patches:
+ pbx_config.patch uploaded by ulogic (License #5685) modified
+ format Tested by: rmudgett JIRA SWP-3554 ........
+
+2011-06-24 16:50 +0000 [r324769] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/logger.h, /: Merged revisions 324768 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324768 | jrose | 2011-06-24 11:48:06 -0500 (Fri, 24 Jun 2011) |
+ 11 lines DTMF wasn't being logged on connected consoles when
+ enabled in logger.conf Previously in order for DTMF to be logged
+ in a connected console session, the user would have to do logger
+ set channel DTMF on. This corrects that so that it is on by
+ default. This issue was caused by an off by one error incurred by
+ a logger level count of 6 in logger.h where it should have been
+ 7. (closes issue: ASTERISK-17974) Reported by: Luke H ........
+
+2011-06-23 18:56 +0000 [r324708-324709] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_confbridge.c: ConfBridge: redundant code cleanup There
+ is no reason to clean up features twice. Review:
+ https://reviewboard.asterisk.org/r/1279/
+
+ * /, channels/chan_sip.c: Merged revisions 324678 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r324678 | kmoore | 2011-06-23 13:29:17 -0500
+ (Thu, 23 Jun 2011) | 11 lines Merged revisions 324643 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) |
+ 4 lines Addresses AST-2011-008, memory corruption and remote
+ crash in SIP driver. AST-2011-008 ........ ................
+
+2011-06-23 18:31 +0000 [r324664-324689] David Vossel <dvossel@digium.com>
+
+ * /, channels/sip/reqresp_parser.c: Merged revisions 324685 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324685 | dvossel | 2011-06-23 13:31:00 -0500 (Thu, 23 Jun 2011)
+ | 8 lines Fixes sip crash when calling remove_uri_parameters with
+ NULL AST-2011-009 (closes issue ASTERISK-18017) Reported by:
+ jaredmauch ........
+
+ * /, main/features.c, channels/chan_iax2.c,
+ include/asterisk/frame.h: Merged revisions 324652 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r324652 | dvossel | 2011-06-23 13:23:21 -0500
+ (Thu, 23 Jun 2011) | 20 lines Merged revisions 324634 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500
+ (Thu, 23 Jun 2011) | 13 lines Merged revisions 324627 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011)
+ | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver
+ Thanks to twilson for identifying the issue and providing the
+ patches. AST-2011-010 ........ ................ ................
+
+2011-06-23 03:16 +0000 [r324558] Terry Wilson <twilson@digium.com>
+
+ * /, tests/test_netsock2.c: Merged revisions 324557 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r324557 | twilson | 2011-06-22 22:10:38 -0500 (Wed, 22
+ Jun 2011) | 5 lines Remove tests for parsing address with invalid
+ port getaddrinfo on OS X returns with EAI_NONAME error when
+ passed a port greater than 65535. Linux throws no error, so
+ remove the tests for now. ........
+
+2011-06-22 19:17 +0000 [r324495] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 324491 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324491 | rmudgett | 2011-06-22 14:16:29 -0500 (Wed, 22 Jun 2011)
+ | 1 line Use correct variable for text SRTP media. ........
+
+2011-06-22 19:12 +0000 [r324487] Terry Wilson <twilson@digium.com>
+
+ * main/netsock2.c, /, channels/chan_sip.c,
+ include/asterisk/netsock2.h, tests/test_netsock2.c (added):
+ Merged revisions 324484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011)
+ | 20 lines Stop sending IPv6 link-local scope-ids in SIP messages
+ The idea behind the patch listed below was used, but in a more
+ targeted manner. There are now address stringification functions
+ for addresses that are meant to be sent to a remote party.
+ Link-local scope-ids only make sense on the machine from which
+ they originate and so are stripped in the new functions. There is
+ also a host sanitization function added to chan_sip which is used
+ for when peer and dialog tohost fields or sip_registry hostnames
+ are used to craft a SIP message. Also added are some basic unit
+ tests for netsock2 address parsing. (closes issue ASTERISK-17711)
+ Reported by: ch_djalel Patches:
+ asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel
+ (license 1251) Review: https://reviewboard.asterisk.org/r/1278/
+ ........
+
+2011-06-22 18:45 +0000 [r324480-324482] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 324481 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 Also fixed a
+ reference leak in an error path in sip_msg_send(). ........
+ r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011)
+ | 19 lines Timout or error on INFO or MESSAGE transaction causes
+ call to be lost. When exchanging INFO messages within a call, 4xx
+ error causes the call to be disconnected although RFC 2976
+ explicitly states that such transactions do not modify the state
+ of the dialog. When exchanging MESSAGE messages within a call,
+ 4xx error causes the call to be disconnected. To provide least
+ surprise, we should not disconnect the call since a MESSAGE is
+ like INFO in this case. (Implied by RFC 3428 Section 2) (closes
+ issue ASTERISK-17901) Reported by: neutrino88 Review:
+ https://reviewboard.asterisk.org/r/1257/ Review:
+ https://reviewboard.asterisk.org/r/1258/ JIRA SWP-3486 ........
+
+ * /, channels/chan_sip.c: Merged revisions 324479 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324479 | rmudgett | 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011)
+ | 1 line Comments and whitespace in chan_sip.c ........
+
+2011-06-21 21:55 +0000 [r324365-324422] David Vossel <dvossel@digium.com>
+
+ * apps/app_confbridge.c: Fixes issue with channel write format
+ being incorrectly restored when MOH is used in confbridge.
+
+ * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 324364
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324364 | dvossel | 2011-06-21 15:11:52 -0500 (Tue, 21 Jun 2011)
+ | 10 lines Fixes locking inversion issue in ast_async_goto()
+ During this function we can not hold the "chan" lock while doing
+ the masquerade, the explicit goto on the tmp chan, or the channel
+ alloc. Instead we need to get the channel lock, store off
+ information about the channel that we need, and then let the
+ channel lock go for the remainder of the function. Review:
+ https://reviewboard.asterisk.org/r/1275/ ........
+
+2011-06-21 16:06 +0000 [r324304] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_confbridge.c: ConfBridge does not handle hangup properly
+ When playing back a prompt to a channel, confbridge neglects to
+ check for hangup events causing lockup condititions for hangups
+ that occur before actually joining the conference. This change
+ ensures that the user is removed from the conference in the event
+ of a premature hangup. Review:
+ https://reviewboard.asterisk.org/r/1277/
+
+2011-06-21 15:49 +0000 [r324302] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes issue with finding correct extension
+ when message context is used.
+
+2011-06-20 18:13 +0000 [r324242] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/queuerules.conf.sample: Merged revisions 324241 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324241 | lmadsen | 2011-06-20 13:12:32 -0500 (Mon, 20 Jun 2011)
+ | 2 lines Remove extra 'the'. Reported by Vlad Povorozniuc
+ ........
+
+2011-06-20 17:34 +0000 [r324238] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 324237 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324237 | twilson | 2011-06-20 12:33:07 -0500 (Mon, 20 Jun 2011)
+ | 12 lines Ignore media offers with a port of 0 Section 5.1 of
+ RFC3264 states: A port number of zero in the offer indicates that
+ the stream is offered but MUST NOT be used. (closes issue
+ ASTERISK-17845) Reported by: jacco Patches: issue19281_2.patch
+ uploaded by jacco (license 1277) Tested by: jacco, twilson
+ ........
+
+2011-06-17 18:52 +0000 [r324177-324179] Leif Madsen <lmadsen@digium.com>
+
+ * main/manager.c, /: Merged revisions 324178 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324178 | lmadsen | 2011-06-17 14:51:16 -0400 (Fri, 17 Jun 2011)
+ | 2 lines Add Username and Secret fields to manager Login action.
+ Pointed out by Vlad Povorozniuc ........
+
+ * /, apps/app_meetme.c: Merged revisions 324176 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324176 | lmadsen | 2011-06-17 14:38:40 -0400 (Fri, 17 Jun 2011)
+ | 2 lines Fix typo in documentation. Pointed out by Vlad
+ Povorozniuc ........
+
+2011-06-17 18:23 +0000 [r324175] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 324174 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r324174 | rmudgett | 2011-06-17 13:23:19 -0500 (Fri, 17
+ Jun 2011) | 5 lines Add header string to libpri debug output. Add
+ header string to libpri debug output so the libpri output can be
+ found/extracted easier from huge debug trace files. ........
+
+2011-06-17 15:32 +0000 [r324131] Leif Madsen <lmadsen@digium.com>
+
+ * main/pbx.c, /: Merged revisions 324115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r324115 | lmadsen | 2011-06-17 11:14:54 -0400 (Fri, 17 Jun 2011)
+ | 3 lines Fix grammar in documentation for Goto() and GotoIf()
+ (closes issue ASTERISK-18023) Reported by: Tim Osman ........
+
+2011-06-16 22:49 +0000 [r324050] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, channels/chan_local.c, /, channels/chan_sip.c,
+ include/asterisk/channel.h: Merged revisions 324048 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16
+ Jun 2011) | 8 lines Lock the channel before calling the setoption
+ callback The channel needs to be locked before calling these
+ callback functions. Also, sip_setoption needs to lock the pvt and
+ a check p->rtp is non-null before using it. Review:
+ https://reviewboard.asterisk.org/r/1220/ ........
+
+2011-06-16 18:13 +0000 [r323991] Richard Mudgett <rmudgett@digium.com>
+
+ * /, tests/test_event.c: Merged revisions 323990 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r323990 | rmudgett | 2011-06-16 13:12:32 -0500 (Thu, 16 Jun 2011)
+ | 5 lines The test_event unit test is occasionally failing. Wait
+ for the special posted event to process before adding a new
+ subscription. ........
+
+2011-06-16 15:59 +0000 [r323673-323933] Terry Wilson <twilson@digium.com>
+
+ * Makefile, /: Merged revisions 323932 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r323932 | twilson | 2011-06-16 10:58:22 -0500 (Thu, 16 Jun 2011)
+ | 4 lines Don't assume ASTDBDIR exists It most likely doesn't on
+ FreeBSD ........
+
+ * /, tests/test_db.c: Merged revisions 323866 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r323866 | twilson | 2011-06-15 15:03:58 -0500 (Wed, 15 Jun 2011)
+ | 2 lines Remove now-useless cast of ARRAY_LEN ........
+
+ * include/asterisk/utils.h, /: Merged revisions 323863 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r323863 | twilson | 2011-06-15 14:58:18 -0500 (Wed, 15
+ Jun 2011) | 2 lines Make ARRAY_LEN() return the same type on x86
+ and x86_64 systems ........
+
+ * /, tests/test_db.c: Merged revisions 323859 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r323859 | twilson | 2011-06-15 14:45:20 -0500 (Wed, 15 Jun 2011)
+ | 2 lines Fix more ARRAY_LEN format string issues ........
+
+ * /, main/features.c: Merged revisions 323754 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r323754 | twilson | 2011-06-15 13:21:52 -0500
+ (Wed, 15 Jun 2011) | 23 lines Merged revisions 323733 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r323733 | twilson | 2011-06-15 13:13:00 -0500
+ (Wed, 15 Jun 2011) | 16 lines Merged revisions 323732 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011)
+ | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a
+ recent DTMF change. This patch makes sure that dynamic features
+ are also checked when deciding whether or not to pass DTMF
+ through or store it for interpreting. (closes issue
+ ASTERISK-17914) Reported by: vrban ........ ................
+ ................
+
+ * /, tests/test_db.c: Merged revisions 323672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r323672 | twilson | 2011-06-15 10:09:51 -0700 (Wed, 15 Jun 2011)
+ | 5 lines Cast ARRAY_LEN to size_t for ast_logging 32-bit and
+ 64-bit machines return different types for ARRAY_LEN(), so cast
+ it before using in a format string. ........
+
+2011-06-15 16:49 +0000 [r323671] Richard Mudgett <rmudgett@digium.com>
+
+ * /, tests/test_event.c, main/event.c: Merged revisions
+ 323669-323670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r323669 | rmudgett | 2011-06-15 11:43:18 -0500 (Wed, 15 Jun 2011)
+ | 21 lines [regression] Voicemail MWI is no longer sent. When
+ leaving a voicemail, the MWI message is never sent. The same
+ thing happens when checking a voicemail and marking it as read.
+ If you restart Asterisk, everything comes up at that state
+ correctly, but changes to the messages in voicemail causes the
+ light to not be set appropriately. Very easy to reproduce. * Made
+ ast_event_check_subscriber() return TRUE if there are ANY
+ subscribers to an event type when there are no restricting ie
+ values passed. This allows an event being queued to be queued.
+ (closes issue ASTERISK-18002) Reported by: lmadsen Tested by:
+ lmadsen, irroot Patches: jira_asterisk_18002_v1.8.patch uploaded
+ by rmudgett (License #5621) (closes issue ASTERISK-18019)
+ ........ r323670 | rmudgett | 2011-06-15 11:43:31 -0500 (Wed, 15
+ Jun 2011) | 7 lines Add a test to the event unit tests to catch
+ ASTERISK-18002. The new tests check to see if there are ANY
+ subscribers to the event type when ast_event_check_subscriber()
+ is not passed any specific ie values. (issue ASTERISK-18002)
+ ........
+
+2011-06-15 16:19 +0000 [r323621] Jonathan Rose <jrose@digium.com>
+
+ * res/res_config_pgsql.c, /: Merged revisions 323610 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r323610 | jrose | 2011-06-15 11:09:24 -0500 (Wed, 15 Jun
+ 2011) | 7 lines Adds PQclear calls on result to various parts of
+ res_conf_pgsql (closes issue ASTERISK-17812) Reported by:
+ byronclark Patches: pgsql_pqclear.patch uploaded by byronclark
+ (license 1200) ........
+
+2011-06-15 15:33 +0000 [r323609] Sean Bright <sean@malleable.com>
+
+ * main/manager.c, /: Merged revisions 323608 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400
+ (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400
+ (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun
+ 2011) | 25 lines Resolve a segfault/bus error when we try to map
+ memory that falls on a page boundary. The fix for ASTERISK-15359
+ was incorrect in that it added 1 to the length of the mmap'd
+ region. The problem with this is that reading/writing to that
+ extra byte outside of the bounds of the underlying fd causes a
+ bus error. The real issue is that we are working with both a FILE
+ * and the raw fd underneath it and not synchronizing between
+ them. The code that was removed in ASTERISK-15359 was correct,
+ but we weren't flushing the FILE * before mapping the fd. Looking
+ at the manager code in 1.4 reveals that the FILE * in 'struct
+ mansession' is never used except to create a temporary file that
+ we immediately fdopen. This means we just need to write a 0 byte
+ to the fd and everything will just work. The other branches
+ require a call to fflush() which, while not a guaranteed fix,
+ should reduce the likelihood of a crash. This all makes sense in
+ my head. (closes issue ASTERISK-16460) Reported by:
+ Ravelomanantsoa Hoby (hoby) Patches:
+ issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license
+ #5060) ........ ................ ................
+
+2011-06-15 13:45 +0000 [r323517] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_confbridge.c, CHANGES: CONFBRIDGE_INFO function to get
+ conference data Added the CONFBRIDGE_INFO dialplan function to
+ get information about a conference bridge including locked status
+ and number of parties, admins, and marked users. Review:
+ https://reviewboard.asterisk.org/r/1271/
+
+2011-06-15 00:51 +0000 [r323397-323457] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/event.c: Merged revisions 323456 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r323456 | rmudgett | 2011-06-14 19:50:20 -0500 (Tue, 14 Jun 2011)
+ | 1 line Add missing break in ast_event_get_cached(). ........
+
+ * main/netsock2.c, main/dnsmgr.c, /: Merged revisions 323392,323394
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r323392 | rmudgett | 2011-06-14 12:21:24 -0500 (Tue, 14 Jun 2011)
+ | 6 lines Add more strict hostname checking to
+ ast_dnsmgr_lookup(). Change suggested in review. Review:
+ https://reviewboard.asterisk.org/r/1240/ ........ r323394 |
+ rmudgett | 2011-06-14 12:21:39 -0500 (Tue, 14 Jun 2011) | 2 lines
+ Made ast_sockaddr_split_hostport() port warning msgs more
+ meaningful. ........
+
+2011-06-14 17:03 +0000 [r323374] Terry Wilson <twilson@digium.com>
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, /,
+ channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged
+ revisions 323370 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011)
+ | 10 lines Add rtpkeepalives back to 1.8 The RTP-engine
+ conversion left out support for handling rtpkeepalives. This
+ patch adds them back. (closes issue ASTERISK-17304) Reported by:
+ lmadsen Review: https://reviewboard.asterisk.org/r/1226/ ........
+
+2011-06-14 16:47 +0000 [r323372] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 323371 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r323371 | jrose | 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) |
+ 12 lines Changes contact use in build_peer to use the FORCE_RPORT
+ flag instead of RPORT_PRESENT It turned out that this was causing
+ NAT=Yes to always use rport when present which was against 1.6.2
+ behavior and the check itself was redundant since the only way
+ this segment of code could be reached was if RPORT_PRESENT was
+ already evaluated as true earlier. (closes issue ASTERISK-17789)
+ Reported by: byronclark Patches: use_sip_nat_force_rport.patch
+ uploaded by byronclark (license 1200) ........
+
+2011-06-14 14:37 +0000 [r323325] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Store sip peer name as var data on a
+ outofcall msg.
+
+2011-06-13 20:44 +0000 [r323272] Kinsey Moore <kmoore@digium.com>
+
+ * apps/confbridge/conf_config_parser.c: Config inheritance doesn't
+ work with ConfBridge() menu definitions Current behavior in
+ ConfBridge menu definitions is that first definition takes
+ precedence, even in templated situations. This change allows
+ inheritance and overriding to work as expected so that the last
+ definition takes precedence. (closes ASTERISK-17986) Review:
+ https://reviewboard.asterisk.org/r/1267/
+
+2011-06-13 19:54 +0000 [r323214] Leif Madsen <lmadsen@digium.com>
+
+ * main/channel.c, /: Merged revisions 323213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r323213 | lmadsen | 2011-06-13 15:51:52 -0400 (Mon, 13 Jun 2011)
+ | 6 lines Avoid dividing by zero with L() option to Dial()
+ Reported by: nicolasom Patches: issue-17995.patch - nicolasom
+ (License #5994) ........
+
+2011-06-13 19:43 +0000 [r323212] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
+ channels/sip/include/sip.h: Addition of
+ "outofcall_message_context" sip.conf option. Review:
+ https://reviewboard.asterisk.org/r/1265/
+
+2011-06-13 19:03 +0000 [r323155] Leif Madsen <lmadsen@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 323154 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r323154 | lmadsen | 2011-06-13 15:00:41 -0400 (Mon, 13 Jun 2011)
+ | 6 lines Tweak documentation for AGI Hangup command. (closes
+ issue ASTERISK-17999) Reported by: Ben Klang Patches:
+ hangup-doc.diff - uploaded by Ben Klang (License #5876) ........
+
+2011-06-13 14:38 +0000 [r323106-323107] Kinsey Moore <kmoore@digium.com>
+
+ * apps/confbridge/include/confbridge.h, apps/app_confbridge.c: MOH
+ for only user not working with ConfBridge This adds the
+ playing_moh flag to the conference_bridge_user struct that
+ signifies when MOH should be playing so code doesn't have to
+ guess whether MOH is playing. This change also adds the necessary
+ checking to ensure that MOH continues playing for a single user
+ in a conference after the join sound is played when configured to
+ do so. (closes ASTERISK-17988) Review:
+ https://reviewboard.asterisk.org/r/1263/
+
+ * apps/app_confbridge.c: ConfBridge: Use of bridge or user profiles
+ that don't exist Bridge and user profiles are not checked for
+ existence before use. The lack of a fully formed bridge profile
+ can cause a segfault when sounds are accessed. This change
+ ensures that bridge and user profiles exist prior to usage
+ attempts. Review: https://reviewboard.asterisk.org/r/1264/
+
+2011-06-10 19:22 +0000 [r323041] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 323040 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r323040 | mnicholson | 2011-06-10 14:20:41 -0500 (Fri, 10 Jun
+ 2011) | 5 lines Unlock the sip channel during fax detection like
+ chan_dahdi does to prevent a deadlock with ast_autoservice_stop.
+ (closes issue ASTERISK-17798) tested by mnicholson ........
+
+2011-06-10 15:30 +0000 [r322866-322982] Terry Wilson <twilson@digium.com>
+
+ * /, main/db.c: Merged revisions 322981 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r322981 | twilson | 2011-06-10 08:29:00 -0700 (Fri, 10 Jun 2011)
+ | 11 lines Avoid a DB1 infinite loop bug Explicity check the last
+ entry in the DB and make sure that we don't iterate past it.
+ Since there can be no duplicates, this just makes sure that we
+ stop after matching the last key. This patch also refactors the
+ code to get away from some code duplication. A previous patch
+ added many astdb tests and this patch passed them. Review:
+ https://reviewboard.asterisk.org/r/1259/ ........
+
+ * /, tests/test_db.c (added): Merged revisions 322923 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r322923 | twilson | 2011-06-09 19:33:23 -0700 (Thu, 09
+ Jun 2011) | 2 lines Add some astdb unit tests ........
+
+ * /, include/asterisk/astdb.h: Merged revisions 322865 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r322865 | twilson | 2011-06-09 15:29:20 -0700 (Thu, 09
+ Jun 2011) | 4 lines Correct ast_db_deltree documentation
+ ast_db_deltree returns -1 on error, otherwise the number of
+ deletions ........
+
+2011-06-09 17:43 +0000 [r322808] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 322807 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r322807 | mnicholson | 2011-06-09 12:37:07 -0500 (Thu, 09 Jun
+ 2011) | 5 lines don't drop any voice frames when checking for
+ T.38 during early media (closes issue ASTERISK-17705) Review:
+ https://reviewboard.asterisk.org/r/1186/ patch by oej reported by
+ oej ........
+
+2011-06-09 16:47 +0000 [r322750] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_directed_pickup.c, main/features.c,
+ include/asterisk/features.h: Merged revisions 322749 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09
+ Jun 2011) | 15 lines Remove potential deadlock in call pickup
+ race. Deadlock is possible in ast_do_pickup() when holding the
+ target channel lock and trying to get the chan channel lock.
+ Also, holding the target lock when calling
+ ast_channel_masquerade() is not a good idea because that routine
+ does deadlock avoidance. * Removed the need to hold the target
+ lock after marking the target with a datastore and getting the
+ connected line data off of the target channel. * Moved
+ can_pickup() to ast_can_pickup() in features.c. Now all the call
+ pickup methods use the same basic call pickup availability check.
+ Review: https://reviewboard.asterisk.org/r/1234/ ........
+
+2011-06-09 11:05 +0000 [r322544] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Add autoanswer to skinny. Autoanswer
+ added to skinny based on incoming chan var SKINNY_AUTOANSWER.
+ Initial value must be the time to autoanswer in ms, then
+ optionally :BEEP to play a tone when answered and :MUTE to mute
+ the mic when answering. eg 3000:MUTE:BEEP will ring for 3 secs,
+ then answer, mute the mic, and play a beep. just 3000 would
+ answer afer 3 secs of ringing with no beep and full two way
+ audio.
+
+2011-06-08 20:48 +0000 [r322426-322485] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 322484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r322484 | rmudgett | 2011-06-08 15:46:55 -0500 (Wed, 08 Jun 2011)
+ | 15 lines Ring all queue with more than 255 agents will cause
+ crash. 1. Create a ring-all queue with 500 permanent agents. 2.
+ Call it. 3. Asterisk will crash. The watchers array in
+ app_queue.c has a hard limit of 255. Bounds checking is not done
+ on this array. No sane person should put 255 people in a ring-all
+ queue, but we should not crash anyway. * Added bounds checking to
+ the watchers array. JIRA AST-464 JIRA SWP-2903 ........
+
+ * main/dnsmgr.c, /: Merged revisions 322425 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r322425 | rmudgett | 2011-06-08 13:46:30 -0500 (Wed, 08 Jun 2011)
+ | 16 lines SRV lookup attempted for SIP peers listed as an IP
+ address. Asterisk attempts to SRV lookup a host name even if the
+ host name is an IP address. Regression introduced when IPv6
+ support was added. * Restored the check in ast_dnsmgr_lookup() to
+ see if the given host name is an IP address. The IP address could
+ be in either IPv4 or IPv6 formats. (closes issue ASTERISK-17815)
+ Reported by: Byron Clark Tested by: Byron Clark, Richard Mudgett
+ Patches: issue19248_v1.8.patch - uploaded by Richard Mudgett
+ (License #5621) Review: https://reviewboard.asterisk.org/r/1240/
+ ........
+
+2011-06-08 11:38 +0000 [r322381] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Remove skinny do_monitor and use
+ ast_sched_start instead The do_monitor seemed to be there for
+ task scheduling and network monitoring. However, the network
+ monitoring has a dedicated thread so the ast_io_wait was
+ basically just a usleep as it didn't actually seem to be
+ monitoring anything. Review:
+ https://reviewboard.asterisk.org/r/1256/
+
+2011-06-08 06:45 +0000 [r322323] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, channels/chan_sip.c: Merged revisions 322322 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) |
+ 18 lines Make handle_request_publish do dialog expiration and
+ destruction. This patch fixes handle_request_publish so that it
+ does dialog expiration and destruction. Without this patch the
+ incoming PUBLISH requests will get stuck in the dialog list.
+ Restarting asterisk is the only way to remove them. Personal
+ observation on one system the server hung up while looping
+ through the channels rendering asterisk unusable and all sip
+ phones unregisterd when they try reregister more requests are
+ added. (closes issue #18898) Reported by: gareth Tested by:
+ loloski, Chainsaw, wimpy, se, kuj, irroot Jira:
+ https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review:
+ https://reviewboard.asterisk.org/r/1253 ........
+
+2011-06-07 23:14 +0000 [r322284] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c, include/asterisk/message.h: Correct some
+ whitespace and a reference debug message.
+
+2011-06-07 19:17 +0000 [r322244] Russell Bryant <russell@digium.com>
+
+ * res/res_jabber.c: Actually check the "sendtodialplan" option
+ setting for xmpp. (closes issue ASTERISK-17978) Reported by:
+ elguero Patches: stop_messages_going_to_dialplan.patch (license
+ #5026)
+
+2011-06-07 18:01 +0000 [r322190] Paul Belanger <pabelanger@digium.com>
+
+ * configs/sip_notify.conf.sample, /: Merged revisions 322189 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r322189 | pabelanger | 2011-06-07 13:59:13 -0400 (Tue, 07 Jun
+ 2011) | 4 lines Use correct syntax for 'sip notify snom-reboot'
+ (closes issue ASTERISK-17915) ........
+
+2011-06-06 19:39 +0000 [r322111-322128] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * apps/app_queue.c: Remove Unused Var Warning
+ rt_handle_member_record
+
+ * apps/app_queue.c: Refactor rt_handle_member_record Review:
+ https://reviewboard.asterisk.org/r/1172
+
+2011-06-06 19:15 +0000 [r322070] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/logger.h, /, main/asterisk.c: Merged revisions
+ 322069 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r322069 | jrose | 2011-06-06 14:07:56 -0500 (Mon, 06 Jun 2011) |
+ 8 lines Fixes level toggling for logger set levels since it was
+ reversed (closes issue ASTERISK-17850) Reported by: Luke H Tested
+ by: jrose, Luke H Review:
+ https://reviewboard.asterisk.org/r/1244/ ........
+
+2011-06-03 22:15 +0000 [r321814-321927] Richard Mudgett <rmudgett@digium.com>
+
+ * cel/cel_radius.c, /, cdr/cdr_radius.c: Merged revisions 321926
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321926 | rmudgett | 2011-06-03 17:09:36 -0500 (Fri, 03 Jun 2011)
+ | 18 lines Asterisk crash when unloading cdr_radius/cel_radius.
+ The rc_openlog() API call is passed a string that is used by
+ openlog() to format log messages. The openlog() does not copy the
+ string it just keeps a pointer to it. When the module is
+ unloaded, the string is gone from memory. Depending upon module
+ load order and if the other module then has an error, a crash
+ happens. * Pass rc_openlog() a strdup'd string with the
+ understanding that there will be a small memory leak if the
+ cdr_radius/cel_radius modules are unloaded. * Call rc_destroy()
+ to free the rc handle memory when the module is unloaded. JIRA
+ AST-483 JIRA SWP-3062 ........
+
+ * /, main/ccss.c: Merged revisions 321924 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321924 | rmudgett | 2011-06-03 16:49:17 -0500 (Fri, 03 Jun 2011)
+ | 5 lines Be more explicit for CCSS generic device state event
+ subscription. Make CCSS generic device state event subscription
+ specify the AST_EVENT_IE_STATE ie exists to be safe. ........
+
+ * /, tests/test_event.c, main/event.c: Merged revisions 321871 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321871 | rmudgett | 2011-06-03 15:58:13 -0500 (Fri, 03 Jun 2011)
+ | 27 lines Event subscription fixes. Must commit the subscription
+ fixes together with the integration subscription tests. The
+ subscription fixes cause an erroneously passing test to fail. The
+ new subscription tests detect errors without the subscription
+ fixes. * Added missing event_names[] table entry. * Reworked
+ ast_event_check_subscriber()/match_sub_ie_val_to_event() to
+ correctly detect if a subscriber exists for the proposed event. *
+ Made match_ie_val() and match_sub_ie_val_to_event() check the
+ buffer length for RAW payload types. * Fixed error handling
+ memory leak in ast_event_sub_activate(), ast_event_unsubscribe(),
+ and ast_event_queue(). * Made ast_event_new() and
+ ast_event_check_subscriber() better protect themselves from an
+ invalid payload type. * Added container lock protection between
+ removing old cache events and adding the new cached event in
+ ast_event_queue_and_cache()/event_update_cache(). * Added new
+ event subscription tests. ........
+
+ * include/asterisk/event.h, /, channels/chan_sip.c, main/event.c,
+ channels/chan_iax2.c: Merged revisions 321812-321813 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r321812 | rmudgett | 2011-06-03 14:55:21 -0500 (Fri, 03
+ Jun 2011) | 1 line Correct IAX2 and SIP event subscription
+ description string. ........ r321813 | rmudgett | 2011-06-03
+ 14:56:09 -0500 (Fri, 03 Jun 2011) | 1 line Constify subscription
+ description parameter string. ........
+
+2011-06-03 18:25 +0000 [r321752] Russell Bryant <russell@digium.com>
+
+ * tests/test_astobj2.c, main/astobj2.c: Fix some astobj2 iterator
+ breakage, add another unit test. Review:
+ https://reviewboard.asterisk.org/r/1254/
+
+2011-06-03 13:18 +0000 [r321689] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 321685 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321685 | lmadsen | 2011-06-03 08:17:50 -0500 (Fri, 03 Jun 2011)
+ | 5 lines Also document the 'queue-minute' option. (closes issue
+ #19386) Reported by: juanmol ........
+
+2011-06-02 22:09 +0000 [r321617] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix message destination extension. Don't
+ send all messages to 's'. Get the destination from the request
+ URI. (Found using automated test cases).
+
+2011-06-01 23:12 +0000 [r321548] Richard Mudgett <rmudgett@digium.com>
+
+ * main/cdr.c, /: Merged revisions 321547 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321547 | rmudgett | 2011-06-01 18:11:55 -0500 (Wed, 01 Jun 2011)
+ | 1 line CDR comment tweaks. ........
+
+2011-06-01 21:31 +0000 [r321546] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, channels/chan_sip.c, configs/jabber.conf.sample,
+ include/asterisk/message.h (added), include/asterisk/jabber.h,
+ include/asterisk/channel.h, configs/sip.conf.sample,
+ include/asterisk/_private.h, CHANGES, res/res_jabber.c,
+ main/message.c (added), channels/sip/include/sip.h,
+ main/asterisk.c: Support routing text messages outside of a call.
+ Asterisk now has protocol independent support for processing text
+ messages outside of a call. Messages are routed through the
+ Asterisk dialplan. SIP MESSAGE and XMPP are currently supported.
+ There are options in sip.conf and jabber.conf that enable these
+ features. There is a new application, MessageSend(). There are
+ two new functions, MESSAGE() and MESSAGE_DATA(). Documentation
+ will be available on the project wiki, wiki.asterisk.org. Thanks
+ to Terry Wilson for the assistance with development and to David
+ Vossel for helping with some additional testing. Review:
+ https://reviewboard.asterisk.org/r/1042/
+
+2011-06-01 20:11 +0000 [r321538] Brett Bryant <bbryant@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 321537 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r321537 | bbryant | 2011-06-01 16:10:02 -0400 (Wed, 01
+ Jun 2011) | 8 lines This patch fixes an issue with using the
+ wrong voicemail folders with greetings. (closes issue #17871)
+ Reported by: edhorton Patches: digium_bug_17871_2 uploaded by
+ fhackenberger (license 592) Tested by: edhorton, fhackenberger
+ ........
+
+2011-06-01 10:45 +0000 [r321529] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, /, addons/ooh323c/src/ooh245.c,
+ addons/ooh323c/src/oochannels.c: Merged revisions 321528 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14
+ lines Fix double alerting, add forced alerting before answer Fix
+ double alerting (it wasn't fixed here by issue #18542) Add forced
+ alerting before connect (if it wasn't before) Try to send all
+ packets from outgoing queue rather than one only Call goes into
+ clearing state when disconnect command is received (closes issue
+ #19361) Reported by: vmikhelson Patches: issue19361-3.patch
+ uploaded by may213 (license 454) Tested by: vmikhelson ........
+
+2011-05-31 20:55 +0000 [r321518] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/acl.h, /, include/asterisk/dnsmgr.h: Merged
+ revisions 321517 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321517 | rmudgett | 2011-05-31 15:54:35 -0500 (Tue, 31 May 2011)
+ | 1 line Update some comments. ........
+
+2011-05-31 19:01 +0000 [r321516] David Vossel <dvossel@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 321515 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r321515 | dvossel | 2011-05-31 13:52:54 -0500 (Tue, 31
+ May 2011) | 12 lines Chan_local locking cleanup. This patch
+ removes all of the unnecessary deadlock avoidance loops that
+ occur in chan_local. It also resolves an issue with a deadlock
+ triggered by local channel optimizations. (issue #18028) Review:
+ https://reviewboard.asterisk.org/r/1231/ ........
+
+2011-05-31 16:06 +0000 [r321512] Leif Madsen <lmadsen@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 321511 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321511 | lmadsen | 2011-05-31 12:04:47 -0400 (Tue, 31 May 2011)
+ | 8 lines Enhance NOTICE message to know who couldn't access the
+ dialplan. (closes issue #19390) Reported by: lmadsen Patches:
+ __20110531-sip-notice-tweak.txt uploaded by lmadsen (license 10)
+ Tested by: russell ........
+
+2011-05-28 00:29 +0000 [r321338-321445] Richard Mudgett <rmudgett@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 321436 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321436 | rmudgett | 2011-05-27 19:27:52 -0500 (Fri, 27 May 2011)
+ | 4 lines Some hagi launch cleanup. Inspired by issue 19256. This
+ patch would also fix the crash. ........
+
+ * main/srv.c, /: Merged revisions 321392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321392 | rmudgett | 2011-05-27 18:45:41 -0500 (Fri, 27 May 2011)
+ | 12 lines Crash when using hagi and no servers are available.
+ When none of the servers returned by the SRV querey respond,
+ asterisk crashes. The problem is that if the loop over all the
+ SRV entries finishes then the srv_context has already been
+ cleaned up. * Make ast_srv_cleanup() check to see if the context
+ is already cleaned up. (closes issue #19256) Reported by:
+ byronclark ........
+
+ * /, apps/app_privacy.c, UPGRADE.txt, CHANGES: Merged revisions
+ 321337 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 Also revert
+ -r321331 and -r321332. ........ r321337 | rmudgett | 2011-05-27
+ 17:06:43 -0500 (Fri, 27 May 2011) | 7 lines The app_privacy args
+ have undocumented "options" position, interferes with "context"
+ position. * Add documention for unused "options" position to
+ match existing code. (closes issue #19273) Reported by:
+ mdavenport ........
+
+2011-05-27 21:40 +0000 [r321334] Leif Madsen <lmadsen@digium.com>
+
+ * /, main/features.c: Merged revisions 321333 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321333 | lmadsen | 2011-05-27 17:40:23 -0400 (Fri, 27 May 2011)
+ | 7 lines Allow parking lot hints and musicclass to be set.
+ (closes issue #19378) Reported by: sboily_proformatique Patches:
+ pf_parkinghint_music_fix uploaded by sboily proformatique
+ (license 206) Tested by: russell ........
+
+2011-05-27 21:37 +0000 [r321331-321332] Richard Mudgett <rmudgett@digium.com>
+
+ * UPGRADE.txt: Add note about PrivacyManager to UPGRADE.txt
+
+ * /, apps/app_privacy.c, CHANGES: Merged revisions 321330 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011)
+ | 8 lines The app_privacy args have undocumented "options"
+ position, interferes with "context" position. * Add documention
+ for unused "options" position to match existing code. The
+ trunk(v1.10) version will remove the unused options position.
+ (closes issue #19273) Reported by: mdavenport ........
+
+2011-05-27 16:35 +0000 [r321289] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/sip/reqresp_parser.c: Merged revisions 321273 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321273 | jrose | 2011-05-27 09:59:34 -0500 (Fri, 27 May 2011) |
+ 3 lines markm committed a patch I was working on yesterday, this
+ fixes it to mesh up with suggestions by mnicholson. ........
+
+2011-05-27 08:37 +0000 [r321212] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, main/features.c: Merged revisions 321211 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321211 | alecdavis | 2011-05-27 20:31:15 +1200 (Fri, 27 May
+ 2011) | 16 lines Fix *8 directed pickup locks system during
+ pickupsound play out move playout from sip_pickup_thread to
+ bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2
+ threads trying to write audio to same channel. In addition fixes
+ choppy audio beep in issue 19177. (issue #18654) (issue #19177)
+ Reported by: Docent Patches: review1232-1.8.diff.txt alecdavis
+ (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1232/ ........
+
+2011-05-26 21:50 +0000 [r321101-321156] Mark Murawki <markm@intellasoft.net>
+
+ * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Merged
+ revisions 321155 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321155 | markm | 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) |
+ 10 lines Fixed build problem with dev mode enabled, which was
+ caused by commit 321100. Reformulated patch to be more generic.
+ Moved the sip uri parse variable initalization to parse_uri_full
+ in reqresp_parser.c. This will ensure that any use of parse uri
+ will have null output variables if the parse fails. (closes issue
+ #19346) Reported by: kobaz Tested by: kobaz,JonathanRose Review:
+ [full review board URL with trailing slash] ........
+
+ * main/netsock2.c, /, channels/chan_sip.c: Merged revisions 321100
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321100 | markm | 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) |
+ 11 lines ast_sockaddr_resolve() in netsock2.c may deref a null
+ pointer Added a null check in netsock2 ast_sockaddr_resolve() as
+ well as added default initalizers in chan_sip
+ parse_uri_legacy_check() to make sure that invalid uris will make
+ null (and not undefined) user,pass,domain,transport variables
+ (closes issue #19346) Reported by: kobaz Patches: netsock2.patch
+ uploaded by kobaz (license 834) Tested by: kobaz, Marquis
+ ........
+
+2011-05-26 18:10 +0000 [r321045] Richard Mudgett <rmudgett@digium.com>
+
+ * /, include/asterisk/netsock2.h: Merged revisions 321044 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321044 | rmudgett | 2011-05-26 13:10:17 -0500 (Thu, 26 May 2011)
+ | 1 line Update ast_sockaddr comment with an important note.
+ ........
+
+2011-05-26 17:35 +0000 [r321043] Terry Wilson <twilson@digium.com>
+
+ * main/rtp_engine.c, /: Merged revisions 321042 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r321042 | twilson | 2011-05-26 10:29:54 -0700 (Thu, 26 May 2011)
+ | 6 lines Initialize stack-allocated ast_sockaddrs before use It
+ is important to always initialize ast_sockaddrs before use--even
+ if they are passed to ast_sockaddr_copy as the underlying storage
+ could be bigger than what ends up being copied--leaving part of
+ the data unitialized. ........
+
+2011-05-26 16:54 +0000 [r321003] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_alsa.c: Merged revisions 320947 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r320947 | russell | 2011-05-26 10:57:13 -0500 (Thu, 26
+ May 2011) | 2 lines Remove some variables that were set but
+ unused. ........
+
+2011-05-26 15:55 +0000 [r320946] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, main/utils.c, include/asterisk/stringfields.h:
+ Use va_copy for stringfields The ast_string_field_build_va
+ functions were written to take to separate va_lists to work
+ around FreeBSD 4 not having va_copy defined. In the end, we don't
+ support anything using gcc < 3 anyway because we use va_copy all
+ over the place anyway. This patch just simplifies things by
+ removing the second va_list function arguments in favor of
+ va_copy. Review: https://reviewboard.asterisk.org/r/1233/ --This
+ line, and those below, will be ignored-- M
+ include/asterisk/stringfields.h M main/utils.c M main/channel.c
+
+2011-05-25 22:28 +0000 [r320820-320884] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 320883 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r320883 | rmudgett | 2011-05-25 17:25:18 -0500 (Wed, 25 May 2011)
+ | 17 lines Native SIP CCSS sends bad CC cancel SUBSCRIBE message.
+ The SUBSCRIBE message used to cancel a CC request has incorrect
+ To/From SIP headers. They are reversed and the dialog tags are
+ the same when they should not be. If pedantic mode was disabled,
+ then the cancel would have succeeded despite the incorrect
+ message. * The SIP_OUTGOING flag was not set correctly for the
+ dialog and I had to move some CC subscribe handling code as a
+ result. * Initialized the dialog subscribed type to
+ CALL_COMPLETION earlier. If a CC request SUBSCRIBE message comes
+ in and the CC instance is not found, the 404 response was
+ duplicated. JIRA AST-568 JIRA SWP-3493 ........
+
+ * apps/app_dial.c, main/channel.c, main/manager.c, /,
+ apps/app_meetme.c, apps/app_fax.c, main/features.c, CHANGES,
+ apps/app_queue.c, UPGRADE-1.8.txt: Merged revisions 320823 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011)
+ | 18 lines The AMI Newstate event contains different information
+ between v1.4 and v1.8. The addition of connected line support in
+ v1.8 changes the behavior of the channel caller ID somewhat. The
+ channel caller ID value no longer time shares with the connected
+ line ID on outgoing call legs. The timing of some AMI
+ events/responses output the connected line ID as caller ID. These
+ party ID's are now separate. * The ConnectedLineNum and
+ ConnectedLineName headers were added to many AMI events/responses
+ if the CallerIDNum/CallerIDName headers were also present.
+ (closes issue #18252) Reported by: gje Tested by: rmudgett
+ Review: https://reviewboard.asterisk.org/r/1227/ ........
+
+ * main/channel.c, /, main/format_cap.c, main/features.c,
+ include/asterisk/channel.h: Merged revisions 320796 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r320796 | rmudgett | 2011-05-25 11:23:11 -0500 (Wed, 25
+ May 2011) | 17 lines Give zombies a safe channel driver to use.
+ Recent crashes from zombie channels suggests that they need a
+ safe home to goto. When a masquerade happens, the physical part
+ of the zombie channel is hungup. The hangup normally sets the
+ channel private pointer to NULL. If someone then blindly does a
+ callback to the channel driver, a crash is likely because the
+ private pointer is NULL. The masquerade now sets the channel
+ technology of zombie channels to the kill channel driver. Related
+ to the following issues: (issue #19116) (issue #19310) Review:
+ https://reviewboard.asterisk.org/r/1224/ ........
+
+2011-05-25 15:43 +0000 [r320772] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * funcs/func_channel.c, CHANGES: CHANNEL(pickupgroup) Allow Setting
+ / Reading the pickupgroup of a channel with func_channel.c
+ (closes issue #19045) Reported by: irroot Review:
+ https://reviewboard.asterisk.org/r/1148/
+
+2011-05-25 00:52 +0000 [r320717] Terry Wilson <twilson@digium.com>
+
+ * /, addons/chan_mobile.c: Merged revisions 320716 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r320716 | twilson | 2011-05-24 17:49:10 -0700 (Tue, 24
+ May 2011) | 4 lines Cast data as char * before using S_OR This is
+ required for compiling successfully under dev mode ........
+
+2011-05-23 18:00 +0000 [r320651] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, /, CHANGES: Merged revisions 320650 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r320650 | rmudgett | 2011-05-23 12:53:44 -0500 (Mon, 23
+ May 2011) | 16 lines Add ConnectedLineNum/Name headers to output
+ of AMI action Status. * Add ConnectedLineNum and
+ ConnectedLineName headers to the output of the AMI action Status.
+ This makes it easier to find out who the channel is connected to
+ without having to lookup BridgedChannel or when they are
+ connected to an application (e.g.: VoiceMail) which has no
+ bridged channel. * Bridged channels with no CallerID had ""
+ instead of "<unknown>" output, that might be a bug as "<unknown>"
+ was what older versions used. (closes issue #18158) Reported by:
+ gareth Patches: svn-292308.diff uploaded by gareth (license 208)
+ ........
+
+2011-05-23 16:28 +0000 [r320606] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 320568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r320568 | dvossel | 2011-05-23 11:18:33 -0500
+ (Mon, 23 May 2011) | 14 lines Merged revisions 320562 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r320562 | dvossel | 2011-05-23 11:15:18 -0500 (Mon, 23 May 2011)
+ | 9 lines Adds missing part to the ast_tcptls_server_start fails
+ second attempt to bind patch. (closes issue #19289) Reported by:
+ wdoekes Patches:
+ issue19289_delay_old_address_setting_tcptls_2.patch uploaded by
+ wdoekes (license 717) ........ ................
+
+2011-05-23 16:20 +0000 [r320579] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, configure, configure.ac: Merged revisions 320573 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r320573 | tilghman | 2011-05-23 11:19:32 -0500 (Mon, 23
+ May 2011) | 7 lines GNU libiconv uses symbol "libiconv_open"
+ instead of "iconv_open". (closes issue #19344) Reported by:
+ rohanl Patches: iconv-check.patch uploaded by rohanl (license
+ 1284) ........
+
+2011-05-23 15:48 +0000 [r320561] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 320560 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r320560 | kpfleming | 2011-05-23 10:47:14 -0500 (Mon, 23 May
+ 2011) | 4 lines Don't generate spurious "No: command not found"
+ messages when running the configure script on a system that has
+ neither gmime-config nor pkg-config. ........
+
+2011-05-23 14:40 +0000 [r320505] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 320504 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r320504 | jrose | 2011-05-23 09:33:20 -0500 (Mon, 23 May 2011) |
+ 10 lines Fixes segfault occuring in chan_sip.c at
+ __set_address_from_contact Checks to see if domain contains
+ anything before sending it off to ast_sockaddr_resolve which is
+ where the segfault was occuring due to null str. (closes issue
+ #18857) Reported by: sybasesql Review:
+ https://reviewboard.asterisk.org/r/1225/ ........
+
+2011-05-22 23:36 +0000 [r320446] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, res/res_odbc.c: Merged revisions 320445 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r320445 | tilghman | 2011-05-22 18:34:57 -0500
+ (Sun, 22 May 2011) | 15 lines Merged revisions 320444 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r320444 | tilghman | 2011-05-22 18:25:51 -0500 (Sun, 22 May 2011)
+ | 8 lines Don't crash when the connection fails. (closes issue
+ #19250) Reported by: seadweller Patches:
+ 20110514__issue19250.diff.txt uploaded by tilghman (license 14)
+ Tested by: seadweller, sum ........ ................
+
+2011-05-20 21:40 +0000 [r320340] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 320338 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r320338 | dvossel | 2011-05-20 16:39:36 -0500
+ (Fri, 20 May 2011) | 14 lines Merged revisions 320271 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r320271 | dvossel | 2011-05-20 16:24:48 -0500 (Fri, 20 May 2011)
+ | 8 lines Fixes issue with ast_tcptls_server_start failing on
+ second attempt to bind. (closes issue #19289) Reported by:
+ wdoekes Patches:
+ issue19289_delay_old_address_setting_tcptls.patch uploaded by
+ wdoekes (license 717) ........ ................
+
+2011-05-20 20:53 +0000 [r320238] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 320237 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r320237 | rmudgett | 2011-05-20 15:49:03 -0500
+ (Fri, 20 May 2011) | 27 lines Merged revisions 320236 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r320236 | rmudgett | 2011-05-20 15:44:54 -0500
+ (Fri, 20 May 2011) | 20 lines Merged revisions 320235 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011)
+ | 13 lines The meetme CLI command completion leaves conferences
+ mutex locked. When issuing a meetme kick CLI command and an
+ invalid (non-existent) conference number is specified, pressing
+ Tab leaves the conferences mutex locked and, therefore, all
+ conferences deadlock. Add missing unlock. (closes issue #19336)
+ Reported by: zvision Patches: app_meetme.diff uploaded by zvision
+ (license 798) ........ ................ ................
+
+2011-05-20 18:49 +0000 [r320181] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 320180 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r320180 | mnicholson | 2011-05-20 13:48:46 -0500 (Fri, 20 May
+ 2011) | 16 lines This commit modifies the way polling is done on
+ TLS sockets. Because of the buffering the TLS layer does, polling
+ is unreliable. If poll is called while there is data waiting to
+ be read in the TLS layer but not at the network layer, the
+ messaging processing engine will not proceed until something else
+ writes data to the socket, which may not occur. This change
+ modifies the logic around TLS sockets to only poll after a failed
+ read on a non-blocking socket. This way we know that there is no
+ data waiting to be read from the buffering layer. (closes issue
+ #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by
+ mnicholson (license 96) Tested by: mnicholson ........
+
+2011-05-20 18:29 +0000 [r320178] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 320162 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r320162 | jrose | 2011-05-20 13:12:21 -0500 (Fri, 20 May
+ 2011) | 15 lines Fixes an imapfolder related crash imapfolders
+ being set in the general section of voicemail would cause the
+ inbox folder name to change. Since sound file names are made
+ based on the names of the folders, this would cause the audio
+ related to that folder name to change and if Asterisk attempted
+ to play it, the channel would instantly hang up when the audio
+ file couldn't be found. This patch searches for the name of the
+ folder first to leave existing behavior in tact and if that
+ fails, it uses the normal inbox name to get the sound file
+ instead. (closes issue #16104) Reported by: blkline Review:
+ https://reviewboard.asterisk.org/r/1215/ ........
+
+2011-05-20 17:04 +0000 [r320058-320060] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 320059 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r320059 | rmudgett | 2011-05-20 12:03:49 -0500 (Fri, 20 May 2011)
+ | 1 line Misc comment cleanup in features.c. ........
+
+ * main/channel.c, /, main/features.c: Merged revisions 320057 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r320057 | rmudgett | 2011-05-20 11:43:02 -0500 (Fri, 20 May 2011)
+ | 19 lines Crash while transferring a call during DTMF feature
+ timeout. When a call is being attended transferred during the
+ time between AST_FRAME_DTMF_BEGIN and AST_FRAME_DTMF_END, the
+ transferred channel becomes a zombie (so tech data is not
+ available), making ast_dtmf_stream() segfault when it tries to
+ send the DTMF digit (at least with SIP channels). Patch based on
+ feature-end-zombie.patch uploaded by Irontec (license 1256) *
+ Check for zombies when ast_channel_bridge() returns. * Guarantee
+ that the fo parameter value is initialized in
+ ast_channel_bridge() before any returns. (closes issue #19116)
+ Reported by: Irontec Tested by: rmudgett ........
+
+2011-05-20 16:27 +0000 [r320040] Jonathan Rose <jrose@digium.com>
+
+ * funcs/func_strings.c, CHANGES: Adds STRREPLACE function Adds a
+ new STRREPLACe function to func_strings.c that allows users to
+ search and replace against a variable in the dialplan. (closes
+ issue #18023) Reported by: wdoekes Review:
+ https://reviewboard.asterisk.org/r/1219/
+
+2011-05-20 16:20 +0000 [r319998-320013] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_directed_pickup.c, main/features.c: Merged revisions
+ 320007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r320007 | rmudgett | 2011-05-20 11:19:01 -0500 (Fri, 20 May 2011)
+ | 2 lines Change some variable names to make pickup code easier
+ to understand. ........
+
+ * /, apps/app_directed_pickup.c, main/features.c: Merged revisions
+ 319997 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r319997 | rmudgett | 2011-05-20 10:48:25 -0500 (Fri, 20 May 2011)
+ | 25 lines Crash when using directed pickup applications. The
+ directed pickup applications can cause a crash if the pickup was
+ successful because the dialplan keeps executing. This patch does
+ the following: * Completes the channel masquerade on a successful
+ pickup before the application returns. The channel is now
+ guaranteed a zombie and must not continue executing the dialplan.
+ * Changes the return value of the directed pickup applications to
+ return zero if the pickup failed and nonzero(-1) if the pickup
+ succeeded. * Made some code optimizations that no longer require
+ re-checking the pickup channel to see if it is still available to
+ pickup. (closes issue #19310) Reported by: remiq Patches:
+ issue19310_v1.8_v2.patch uploaded by rmudgett (license 664)
+ Tested by: alecdavis, remiq, rmudgett Review:
+ https://reviewboard.asterisk.org/r/1221/ ........
+
+2011-05-20 13:42 +0000 [r319867-319939] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample,
+ channels/sip/include/sip.h: Merged revisions 319938 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May
+ 2011) | 12 lines Adds legacy_useroption_parsing to address
+ interoperability concerns. With the new option engaged, Asterisk
+ should interpret user fields with useroptions contained within
+ the userfield of the uri by stripping them out of the original
+ message whenever a semicolon is encountered in the userfield
+ string. (closes issue #18344) Reported by: danimal Tested by:
+ jrose Review: https://reviewboard.asterisk.org/r/1223/ ........
+
+ * /, main/features.c: Merged revisions 319866 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r319866 | jrose | 2011-05-19 13:32:38 -0500 (Thu, 19 May 2011) |
+ 11 lines Fix Randomize option on Park() The randomize option was
+ generally not working like it should have at all on Park(). This
+ patch restores intended functionality. (closes issue #18862)
+ Reported by: davidw Tested by: jrose Review:
+ https://reviewboard.asterisk.org/r/1222/ ........
+
+2011-05-19 18:12 +0000 [r319813] Mark Murawki <markm@intellasoft.net>
+
+ * cel/cel_odbc.c, /: Merged revisions 319812 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r319812 | markm | 2011-05-19 13:59:01 -0400 (Thu, 19 May 2011) |
+ 9 lines In cel_odbc, an uninitialized RWLIST is attempted to be
+ locked. Added INIT and DESTROY for the RWLIST odbc_tables (closes
+ issue #19331) Reported by: kobaz Patches: odbc_cel.patch uploaded
+ by kobaz (license 834) ........
+
+2011-05-19 16:52 +0000 [r319759] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/ccss.c: Merged revisions 319758 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r319758 | rmudgett | 2011-05-19 11:50:48 -0500 (Thu, 19 May 2011)
+ | 21 lines CCSS generic agent with POTS and ISDN phones fail
+ caller busy call-back test. If the following is true after a CCSS
+ activation: * The generic agent is for an analog phone or ISDN
+ phone. (Caller party) * The called party becomes available. * The
+ caller party is not available. When the caller party becomes
+ available, the caller is not alerted to the called party being
+ available. The generic agent still thinks the caller is busy. *
+ Fixed the generic agent device state event subscription to look
+ for all device states that are considered available. *
+ Encapsulated the device state test for CCSS generic device
+ available in cc_generic_is_device_available(). Made the generic
+ agent and monitor use the new function instead of the manually
+ coded inline equivalent. JIRA AST-559 JIRA SWP-3462 ........
+
+2011-05-18 23:18 +0000 [r319530-319661] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 319654 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r319654 | twilson | 2011-05-18 16:15:58 -0700
+ (Wed, 18 May 2011) | 22 lines Merged revisions 319653 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r319653 | twilson | 2011-05-18 16:11:57 -0700
+ (Wed, 18 May 2011) | 15 lines Merged revisions 319652 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011)
+ | 8 lines Make sure everyone gets an unhold when a transfer
+ succeeds Some phones, like the Snom phones, send a hold to the
+ transfer target after before sending the REFER. We need to make
+ sure that we unhold the parties that are being connected after
+ the masquerade. If Local channels with the /nm option are used
+ when dialing the parties, hold music would still be playing on
+ the transfer target, even after being connected with the
+ transferee. ........ ................ ................
+
+ * /, channels/chan_sip.c: Merged revisions 319552 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r319552 | twilson | 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011)
+ | 11 lines Unbreak the storing of registrations for restart The
+ fix for issue 18882 broke retrieving non-realtime peers from the
+ ast_db on restart/reload. This patch tries to unbreak things
+ while leaving the intent of the original fix intact. (closes
+ issue #19318) Reported by: remiq Patches: diff.txt uploaded by
+ twilson (license 396) Tested by: lmadsen, remiq ........
+
+ * apps/app_dial.c, /: Merged revisions 319529 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r319529 | twilson | 2011-05-18 13:05:34 -0700
+ (Wed, 18 May 2011) | 24 lines Merged revisions 319528 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r319528 | twilson | 2011-05-18 13:02:06 -0700
+ (Wed, 18 May 2011) | 17 lines Merged revisions 319527 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011)
+ | 10 lines Fix app_dial ring groups Revert part of r315643. We
+ need to remove the datastore here as well. The code in bridging
+ code will catch anything that app_dial might miss. (closes issue
+ #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff
+ uploaded by elguero (license 37) ........ ................
+ ................
+
+2011-05-17 22:04 +0000 [r319471] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/misdn/isdn_lib.c: Merged revisions 319469 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r319469 | rmudgett | 2011-05-17 16:57:56 -0500
+ (Tue, 17 May 2011) | 22 lines Merged revision 319468 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue,
+ 17 May 2011) | 15 lines The mISDN HDLC mode is prevented on
+ dialed channels. The use of mISDN HDLC mode is prevented if the
+ mISDN dial technology option 'h1' is used when config option
+ astdtmf=yes. There is a bug in channels/misdn/isdn_lib.c which
+ prevents the use of HDLC mode. Instead of setting the channel to
+ HDLC mode it is set to transparent(no dsp, no hdlc), although
+ hdlc is not "no hdlc". I.e the logging message is correct, but
+ the if condition is not. Make check the nodsp and hdlc flags.
+ JIRA ABE-2787 JIRA SWP-3437 .......... ................
+
+2011-05-17 21:59 +0000 [r319470] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Remove extraneous line variables. The
+ vars were either explicitly or implicitly not used.
+
+2011-05-17 20:13 +0000 [r319427] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sig_pri.c: Option needed for Q931_IE_TIME_DATE to be
+ optional in CONNECT message. The NEC SV8300 rejects the
+ Q931_IE_TIME_DATE for Q.SIG. Add option to specify if and how
+ much of the current time is put in Q931_IE_TIME_DATE. * Send
+ date/time ie never. * Send date/time ie date only. * Send
+ date/time ie date and hour. * Send date/time ie date, hour, and
+ minute. * Send date/time ie date, hour, minute, and second. *
+ Send date/time ie default: Libpri will send date and hhmm only
+ when in NT PTMP mode to support ISDN phones. (closes issue
+ #19221) Reported by: kenner JIRA SWP-3396
+
+2011-05-17 12:54 +0000 [r319366-319368] Leif Madsen <lmadsen@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 319367 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r319367 | lmadsen | 2011-05-17 07:53:50 -0500 (Tue, 17
+ May 2011) | 10 lines Don't create [general] voicemail context
+ when using users.conf Prior to this patch, app_voicemail would
+ create a [general] context when parsing users.conf. (closes issue
+ #18891) Reported by: pdugas Patches:
+ app_voicemail-ignore-general.patch uploaded by pdugas (license
+ 1222) app_voicemail-ignore-general-style-guidelines.patch
+ uploaded by seanbright (license 71) Tested by: pdugas ........
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 319365 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r319365 | lmadsen | 2011-05-17 07:39:37 -0500 (Tue, 17 May 2011)
+ | 6 lines Make Debian init script lsb compliant (closes issue
+ #18896) Reported by: manwe Patches: debian_init_lsb.patch
+ uploaded by manwe (license 1223) ........
+
+2011-05-16 21:39 +0000 [r319316] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fix up skinny hints. Probably haven't
+ been working for a couple of years. May still need some more
+ love, but they are now working, both as a hint device and
+ monitoring a hint. Changes centre around the long ago change to
+ remove the requirement for a device name in a skinny line, and
+ changes to the transmit_* functions.
+
+2011-05-16 21:08 +0000 [r319262] Jonathan Rose <jrose@digium.com>
+
+ * main/dsp.c: Merged revisions 319261 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r319261 | jrose | 2011-05-16 16:00:55 -0500 (Mon, 16 May 2011) |
+ 2 lines Makes busy detection in dsp.c always allow for at least
+ one frame (20ms) of error so that 200ms tone lengths don't get
+ ignored by single frame error lengths. ........
+
+2011-05-16 20:41 +0000 [r319260] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/ccss.c: Merged revisions 319259 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r319259 | rmudgett | 2011-05-16 15:33:37 -0500 (Mon, 16 May 2011)
+ | 13 lines Deadlock between generic CCSS agent and native ISDN
+ CCSS. Deadlock can occur when the generic CCSS agent is deleting
+ duplicate CC offers and the native ISDN CC driver is processing
+ an incoming CC message. The cc_core_instances container lock
+ cannot be held when an agent or monitor callback is invoked
+ without the possibility of a deadlock. * Make
+ kill_duplicate_offers() remove the reference in cc_core_instances
+ outside of the container lock. JIRA AST-566 JIRA SWP-3469
+ ........
+
+2011-05-16 18:21 +0000 [r319212] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 319204 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r319204 | twilson | 2011-05-16 13:17:43 -0500
+ (Mon, 16 May 2011) | 11 lines Merged revisions 319202 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r319202 | twilson | 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011)
+ | 4 lines Unlink a peer from peers_by_ip when expiring a
+ registration Review: https://reviewboard.asterisk.org/r/1218/
+ ........ ................
+
+2011-05-16 15:58 +0000 [r319146] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 319145 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r319145 | dvossel | 2011-05-16 10:57:26 -0500
+ (Mon, 16 May 2011) | 9 lines Merged revisions 319144 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r319144 | dvossel | 2011-05-16 10:56:16 -0500 (Mon, 16
+ May 2011) | 2 lines Fixes issue with peer ref-counting during
+ handle_request_subscribe. (closes issue #19293) Reported by:
+ irroot ........ ................
+
+2011-05-16 15:54 +0000 [r319143] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 319142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r319142 | mnicholson | 2011-05-16 10:53:26 -0500 (Mon, 16 May
+ 2011) | 8 lines Make sure tcptls_session exists before
+ dereferencing it. (closes issue #19192) Reported by: stknob
+ Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by
+ Chainsaw (license 723) Tested by: vois, Chainsaw ........
+
+2011-05-16 14:56 +0000 [r319087] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * channels/chan_sip.c, res/res_fax.c, CHANGES,
+ channels/sip/include/sip.h: When a error in T.38 negotiation
+ happens or its rejected on a channel the state of the channel
+ reverts to unknown this should be rejected. this is important for
+ negotiating T.38 gateway see #13405 This patch adds a option
+ T38_REJECTED that behaves as T38_DISABLED except it reports state
+ rejected. Trivial Change to res_fax to honnor UNAVAILABLE and
+ REJECTED states. (closes issue #18889) Reported by: irroot Tested
+ by: irroot, darkbasic, mnicholson Review:
+ https://reviewboard.asterisk.org/r/1115
+
+2011-05-16 14:38 +0000 [r319086] Paul Belanger <pabelanger@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ res/res_http_post.c: Merged revisions 319085 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r319085 | pabelanger | 2011-05-16 10:35:21 -0400 (Mon, 16 May
+ 2011) | 10 lines Support gmime-2.4 (closes issue #18863) Reported
+ by: tzafrir Patches: gmime-2.4-18.diff uploaded by tzafrir
+ (license 46) Tested by: tzafrir Review:
+ https://reviewboard.asterisk.org/r/1213/ ........
+
+2011-05-16 14:29 +0000 [r319084] David Vossel <dvossel@digium.com>
+
+ * /, formats/format_wav.c: Merged revisions 319083 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r319083 | dvossel | 2011-05-16 09:26:33 -0500 (Mon, 16
+ May 2011) | 5 lines Fixes Big Endian build issue. (closes issue
+ #19298) Reported by: tzafrir ........
+
+2011-05-15 23:17 +0000 [r319024] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Add activatesub and dialandactivate sub.
+ When called, activatesub first cleans up the active sub and then
+ handles the sub passed. dialandactivatesub first sets sub->exten
+ and then calls activatesub. Revise handle_offhook to utilise the
+ callid sent to chan_skinny. Some other minor fixes especially
+ around d->hookstate (which still needs some more work).
+
+2011-05-13 18:10 +0000 [r318918-318922] Brett Bryant <bbryant@digium.com>
+
+ * main/channel.c, /: Merged revisions 318921 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r318921 | bbryant | 2011-05-13 14:09:34 -0400 (Fri, 13 May 2011)
+ | 8 lines Fixes a segmentation fault in dynamic hints when a
+ channel technology isn't loaded for a hint. (closes issue #18495)
+ Reported by: bertrand Tested by: bertrand ........
+
+ * /, res/res_srtp.c: Merged revisions 318919 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r318919 | bbryant | 2011-05-13 14:04:50 -0400 (Fri, 13 May 2011)
+ | 10 lines This patch fixes an issue with SRTP which makes
+ HOLD/UNHOLD impossible when too much time has passed between
+ sending audio. (closes issue #18206) Reported by: bernhardsi
+ Patches: res_srtp_unhold.patch uploaded by bernhards (license
+ 1138) Tested by: bernhards, notthematrix ........
+
+ * /, channels/chan_sip.c: Merged revisions 318917 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r318917 | bbryant | 2011-05-13 13:56:04 -0400 (Fri, 13 May 2011)
+ | 11 lines This patch allows TCP peers into the ast_db where they
+ were previously restricted. (closes issue #18882) Reported by:
+ cmaj Patches:
+ patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt
+ uploaded by cmaj (license 830) Tested by: cmaj ........
+
+2011-05-13 16:30 +0000 [r318869] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 318868 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r318868 | rmudgett | 2011-05-13 11:28:26 -0500 (Fri, 13 May 2011)
+ | 19 lines CDR's are being written immediately on caller hangup.
+ CDR's are being written immediately on caller hangup. The
+ dialplan is not able to modify it in the h exten. The h exten in
+ the initial context is not run before closing CDR's when the
+ bridge is unlinked if a macro is active and does not have an h
+ exten. * Make ast_bridge_call() check for an h exten in the
+ current context and if a macro is active then the initial
+ context. The first h exten found is then run before closing the
+ CDR. (closes issue #18212) Reported by: leearcher Patches:
+ issue18212_v1.8.patch uploaded by rmudgett (license 664) Tested
+ by: rmudgett Review: https://reviewboard.asterisk.org/r/1206/
+ ........
+
+2011-05-13 08:33 +0000 [r318833] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Move exten used for dialing from device
+ to subchannel. There were some issues where if a simple switch
+ was cancelled and a new switch started before the first had timed
+ out where the d->exten would be used for both subchannels. This
+ was bad leading to possible invalid extensions if some digits had
+ been entered in the abandoned simple switch and the second one
+ was completed before the first timed out, or the second would be
+ cancelled because d->exten would be set to nothing on the time
+ out of the first.
+
+2011-05-13 01:55 +0000 [r318785] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/sip/reqresp_parser.c: Merged revisions 318720 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r318720 | mnicholson | 2011-05-12 18:35:51 -0500 (Thu, 12 May
+ 2011) | 4 lines Handle ipv6 addresses in the sent-by Via: field.
+ This change fixes a regression in via header parsing and ipv6
+ handling. (closes issue #18951) ........
+
+2011-05-13 01:50 +0000 [r318784] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 318783 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r318783 | rmudgett | 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011)
+ | 14 lines PRI early media won't ring. And another way to pass
+ early media. Don't indicate that there is inband information
+ present, just assume that the B channel is connected. * Restore
+ clearing the dialing flag Rx squelch unconditionally when a
+ PROCEEDING message comes in. (closes issue #19268) Reported by:
+ tbsky Patches: issue19268_v1.8.patch uploaded by rmudgett
+ (license 664) Tested by: tbsky ........
+
+2011-05-12 22:56 +0000 [r318672] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_sip.c, apps/app_directed_pickup.c,
+ main/features.c, include/asterisk/features.h: Merged revisions
+ 318671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May
+ 2011) | 30 lines Fix directed group pickup feature code *8 with
+ pickupsounds enabled Since 1.6.2, the new pickupsound and
+ pickupfailsound in features.conf cause many issues. 1).
+ chan_sip:handle_request_invite() shouldn't be playing out the
+ fail/success audio, as it has 'netlock' locked. 2). dialplan
+ applications for directed_pickups shouldn't beep. 3). feature
+ code for directed pickup should beep on success/failure if
+ configured. Created a sip_pickup() thread to handle the pickup
+ and playout the audio, spawned from handle_request_invite. Moved
+ app_directed:pickup_do() to features:ast_do_pickup(). Functions
+ below, all now use the new ast_do_pickup() app_directed_pickup.c:
+ pickup_by_channel() pickup_by_exten() pickup_by_mark()
+ pickup_by_part() features.c: ast_pickup_call() (closes issue
+ #18654) Reported by: Docent Patches:
+ ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license
+ 585) Tested by: lmadsen, francesco_r, amilcar, isis242,
+ alecdavis, irroot, rymkus, loloski, rmudgett Review:
+ https://reviewboard.asterisk.org/r/1185/ ........
+
+2011-05-12 20:44 +0000 [r318600-318635] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Consolidate setsubstate_* into
+ setsubstate and use a switch. Consolidate the functions and add
+ some debugging info. Allows to be able to set a substate without
+ explicitly knowing what the state is.
+
+ * channels/chan_skinny.c: Add setsubstate_onhook. Add the
+ setsubstate_onhook to complete the initial substate handling
+ procedures. Added dumpsub(sub, forcehangup) which is the common
+ way of calling setsubstate_onhook. Dumpsub attempts to activate
+ another sub after setting the current one onhook.
+
+2011-05-11 18:52 +0000 [r318551-318552] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 318550 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r318550 | twilson | 2011-05-11 13:47:33 -0500 (Wed, 11 May 2011)
+ | 2 lines Comment out the REF_DEBUG that slipped in during
+ debugging ........
+
+ * /, channels/chan_sip.c: Merged revisions 318549 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r318549 | twilson | 2011-05-11 13:39:48 -0500
+ (Wed, 11 May 2011) | 27 lines Merged revisions 318548 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011)
+ | 19 lines Clean up several chan_sip reference leaks Several
+ situations in the code could lead to peers or sip_pvt references
+ being leaked. This would cause RTP ports to never be destroyed
+ (leading to exhaustion of all available RTP ports) and memory
+ leaks. The original patch for this issue from rgagnon was the
+ result of an obscene amount of testing and hard work, for which I
+ am very grateful. I did some cleanup and added a few additional
+ refcount fixes that I found. (closes issue #17255) Reported by:
+ kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff
+ uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson,
+ wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/
+ Review: https://reviewboard.asterisk.org/r/1207/ Review:
+ https://reviewboard.asterisk.org/r/1210/ ........
+ ................
+
+2011-05-10 23:42 +0000 [r318500] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c, channels/sig_ss7.c: Merged revisions
+ 318499 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r318499 | rmudgett | 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011)
+ | 15 lines Unable to pickup DAHDI/PRI call because call state is
+ reported as DIALING. The channel state is not updated to RINGING
+ when an ALERTING message is received. Regression caused when
+ sig_pri.c (also sig_ss7.c) extracted from chan_dahdi.c. * Added
+ missing channel state update to RINGING when the
+ AST_CONTROL_RINGING frame is queued for ISDN and SS7. (closes
+ issue #19257) Reported by: alecdavis Patches:
+ issue19257_v1.8_v2.patch uploaded by rmudgett (license 664)
+ Tested by: alecdavis, rmudgett ........
+
+2011-05-10 15:16 +0000 [r318437] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 318436 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r318436 | russell | 2011-05-10 10:13:16 -0500 (Tue, 10
+ May 2011) | 2 lines chan_iax2: change LOG_NOTICE to LOG_DEBUG in
+ iax2_read(). ........
+
+2011-05-10 00:22 +0000 [r318400] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 318337 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r318337 | twilson | 2011-05-09 15:23:15 -0500
+ (Mon, 09 May 2011) | 18 lines Merged revisions 318331 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011)
+ | 12 lines Don't offer video to directmedia callee unless caller
+ offered it as well Make sure that when directmedia is enabled,
+ that video is not offered to the callee even if it supports it.
+ p->vrtp will not exist since the caller didn't offer video.
+ (closes issue #19195) Reported by: one47 Patches:
+ sip_cant_add_video_rtp uploaded by one47 (license 23) ........
+ ................
+
+2011-05-09 23:16 +0000 [r318283-318352] Richard Mudgett <rmudgett@digium.com>
+
+ * /, res/Makefile, res/res_features.exports.in (removed): Merged
+ revisions 318351 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r318351 | rmudgett | 2011-05-09 18:15:32 -0500 (Mon, 09 May 2011)
+ | 6 lines Remove references to res_features and its export file.
+ The contents of res/res_features.c was moved to into
+ main/features.c awhile ago. There is no longer any need for the
+ res/Makefile to reference res_features or the res_features linker
+ exports file to exist. ........
+
+ * /, main/features.c: Merged revisions 318282 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r318282 | rmudgett | 2011-05-09 14:07:01 -0500 (Mon, 09 May 2011)
+ | 24 lines Hangup extension executed twice. When a user hangs up
+ a call, in certain circumstances, the hangup extension can end up
+ being executed twice: 1) If a call is bridged and the 'h'
+ extension executes the Hangup application, then the 'h' extension
+ will be executed twice. 2) If a call is bridged within a macro
+ (Dial or Queue), it has its own 'h' extension, the main context
+ also has an 'h' extension, and the macro 'h' extension executes
+ the Hangup application, then both 'h' extensions will be
+ executed. * Revert originally commited fix for #16106 and just
+ set AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in
+ ast_bridge_call(). The bridge code just executed an 'h' extension
+ so the main PBX loop does not need to execute one as well. (issue
+ #16106) Reported by: ajohnson (issue #16548) Reported by: hajekd
+ ........
+
+2011-05-09 17:13 +0000 [r318234] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 318233 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r318233 | dvossel | 2011-05-09 12:09:55 -0500
+ (Mon, 09 May 2011) | 14 lines Merged revisions 318230 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r318230 | dvossel | 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011)
+ | 7 lines Fixes cases where sip_set_rtp_peer can return too early
+ during media path reset. (closes issue #19225) Reported by: one47
+ Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23)
+ ........ ................
+
+2011-05-09 17:00 +0000 [r318232] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 318231 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r318231 | rmudgett | 2011-05-09 11:57:18 -0500
+ (Mon, 09 May 2011) | 41 lines Don't get early media for ISDN on
+ outgoing calls. It looks to be a long-standing misinterpretation
+ of the progress indicator ie values: 1 - Call is not end-to-end
+ ISDN; further call progress information may be available in-band.
+ 8 - In-band information or an appropriate pattern is now
+ available. Only value 8 is handled by chan_dahdi/sig_pri. The 1
+ value is not handled as early media probably because the meaning
+ of the second half of it's description was overlooked. * Test to
+ see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or
+ PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path.
+ (closes issue #18868) Reported by: isrl Patches:
+ issue18868_19246_v1.8.patch uploaded by rmudgett (license 664)
+ Tested by: satish_lx .......... No inband progress on
+ PRI_EVENT_RINGING even if inband flag set. My ISDN-PRI provider
+ sends an ALERTING with "Inband information or appropriate pattern
+ now available", but Asterisk only generates and passes the RING
+ to the SIP extension, not the inband message. Unfortunately, the
+ inband message is not a ringback tone but a prompt that says the
+ number is not in service. The SIP extension then hears two rings
+ and the call is hungup which confuses the caller. * Post an
+ AST_CONTROL_PROGRESS as well as opening the media path if inband
+ audio is indicated with an ALERTING message. (closes issue
+ #19246) Reported by: cristiandimache Patches:
+ issue19246_v1.8.patch uploaded by rmudgett (license 664) Tested
+ by: cristiandimache ................
+
+2011-05-09 14:41 +0000 [r318194] Leif Madsen <lmadsen@digium.com>
+
+ * main/app.c: Increase prepend filename length. (closes issue
+ #19238) Reported by: byronclark Patches:
+ increase_prepend_filename_length.patch uploaded by byronclark
+ (license 1200)
+
+2011-05-09 14:37 +0000 [r318162-318193] Jonathan Rose <jrose@digium.com>
+
+ * main/features.c: Minor change to 318141 to improve parsing
+ behavior.
+
+ * /, configs/features.conf.sample: Merged revisions 318148 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r318148 | jrose | 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) |
+ 4 lines Documenting an observed behavior of features in
+ features.conf. Since parkinglots use an integer for the
+ parkinglot extensions, leading zeros specified in the
+ configuration file are ignored. ........
+
+2011-05-09 14:11 +0000 [r318143] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 318142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r318142 | mnicholson | 2011-05-09 09:09:38 -0500 (Mon, 09 May
+ 2011) | 9 lines Make indicate/control frames WRITE events on
+ framehooks. Also, if a framehook returns a non-control frame,
+ don't forward it to the channel. (closes issue #19251) Reported
+ by: irroot Patches: (modified) framehook_indicate.patch2 uploaded
+ by irroot (license 52) Tested by: irroot ........
+
+2011-05-09 13:56 +0000 [r318141] Jonathan Rose <jrose@digium.com>
+
+ * main/features.c, CHANGES: Allows ParkedCall application to
+ specify a parkinglot. When invoking the app parkedcall, the
+ argument can now include '@parkinglot' after the extension.
+ (closes issue #18777) Reported by: cartama Patches: 0018777.diff
+ uploaded by cartama (license 1157) Review:
+ https://reviewboard.asterisk.org/r/1209/
+
+2011-05-09 07:40 +0000 [r318106] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Add setsubstate_callwait. If a call is
+ made to a line that already has a call and the device is offhook
+ (ie activeish call), the call is set to CALLWAIT rather than
+ RINGIN.
+
+2011-05-07 23:36 +0000 [r318056-318058] Russell Bryant <russell@digium.com>
+
+ * res/res_config_curl.c, /: Merged revisions 318057 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r318057 | russell | 2011-05-07 18:35:37 -0500 (Sat, 07
+ May 2011) | 8 lines res_config_curl: fix a crash with static
+ realtime. (closes issue #18413) Reported by: jmls Patches:
+ 20101202__issue18413.diff.txt uploaded by tilghman (license 14)
+ Tested by: jmls ........
+
+ * /, channels/chan_iax2.c: Merged revisions 318055 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r318055 | russell | 2011-05-07 18:24:18 -0500 (Sat, 07
+ May 2011) | 7 lines chan_iax2: Don't overwrite port found with an
+ SRV lookup. (closes issue #17291) Reported by: jcovert Patches:
+ chan_iax2.c.1.8.3-srvlookup-corrected.patch uploaded by jcovert
+ (license 551) ........
+
+2011-05-06 23:07 +0000 [r317996-318019] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Only allow voicemail if substate is
+ OFFHOOK or no channel active (UNSET). (closes issue #17901)
+ Reported by: salecha
+
+ * channels/chan_skinny.c: Rename sub->parent to sub->line. Improve
+ readability of code, eg, (sub->parent == d->activeline) becomes
+ (sub->line == d->activeline).
+
+ * channels/chan_skinny.c: Move the hookstate from line to device.
+ Long time coming, finally moving the hookstate from line to
+ device. This may fix some issues where a device has multiple
+ lines. Previously we had to run through all lines on a device to
+ see if it was actually onhook or not.
+
+2011-05-06 21:49 +0000 [r317968-317970] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 317969 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317969 | russell | 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011)
+ | 10 lines Use the right variable to print the time in a debug
+ message. The original patch also increased some buffer sizes, but
+ that was already done in this version. (closes issue #17034)
+ Reported by: sysreq Patches: asterisk-issue-17034.patch uploaded
+ by sysreq (license 1009) ........
+
+ * /, apps/app_meetme.c: Merged revisions 317967 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317967 | russell | 2011-05-06 16:38:54 -0500 (Fri, 06 May 2011)
+ | 2 lines Fix some more "set but unused" compiler warnings.
+ ........
+
+2011-05-06 21:10 +0000 [r317920] David Vossel <dvossel@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Merged revisions 317918 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r317918 | dvossel | 2011-05-06 16:06:55 -0500 (Fri, 06
+ May 2011) | 7 lines Fixes missing colon from To/From headers in
+ RTCP manager events. (closes issue #18221) Reported by:
+ clegall_proformatique Patches: 18221_1.patch uploaded by ebroad
+ (license 878) ........
+
+2011-05-06 21:07 +0000 [r317843-317919] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, /: Merged revisions 317917 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317917 | russell | 2011-05-06 16:06:33 -0500 (Fri, 06 May 2011)
+ | 7 lines Fix calculation of free RAM to make minmemfree option
+ work. (closes issue #17124) Reported by: loic Patches: pbx_c.diff
+ uploaded by loic (license 1020) ........
+
+ * contrib/scripts/import-cdr-csv-mysql.pl (added): Add a cdr_csv to
+ MySQL import script to contrib/scripts. (closes issue #17036)
+ Reported by: precisenetworks Patches: import-cdr-csv-mysql.pl
+ uploaded by precisenetworks (license 1010)
+
+ * apps/app_userevent.c, CHANGES: Add the Uniqueid header to
+ Userevent. (closes issue #16962) Reported by: jlpedrosa Patches:
+ patch.diff uploaded by jlpedrosa (license 1002)
+
+ * /, channels/chan_sip.c: Merged revisions 317867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317867 | russell | 2011-05-06 15:01:16 -0500 (Fri, 06 May 2011)
+ | 10 lines chan_sip: Destroy variables on a sip_pvt before
+ copying vars from the sip_peer. Don't duplicate variables on the
+ sip_pvt. Just reset the variable list each time. (closes issue
+ #19202) Reported by: wdoekes Patches:
+ issue19202_destroy_challenged_invite_chanvars.patch uploaded by
+ wdoekes (license 717) ........
+
+ * /, channels/chan_sip.c: Merged revisions 317865 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317865 | russell | 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011)
+ | 11 lines chan_sip: fix a deadlock in check_rtp_timeout. Don't
+ block doing silly deadlock avoidance. Just return and try again
+ later. The funciton gets called often enough that it's fine.
+ Also, this change was already made in trunk. (closes issue
+ #18791) Reported by: irroot Patches: chan_sip.rtptimeout.patch
+ uploaded by irroot (license 52) ........
+
+ * addons/app_mysql.c, /: Merged revisions 317837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317837 | russell | 2011-05-06 14:24:11 -0500 (Fri, 06 May 2011)
+ | 11 lines Fix a crash in the MySQL() application. This code was
+ not handling channel datastores safely. The channel must be
+ locked. (closes issue #17964) Reported by: wuwu Patches:
+ issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license
+ 71) Tested by: wuwu ........
+
+2011-05-06 19:23 +0000 [r317818-317833] Matthew Nicholson <mnicholson@digium.com>
+
+ * CHANGES: Updated CHANGES to note the autoservice changes for
+ pbx_lua
+
+ * configs/extensions.lua.sample: Updated the sample pbx_lua config
+ file to reflect autoservice changes.
+
+2011-05-06 19:15 +0000 [r317807] Russell Bryant <russell@digium.com>
+
+ * /, contrib/realtime/mysql/sipfriends.sql: Merged revisions 317805
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317805 | russell | 2011-05-06 14:14:39 -0500 (Fri, 06 May 2011)
+ | 7 lines Add a new sipfriends.sql for MySQL that has more fields
+ in it. (closes issue #16399) Reported by: pabelanger Patches:
+ sipfriends.sql.v3 uploaded by pabelanger (license 224) ........
+
+2011-05-06 19:14 +0000 [r317721-317806] Matthew Nicholson <mnicholson@digium.com>
+
+ * pbx/pbx_lua.c, UPGRADE.txt: Default to starting an autoservice in
+ pbx_lua. The autoservice is automatically stopped when
+ applications are executed, so this shouldn't cause any problems.
+
+ * pbx/pbx_lua.c, UPGRADE.txt: Make pbx_lua handle managing the
+ autoservice better. Make autoservice_start() and
+ autoservice_stop() return nothing. Also check if the autoservice
+ flag is set before starting or stopping the autoservice and stop
+ and start the autoservice when returning control to and getting
+ control from the pbx engine.
+
+ * UPGRADE.txt: Added note about changes in pbx_lua's behavior when
+ applications do dialplan jumps
+
+ * CHANGES: Use two spaces after periods for the recent pbx_lua
+ change descriptions
+
+ * CHANGES: Updated CHANGES for hints support in pbx_lua
+
+ * pbx/pbx_lua.c, CHANGES: Detect Goto in pbx_lua. This code will
+ actually detect any dialplan jump from any application that calls
+ ast_explicit_goto(). This change is only being done in trunk as
+ it may change the way some dialplans execute.
+
+2011-05-06 16:23 +0000 [r317671] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 317670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317670 | rmudgett | 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011)
+ | 22 lines Fix SIP connected line updates. This patch fixes a
+ couple SIP connected line update problems: 1) The connected line
+ needs to be updated when the initial INVITE is sent if there is a
+ peer callerid configured. Previously, the connected line
+ information did not get reported until the call was connected so
+ SIP could not report connected line information in ringing or
+ progress messages. 2) The connected line should not be updated on
+ initial connect if there is no connected line information.
+ Previously, all it did was wipe out any default preset
+ CONNECTEDLINE information set by the dialplan with empty strings.
+ (closes issue #18367) Reported by: GeorgeKonopacki Patches:
+ issue18367_v1.8.patch uploaded by rmudgett (license 664) Tested
+ by: rmudgett Review: https://reviewboard.asterisk.org/r/1199/
+ ........
+
+2011-05-06 08:21 +0000 [r317596] Terry Wilson <twilson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 317584 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r317584 | twilson | 2011-05-06 01:18:53 -0700
+ (Fri, 06 May 2011) | 20 lines Merged revisions 317575 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r317575 | twilson | 2011-05-06 01:04:17 -0700
+ (Fri, 06 May 2011) | 13 lines Merged revisions 317574 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011)
+ | 6 lines Re-fix queue round-robin This part of the change for
+ r315596 was incorrect. No bridge occurs when doing a roundrobin
+ dial and no one answers, so this code shouldn't have been
+ removed. ........ ................ ................
+
+2011-05-05 23:47 +0000 [r317426-317531] Russell Bryant <russell@digium.com>
+
+ * Makefile, /: Merged revisions 317530 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317530 | russell | 2011-05-05 18:46:54 -0500 (Thu, 05 May 2011)
+ | 10 lines If the configure script runs, force a rebuild of
+ menuselect-tree. Some contents in the menuselect tree are
+ dependent on configure script parameters, namely
+ --enable-dev-mode. (closes issue #17219) Reported by: Nick_Lewis
+ Patches: issue_17219.rev1.txt uploaded by russell (license 2)
+ ........
+
+ * /, contrib/realtime/mysql/queue_log.sql,
+ contrib/realtime/mysql/sipfriends.sql: Merged revisions 317486
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317486 | russell | 2011-05-05 18:15:53 -0500 (Thu, 05 May 2011)
+ | 9 lines Fix some more realtime MySQL schema issues. (closes
+ issue #18537) Reported by: denzs Patches: sipfriends.sql.svndiff
+ uploaded by denzs (license 1182) queue_log.sql.svndiff uploaded
+ by denzs (license 1182) meetme.sql.svndiff uploaded by denzs
+ (license 1182) ........
+
+ * /, contrib/realtime/mysql/meetme.sql,
+ contrib/realtime/mysql/sipfriends.sql: Merged revisions 317484
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317484 | russell | 2011-05-05 18:12:35 -0500 (Thu, 05 May 2011)
+ | 8 lines Fix some errors in sample MySQL realtime schema files.
+ (closes issue #18915) Reported by: Dovid Patches:
+ sipfriends.patch uploaded by Dovid (license 652) meetme.patch
+ uploaded by Dovid (license 652) ........
+
+ * CHANGES, res/res_calendar.c: Add "calendar show types" CLI
+ command. (closes issue #18246) Reported by: junky Patches:
+ calendar_types.diff uploaded by junky (license 177)
+
+ * cel/cel_pgsql.c, UPGRADE.txt, configs/cel_pgsql.conf.sample,
+ CHANGES: Add CEL extra field to cel_pgsql. (closes issue #18462)
+ Reported by: joscas Patches: bug_18462.diff uploaded by snuffy
+ (license 35) cel_pgsql.conf.sample.issue18462.patch uploaded by
+ joscas (license 1180)
+
+ * /, cdr/cdr_syslog.c: Merged revisions 317480 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317480 | russell | 2011-05-05 18:00:55 -0500 (Thu, 05 May 2011)
+ | 8 lines Don't lose cdr_syslog config on a reload. (closes issue
+ #18679) Reported by: enegaard Patches:
+ issue18679_seanbright.patch uploaded by seanbright (license 71)
+ Tested by: enegaard ........
+
+ * channels/chan_unistim.c, channels/chan_usbradio.c,
+ channels/chan_dahdi.c, /, channels/chan_sip.c,
+ channels/chan_skinny.c, channels/chan_h323.c,
+ channels/chan_alsa.c, channels/chan_console.c,
+ channels/chan_oss.c, channels/chan_mgcp.c,
+ channels/misdn_config.c: Merged revisions 317478 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05
+ May 2011) | 12 lines Fix some consistency issues with
+ jitterbuffer config. Store the defaults noted in the sample
+ config files in the jitterbuffer config data structure. This
+ makes the CLI commands that output these settings show the right
+ thing. Also only show the settings that are relevant in the
+ settings CLI commands, based on which jitterbuffer is selected
+ and whether it's enabled. (closes issue #19083) Reported by:
+ rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by
+ rgagnon (license 1202) ........
+
+ * /, pbx/pbx_lua.c: Merged revisions 317476 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317476 | russell | 2011-05-05 17:47:57 -0500 (Thu, 05 May 2011)
+ | 8 lines Add a datastore fixup to fix a pbx_lua crash. (closes
+ issue #19055) Reported by: jamhed Patches:
+ lua_datastore_fixup1.diff uploaded by mnicholson (license 96)
+ Tested by: mnicholson, jamhed ........
+
+ * cel/cel_pgsql.c, channels/chan_jingle.c,
+ channels/sip/sdp_crypto.c, res/res_config_odbc.c, /,
+ channels/chan_sip.c, res/res_crypto.c, pbx/pbx_lua.c,
+ channels/iax2-provision.c, pbx/pbx_dundi.c,
+ channels/chan_console.c, cdr/cdr_radius.c, channels/chan_iax2.c,
+ res/res_jabber.c, res/res_config_sqlite.c: Merged revisions
+ 317474 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317474 | russell | 2011-05-05 17:36:33 -0500 (Thu, 05 May 2011)
+ | 2 lines Fix more "set but unused" warnings. ........
+
+ * /, main/dsp.c: Merged revisions 317429 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317429 | russell | 2011-05-05 17:11:19 -0500 (Thu, 05 May 2011)
+ | 5 lines Only display inband DTMF warning if inband DTMF
+ detection is enabled. (closes issue #18901) Reported by: irroot
+ ........
+
+ * /, apps/app_rpt.c: Merged revisions 317427 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317427 | russell | 2011-05-05 16:58:45 -0500 (Thu, 05 May 2011)
+ | 7 lines Fix potential memory leak, and use of uninitialized
+ memory. (closes issue #16476) Reported by: junky Patches:
+ M16476.diff uploaded by junky (license 177) ........
+
+ * main/manager.c, /: Merged revisions 317425 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317425 | russell | 2011-05-05 16:53:13 -0500 (Thu, 05 May 2011)
+ | 7 lines Add missing ActioID handling to Events action. (closes
+ issue #18949) Reported by: edersohe Patches: 0018949.patch
+ uploaded by edersohe (license 1228) ........
+
+2011-05-05 21:20 +0000 [r317395] Sean Bright <sean@malleable.com>
+
+ * main/asterisk.c: Add some new editline bindings by default, and
+ allow for user specified configuration. I excluded the part of
+ this patch that used the HOME environment variable since the
+ built-in editline library goes to great lengths to disallow that.
+ Instead only settings the EDITRC environment variable will use a
+ user specified file. Also, the default environment variable use
+ to determine the edit more is AST_EDITMODE instead of AST_EDITOR
+ (although the latter is still supported). (closes issue #15929)
+ Reported by: kkm Patches: astcli-editrc-v2.diff uploaded by kkm
+ (license 888) 015929-astcli-editrc-trunk.240324.diff uploaded by
+ kkm (license 888) Tested by: seanbright
+
+2011-05-05 20:46 +0000 [r317382] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Move hold stuff to the setsubstate
+ arrangement. skinny_hold moved to setsubstate_hold and
+ skinny_unhold integrated into setsubstate_connected. Removed
+ sub->onhold and replaced with SUBSTATE_HOLD. Also fixed inbound
+ call answering by queueing an AST_CONTROL_ANSWER on answering a
+ SUBSTATE_RINGIN sub (was a typo).
+
+2011-05-05 20:27 +0000 [r317377] Sean Bright <sean@malleable.com>
+
+ * /, addons/res_config_mysql.c: Merged revisions 317370 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317370 | seanbright | 2011-05-05 16:25:52 -0400 (Thu, 05 May
+ 2011) | 10 lines Don't duplicate our data on the stack and just
+ use the MYSQL_ROW directly. With large result sets we were
+ blowing out the stack. (closes issue #19090) Reported by:
+ mickecarlsson Patches: issue19090_trunk_svn.patch uploaded by
+ seanbright (license 71) Tested by: mickecarlsson ........
+
+2011-05-05 19:56 +0000 [r317337] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 317336 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317336 | russell | 2011-05-05 14:55:58 -0500 (Thu, 05 May 2011)
+ | 7 lines Increase buffer size to be PATH_MAX for a path. (closes
+ issue #19239) Reported by: byronclark Patches:
+ queue_announce_length.patch uploaded by byronclark (license 1200)
+ ........
+
+2011-05-05 19:33 +0000 [r317334] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 317283 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317283 | jrose | 2011-05-05 14:09:13 -0500 (Thu, 05 May 2011) |
+ 10 lines Resolves a deadlock that occurs during sip_new This is
+ based on an uncommitted patch by jpeeler for the issue. Instead
+ of relocking and then unlocking the channel though, we keep the
+ lock on the channel until we are finished doing what we need to
+ the channel. (closes issue #18441) Reported by: Alric ........
+
+2011-05-05 18:46 +0000 [r317282] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 317281 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r317281 | russell | 2011-05-05 13:39:44 -0500
+ (Thu, 05 May 2011) | 29 lines Merged revisions 317255 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r317255 | russell | 2011-05-05 13:29:53 -0500
+ (Thu, 05 May 2011) | 22 lines Merged revisions 317211 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011)
+ | 15 lines chan_sip: fix broken realtime peer count, fix memory
+ leak This patch addresses two bugs in chan_sip: 1) The count of
+ realtime peers and users was off. The increment checked the value
+ of the caching option, while the decrement did not. 2) Add a
+ missing regfree() for a regex. (closes issue #19108) Reported by:
+ vrban Patches: missing_regfree.patch uploaded by vrban (license
+ 756) sip_object_counter.patch uploaded by vrban (license 756)
+ ........ ................ ................
+
+2011-05-05 18:09 +0000 [r317198] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 317196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317196 | mnicholson | 2011-05-05 13:02:52 -0500 (Thu, 05 May
+ 2011) | 8 lines Set SO_KEEPALIVE on SIP TCP sockets so that they
+ eventually go away when a peer abruptly disappears. This mostly
+ occurs after a successful registration. (closes issue #17544)
+ Reported by: marcelloceschia Patches: (modified) tcptls.patch
+ uploaded by st (license 907) ........
+
+2011-05-05 18:08 +0000 [r317197] David Vossel <dvossel@digium.com>
+
+ * bridges/bridge_softmix.c, funcs/func_jitterbuffer.c: Fixes
+ reliability issues with func_jitterbuffer's usage in the new
+ ConfBridge application.
+
+2011-05-05 15:06 +0000 [r317059-317105] Leif Madsen <lmadsen@digium.com>
+
+ * /, contrib/scripts/safe_asterisk: Merged revisions 317104 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r317104 | lmadsen | 2011-05-05 11:04:24 -0400
+ (Thu, 05 May 2011) | 15 lines Merged revisions 317102 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011)
+ | 8 lines Disable console colourization inside safe_asterisk
+ checks. (closes issue #19213) Reported by: lefoyer Patches:
+ issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by
+ wdoekes (license 717) Tested by: wdoekes, lefoyer ........
+ ................
+
+ * Makefile, configs/cel.conf.sample, /: Merged revisions 317058 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r317058 | lmadsen | 2011-05-05 08:27:56 -0400 (Thu, 05 May 2011)
+ | 7 lines Remove unused directory and clear up some
+ documentation. (closes issue #19193) Reported by: bchia Patches:
+ cel-csv.diff uploaded by lathama (license 1028) Tested by:
+ lathama, Marquis42 ........
+
+2011-05-05 09:03 +0000 [r316994-317026] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Add setsubstate_congestion and
+ setsubstate_progress. Move handling of both state handling from
+ skinny_indicate to it's own sub. Also, modified behaviour to not
+ hangup the sub and let the dialplan have a chance in doing what
+ it wants for congestion. Added various states to substate2str and
+ added these states where applicable for other set_substate_
+ procs.
+
+ * channels/chan_skinny.c: Add setsubstate_busy. Move handling of
+ setting busy state from skinny_indicate to it's own sub. Also,
+ modified behaviour to not hangup the sub and let the dialplan
+ have a chance in doing what it wants (eg busy(10); hangup() in
+ the dialplan now gives a busy indication for 10 secs and then
+ hangs up.
+
+2011-05-05 07:09 +0000 [r316962] Stefan Schmidt <sst@sil.at>
+
+ * main/astobj2.c: Adding the Move to Front Hash functionality
+ Moving a found object to the front of its bucket to reduce the
+ necessary traversal steps to find an object. This change improves
+ the search time on large system with many data or in link lists.
+ (closes issue #19233) Reported by: schmidts Review:
+ https://reviewboard.asterisk.org/r/1201/
+
+2011-05-05 02:34 +0000 [r316920] Sean Bright <sean@malleable.com>
+
+ * main/manager.c, /, main/http.c, main/utils.c: Merged revisions
+ 316917-316919 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r316917 | seanbright | 2011-05-04 22:23:28 -0400 (Wed, 04 May
+ 2011) | 5 lines Make sure that tcptls_session is properly
+ initialized. (issue #18598) Reported by: ksn ........ r316918 |
+ seanbright | 2011-05-04 22:25:20 -0400 (Wed, 04 May 2011) | 5
+ lines Look at the correct buffer for our digest info instead of
+ an empty one. (issue #18598) Reported by: ksn ........ r316919 |
+ seanbright | 2011-05-04 22:30:45 -0400 (Wed, 04 May 2011) | 10
+ lines Use the correct HTTP method when generating our digest,
+ otherwise we always fail. When calculating the 'A2' portion of
+ our digest for verification, we need the HTTP method that is
+ currently in use. Unfortunately our mapping function was
+ incorrect, resulting in invalid hashes being generated and, in
+ turn, failures in authentication. (closes issue #18598) Reported
+ by: ksn ........
+
+2011-05-04 21:44 +0000 [r316885] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Add setsubstate_ringout (equivalent to
+ AST_STATE ringing). Renamed previous setsubstate_ringout to
+ setsubstate_dialing for a state when attempting to dial a number,
+ substate ringout now for when core has indicated that the channel
+ is actually ringing on the other end. Also added substate2str for
+ debugging purposes.
+
+2011-05-04 18:57 +0000 [r316832] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 316831 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r316831 | rmudgett | 2011-05-04 13:51:40 -0500 (Wed, 04 May 2011)
+ | 9 lines Wait for leader with Music On Hold allows crosstalk
+ between participants. Parenthesis in the wrong position.
+ Regression from issue #14365 when expanding conference flags to
+ use 64 bits. (closes issue #18418) Reported by: MrHanMan Tested
+ by: rmudgett ........
+
+2011-05-04 16:42 +0000 [r316798] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, CHANGES: Reverts rev 316218 as it breaks
+ parsing the [general] section of sip.conf. The functionality this
+ patch attempts to achieve should already be possible using
+ [general](+) in the config file. issue #17957
+
+2011-05-04 16:17 +0000 [r316664-316711] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 316709 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r316709 | seanbright | 2011-05-04 12:15:32 -0400
+ (Wed, 04 May 2011) | 22 lines Merged revisions 316708 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r316708 | seanbright | 2011-05-04 12:10:59 -0400
+ (Wed, 04 May 2011) | 15 lines Merged revisions 316707 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed, 04 May
+ 2011) | 8 lines If sox fails when processing a voicemail, don't
+ delete the original file. (closes issue #18111) Reported by:
+ sysreq Patches: issue18111_trunk.patch uploaded by seanbright
+ (license 71) Tested by: seanbright ........ ................
+ ................
+
+ * main/manager.c, /: Merged revisions 316663 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r316663 | seanbright | 2011-05-04 10:35:05 -0400 (Wed, 04 May
+ 2011) | 8 lines Only return a single error via AMI when
+ requesting a forbidden action. (closes issue #19216) Reported by:
+ oej Patches: issue19216-1.8-r316204.patch uploaded by seanbright
+ (license 71) Tested by: seanbright ........
+
+2011-05-04 14:26 +0000 [r316618-316657] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 316650 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r316650 | dvossel | 2011-05-04 09:25:03 -0500
+ (Wed, 04 May 2011) | 15 lines Merged revisions 316644 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r316644 | dvossel | 2011-05-04 09:23:39 -0500 (Wed, 04 May 2011)
+ | 9 lines Fixes one-way-audio when chanspy activated with the 'o'
+ option (closes issue #18382) Reported by: jkister Patches:
+ 0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt
+ uploaded by malin (license ) Tested by: firstsip, Greenlightcrm,
+ malin, wdoekes, boroda, dvossel ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 316617 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r316617 | dvossel | 2011-05-04 08:44:41 -0500
+ (Wed, 04 May 2011) | 19 lines Merged revisions 316616 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r316616 | dvossel | 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011)
+ | 12 lines Fixes session-timers=refuse not being enforced for
+ *caller* During handle_request_invite, the session timer mode was
+ retrieved from a cached variable. This patch forces a peer lookup
+ of the session timer mode in the case of an incoming invite.
+ (closes issue #18804) Reported by: wdoekes Patches:
+ issue18804_session_timer_refuse_caller.patch uploaded by wdoekes
+ (license 717) issue_18804_v2.diff uploaded by dvossel (license
+ 671) ........ ................
+
+2011-05-04 08:25 +0000 [r316552-316584] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Add setsubstate_ringin. Added
+ setsubstate_ringin. skinny_call now calls sss_ringin rather than
+ inline. Fixed previous issue so that setsubstate_connected now
+ use SUBSTATE_RINGIN to determine is an AST_CONTROL_ANSWER should
+ be queued.
+
+ * channels/chan_skinny.c: Make skinny_answer use
+ setsubsate_connected. Cosolidated the code so that skinny_answer
+ now uses the setsubstate procedures rather than doing the
+ handling inline.
+
+2011-05-04 07:13 +0000 [r316520] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * autoconf/ast_check_pwlib.m4, /, configure: Merged revisions
+ 316193 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r316193 | tzafrir | 2011-05-03 13:57:16 +0300 (ג', 03 מאי 2011) |
+ 8 lines Re-fix bashism in ./configure: s/let/$(( ))/ A
+ forward-port in r278985 accidentally re-introduced issue 17485.
+ Fixing it. Thanks to Jilles Tjoelker for the good report. (closes
+ issue #17485) ........
+
+2011-05-04 07:10 +0000 [r316519] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Cleanup skinny callinfo. Cosolidated the
+ working out of the callinfo to be sent into transmit_callinfo.
+ Replaced ambiguous sub->outgoing with calldirection which can be
+ SKINNY_INCOMING or SKINNY_OUTGOING (same value as the skinny
+ protocol).
+
+2011-05-04 02:39 +0000 [r316477] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_meetme.c: Merged revisions 316476 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r316476 | seanbright | 2011-05-03 22:34:01 -0400
+ (Tue, 03 May 2011) | 17 lines Merged revisions 316475 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May
+ 2011) | 10 lines Honor the C option to MeetMe when L is passed.
+ This fixes a case that r304773 and friends missed. (closes issue
+ #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff
+ uploaded by var (license 1227) Tested by: seanbright ........
+ ................
+
+2011-05-04 00:13 +0000 [r316428-316430] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, addons/cdr_mysql.c, addons/res_config_mysql.c: Merged
+ revisions 316429 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r316429 | tilghman | 2011-05-03 19:12:25 -0500 (Tue, 03 May 2011)
+ | 7 lines Escape column names in case they contain illegal
+ characters ('-') or reserved words. (closes issue #19063)
+ Reported by: festr Patches: patch uploaded by festr (license 443)
+ ........
+
+ * channels/chan_sip.c, CHANGES: If multiple [general] contexts
+ occur from sip.conf (usually due to external includes), merge
+ them. The original implementation of this did the merging of all
+ contexts with the same name in the realtime layer, but that
+ implementation severely breaks drivers which use the same context
+ name (e.g. iax.conf, type={peer,user}). Therefore, the
+ implementation needs to do the merging for particular entries
+ only, based upon what contexts would allow that in the channel
+ driver itself. This implementation is for chan_sip only, but
+ others could be added in the future. (closes issue #17957)
+ Reported by: marcelloceschia Patches:
+ chan-sip_parsing-general_branch162.patch uploaded by
+ marcelloceschia (license 1079) Tested by: tilghman
+
+2011-05-03 22:16 +0000 [r316337] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_skinny.c, pbx/pbx_dundi.c, channels/chan_mgcp.c:
+ Merged revisions 316336 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r316336 | russell | 2011-05-03 17:13:31 -0500 (Tue, 03 May 2011)
+ | 8 lines Use htons() instead of ntohs() in some places. (closes
+ issue #19200) Reported by: wdoekes Patches:
+ issue19200-trunk.patch uploaded by wdoekes (license 717)
+ issue19200-1.8.x.patch uploaded by wdoekes (license 717) ........
+
+2011-05-03 22:07 +0000 [r316335] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 316334 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r316334 | dvossel | 2011-05-03 17:05:59 -0500 (Tue, 03 May 2011)
+ | 8 lines Fixes framehook segfault on indicate (closes issue
+ #19215) Reported by: irroot Patches: framehook_indicate.patch
+ uploaded by irroot (license 52) ........
+
+2011-05-03 21:48 +0000 [r316333] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_minivm.c: Merged revisions 316331 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r316331 | russell | 2011-05-03 16:41:11 -0500 (Tue, 03 May 2011)
+ | 2 lines Resolve another warning. ........
+
+2011-05-03 21:45 +0000 [r316332] David Vossel <dvossel@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 316330 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r316330 | dvossel | 2011-05-03 16:37:59 -0500
+ (Tue, 03 May 2011) | 24 lines Merged revisions 316329 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r316329 | dvossel | 2011-05-03 16:29:55 -0500
+ (Tue, 03 May 2011) | 17 lines Merged revisions 316328 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03 May 2011)
+ | 10 lines Fixes chan_local crashs in local_fixup() Thanks OEJ
+ for tracking down the issue and submitting the patch. (closes
+ issue #19053) Reported by: oej Tested by: oej Review:
+ https://reviewboard.asterisk.org/r/1158/ ........
+ ................ ................
+
+2011-05-03 20:45 +0000 [r316293] Russell Bryant <russell@digium.com>
+
+ * channels/chan_unistim.c, main/udptl.c, main/fskmodem_float.c,
+ main/rtp_engine.c, /, res/res_musiconhold.c, apps/app_ices.c,
+ apps/app_followme.c, main/config.c, main/channel.c, main/cdr.c,
+ channels/chan_phone.c, funcs/func_enum.c, main/manager.c,
+ channels/chan_skinny.c, apps/app_minivm.c, main/features.c,
+ main/plc.c, res/res_agi.c, apps/app_amd.c, main/pbx.c,
+ res/res_fax.c, formats/format_wav.c, apps/app_festival.c,
+ channels/chan_agent.c, apps/app_originate.c, apps/app_queue.c,
+ codecs/lpc10/dyptrk.c, include/asterisk/linkedlists.h,
+ main/file.c, main/audiohook.c, pbx/pbx_config.c, main/asterisk.c,
+ main/dsp.c, res/res_calendar.c, apps/app_voicemail.c: Merged
+ revisions 316265 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011)
+ | 5 lines Fix a bunch of compiler warnings generated by gcc
+ 4.6.0. Most of these are -Wunused-but-set-variable, but there
+ were a few others mixed in here, as well. ........
+
+2011-05-03 19:22 +0000 [r316240] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_pri.c: Merged revisions 316224 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r316224 | rmudgett | 2011-05-03 14:18:30 -0500 (Tue, 03 May 2011)
+ | 16 lines The dahdi_hangup() call does not clean up the channel
+ fully. After dahdi_hangup() has supposedly hungup an ISDN channel
+ there is still traffic on the S0-bus because the channel was not
+ cleaned up fully. Shuffled the hangup code to include some
+ missing cleanup. Also fixed some code formatting in the area. I
+ think the primary missing clean up code was the call to
+ tone_zone_play_tone() to turn off any active tones on the
+ channel. (closes issue #19188) Reported by: jg1234 Patches:
+ issue19188_v1.8.patch uploaded by rmudgett (license 664) Tested
+ by: jg1234 ........
+
+2011-05-03 19:00 +0000 [r316216-316218] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 316217 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r316217 | dvossel | 2011-05-03 13:59:06 -0500 (Tue, 03 May 2011)
+ | 9 lines Never put the Require: timer header in an Invite. This
+ has already been discussed and should have been resolved earlier.
+ View revsion 285565's log for more information about why it is
+ important to not put timer in the Require header. (closes issue
+ #18704) Reported by: mfrager ........
+
+ * /, res/res_odbc.c: Merged revisions 316215 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r316215 | dvossel | 2011-05-03 13:49:48 -0500 (Tue, 03 May 2011)
+ | 9 lines Fixes a random crash (NULL reference) in res_odbc.c.
+ (closes issue #19180) Reported by: pruiz Patches: tmp.diff
+ uploaded by pruiz (license 1152) Tested by: pruiz, seanbright
+ ........
+
+2011-05-03 18:23 +0000 [r316213] Sean Bright <sean@malleable.com>
+
+ * main/manager.c, /: Merged revisions 316206 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r316206 | seanbright | 2011-05-03 14:17:36 -0400 (Tue, 03 May
+ 2011) | 8 lines If we aren't interested in events, don't generate
+ the FullyBooted event on AMI login. (closes issue #19089)
+ Reported by: bklang Patches: issue19089-1.8-r316204.patch
+ uploaded by seanbright (license 71) Tested by: seanbright
+ ........
+
+2011-05-02 19:15 +0000 [r316095] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * funcs/func_curl.c, /: Merged revisions 316094 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r316094 | tilghman | 2011-05-02 14:09:55 -0500
+ (Mon, 02 May 2011) | 15 lines Merged revisions 316093 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r316093 | tilghman | 2011-05-02 14:04:36 -0500 (Mon, 02 May 2011)
+ | 8 lines More possible crashes based upon invalid inputs.
+ (closes issue #18161) Reported by: wdoekes Patches:
+ 20110301__issue18161.diff.txt uploaded by tilghman (license 14)
+ Tested by: wdoekes ........ ................
+
+2011-05-02 15:58 +0000 [r316054] Paul Belanger <pabelanger@digium.com>
+
+ * apps/app_meetme.c: Formatting change, remove red blobs
+
+2011-04-27 19:15 +0000 [r315895] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Merged
+ revisions 315894 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r315894 | mnicholson | 2011-04-27 14:14:27 -0500
+ (Wed, 27 Apr 2011) | 28 lines Merged revisions 315893 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r315893 | mnicholson | 2011-04-27 14:03:05 -0500
+ (Wed, 27 Apr 2011) | 21 lines Merged revisions 315891 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr
+ 2011) | 14 lines Fix our compliance with RFC 3261 section 18.2.2.
+ This change optimizes the free_via() function and removes some
+ redundant null checking. It also fixes compliance with RFC 3261
+ section 18.2.2 by always using the port specified in the Via
+ header for routing responses (even when maddr is not set). Also
+ the htons() function is now used when setting the port.
+ Additional documentation comments have been added in various
+ places to make the logic in the code clearer. (closes issue
+ #18951) Reported by: jmls Patches:
+ issue18951_set_proper_port_from_via.patch uploaded by wdoekes
+ (license 717) (modified) ........ ................
+ ................
+
+2011-04-27 17:51 +0000 [r315855-315856] David Vossel <dvossel@digium.com>
+
+ * apps/app_confbridge.c: Makes the new ConfBridge join and leave
+ sounds be used by default rather than beep and beeperr.
+
+ * main/channel.c: Clears exception flag during ast_read when
+ func_jitterbuffer is enabled
+
+2011-04-27 15:56 +0000 [r315811] Russell Bryant <russell@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 315810 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r315810 | russell | 2011-04-27 10:55:48 -0500 (Wed, 27 Apr 2011)
+ | 2 lines Set the copyright year to 2011 in the startup message.
+ ........
+
+2011-04-27 12:37 +0000 [r315766] Leif Madsen <lmadsen@digium.com>
+
+ * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 315765
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r315765 | lmadsen | 2011-04-27 07:36:17 -0500 (Wed, 27 Apr 2011)
+ | 4 lines Enable Russian core sound selection in menuselect.
+ (closes issue #18724) Reported by: pbxware ........
+
+2011-04-26 23:10 +0000 [r315670-315675] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 315673 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r315673 | twilson | 2011-04-26 15:56:19 -0700
+ (Tue, 26 Apr 2011) | 25 lines Merged revisions 315672 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r315672 | twilson | 2011-04-26 15:52:25 -0700
+ (Tue, 26 Apr 2011) | 18 lines Merged revisions 315671 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011)
+ | 11 lines Make sure unregistering a peer unlinks it from the
+ peer container Instead of mostly copying the code from
+ expire_register, just use the function that "does the right
+ thing". (closes issue #16033) Reported by: kkm Patches:
+ 016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888)
+ Tested by: kkm, tilghman, twilson ........ ................
+ ................
+
+ * channels/chan_sip.c: Make sure to create the caps structure for
+ autocreated peers Because crashing is bad.
+
+ * apps/app_dial.c, main/features.c, apps/app_queue.c: Merged
+ revisions 315644 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r315644 | twilson | 2011-04-26 14:39:01 -0700
+ (Tue, 26 Apr 2011) | 32 lines Merged revisions 315643 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r315643 | twilson | 2011-04-26 14:27:44 -0700
+ (Tue, 26 Apr 2011) | 25 lines Merged revisions 315596 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011)
+ | 18 lines Allow transfer loops without allowing forwarding loops
+ We try to avoid the situation where two phones may be forwarded
+ to each other causing an infinite loop by storing each dialed
+ interface in a channel datastore and checking the list before
+ dialing out. This works, but currently breaks situations like A
+ calls B, A transfers B to C, B transfers C to A, and A transfers
+ C to B. Since human interaction is happening here and not an
+ automated forwarding loop, it should be allowed. This patch
+ removes the dialed_interfaces datastore when a call is bridged (a
+ suggestion from the brilliant mmichelson). If a call is being
+ bridged, it should be safe to assume that we aren't stuck in a
+ loop. Since we are now handling this is the bridge code, the
+ previous attempts at handling it in app_dial and app_queue are
+ removed. Review: https://reviewboard.asterisk.org/r/1195/
+ ........ ................ ................
+
+2011-04-26 22:18 +0000 [r315649] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /: Merged revisions 315645 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r315645 | rmudgett | 2011-04-26 17:14:31 -0500 (Tue, 26 Apr 2011)
+ | 21 lines The 'e' special extension fails to trigger in at least
+ two cases. The 'e' extension is a fall back for the 'i', 't', or
+ 'T' extensions if any of them do not exist. Many of the places
+ the 'e' extension was supposed to be invoked fail because the
+ priority was set wrong. There were two places where the 'e'
+ extension was not even checked for fall back. * Made invoke the
+ 'e' extension similarly to the previous 'i', 't', or 'T'
+ extension check and added the 'e' extension as a fall back to the
+ two missing locations. * Prioritized and optimized some hangup
+ tests associated with the 'e' extension. (closes issue #19136)
+ Reported by: kshumard Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1196/ ........
+
+2011-04-26 19:38 +0000 [r315504] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * include/asterisk/select.h, /: Merged revisions 315503 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r315503 | tilghman | 2011-04-26 14:32:50 -0500
+ (Tue, 26 Apr 2011) | 28 lines Merged revisions 315502 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r315502 | tilghman | 2011-04-26 14:22:52 -0500
+ (Tue, 26 Apr 2011) | 21 lines Merged revisions 315501 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011)
+ | 14 lines Fix the bounds-checking code. The code that set the
+ bit within the select bitfield was correct, but the
+ bounds-checking code was not. The change to that line uses the
+ new _bitsize macro for clarity. Also, FD_ZERO macro did not
+ zero-out anything but the first word of the bitfield, so this
+ could have caused problems with modules using that macro with the
+ expanded bitfield. (closes issue #18773) Reported by: jamicque
+ Patches: 20110423__issue18773.diff.txt uploaded by tilghman
+ (license 14) Tested by: chris-mac ........ ................
+ ................
+
+2011-04-26 18:02 +0000 [r315453] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 315452 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r315452 | rmudgett | 2011-04-26 13:00:34 -0500 (Tue, 26 Apr 2011)
+ | 1 line Add missing set of name valid flag when dialing.
+ ........
+
+2011-04-26 17:41 +0000 [r315447] Russell Bryant <russell@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 315446 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r315446 | russell | 2011-04-26 12:40:23 -0500 (Tue, 26
+ Apr 2011) | 14 lines chan_local: resolve a deadlock. This patch
+ resolves a fairly complex deadlock that can occur with the
+ combination of chan_local and a dialplan switch, such as dynamic
+ realtime extensions, which pulls autoservice into the picture
+ when doing a dialplan lookup. (closes issue #18818) Reported by:
+ nic Patches: issue18818.patch uploaded by jthurman (license 614)
+ 18818.v1.txt uploaded by russell (license 2) Tested by: nic,
+ jthurman, kterzi, steve-howes, sysreq, IshMalik ........
+
+2011-04-26 02:21 +0000 [r315395] Paul Belanger <pabelanger@digium.com>
+
+ * /, pbx/pbx_config.c: Merged revisions 315394 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r315394 | pabelanger | 2011-04-25 22:18:50 -0400
+ (Mon, 25 Apr 2011) | 14 lines Merged revisions 315393 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r315393 | pabelanger | 2011-04-25 22:17:43 -0400 (Mon, 25 Apr
+ 2011) | 7 lines Add back CLI command 'dialplan save' (closes
+ issue #19140) Reported by: lmadsen Patches:
+ __20110419_dialplan_save.patch.txt uploaded by lmadsen (license
+ 10) ........ ................
+
+2011-04-25 21:55 +0000 [r315350] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_mgcp.c: Merged revisions 315349 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r315349 | rmudgett | 2011-04-25 16:49:00 -0500 (Mon, 25
+ Apr 2011) | 9 lines When using MGCP realtime gateway definitions,
+ random crashes occur. Fixed incorrect linked list node removal
+ for realtime gateways. (closes issue #18291) Reported by:
+ nahuelgreco Patches: dangling-pointers-when-pruning.patch
+ uploaded by nahuelgreco (license 162) ........
+
+2011-04-25 19:40 +0000 [r315214-315260] Russell Bryant <russell@digium.com>
+
+ * /, formats/format_wav.c: Merged revisions 315259 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r315259 | russell | 2011-04-25 14:37:32 -0500
+ (Mon, 25 Apr 2011) | 24 lines Merged revisions 315258 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r315258 | russell | 2011-04-25 14:31:44 -0500
+ (Mon, 25 Apr 2011) | 17 lines Merged revisions 315257 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25 Apr 2011)
+ | 10 lines Be more flexible with unknown chunks in wav files.
+ This patch makes format_wav ignore unknown chunks instead of
+ erroring out on them. (closes issue #18306) Reported by: jhirsch
+ Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch
+ (license 1156) ........ ................ ................
+
+ * /, channels/chan_sip.c: Merged revisions 315213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r315213 | russell | 2011-04-25 14:04:28 -0500
+ (Mon, 25 Apr 2011) | 14 lines Merged revisions 315212 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r315212 | russell | 2011-04-25 14:00:24 -0500 (Mon, 25 Apr 2011)
+ | 7 lines Don't link non-cached realtime peers into the
+ peers_by_ip container. (closes issue #18924) Reported by: wdoekes
+ Patches: issue18924_uncached_realtime_peers_leak-1.6.2.17.patch
+ uploaded by wdoekes (license 717) ........ ................
+
+2011-04-25 07:17 +0000 [r315054] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_local.c, /: Merged revisions 315053 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r315053 | alecdavis | 2011-04-25 19:14:32 +1200
+ (Mon, 25 Apr 2011) | 23 lines Merged revisions 315052 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r315052 | alecdavis | 2011-04-25 19:11:12 +1200
+ (Mon, 25 Apr 2011) | 16 lines Merged revisions 315051 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25 Apr
+ 2011) | 11 lines chan_local:check_bridge() misplaced misplaced
+ ast_mutex_unlock if !p->chan->_bridge->_softhangup path isn't
+ followed, brigde remains locked. (closes issue #19176) Reported
+ by: alecdavis Patches: bug19176.diff.txt uploaded by alecdavis
+ (license 585) ........ ................ ................
+
+2011-04-22 23:01 +0000 [r315002] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_dahdi.c, /: Merged revisions 315001 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r315001 | alecdavis | 2011-04-23 10:59:18 +1200 (Sat, 23
+ Apr 2011) | 12 lines chan_dahdi: Can't return to normal ring
+ after distinctive ring on FXS clear a previous distinctivering
+ pattern before each new call (closes issue #18985) Reported by:
+ bromont Patches: bug18985.diff.txt uploaded by alecdavis (license
+ 585) Tested by: alecdavis, bromont ........
+
+2011-04-22 21:33 +0000 [r314960] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 314959 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r314959 | mnicholson | 2011-04-22 16:20:08 -0500
+ (Fri, 22 Apr 2011) | 24 lines Merged revisions 314958 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r314958 | mnicholson | 2011-04-22 15:49:45 -0500
+ (Fri, 22 Apr 2011) | 17 lines Merged revisions 311203,314908 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar
+ 2011) | 4 lines Don't hold the pvt lock while streaming a file.
+ ABE-2756 ........ r314908 | mnicholson | 2011-04-22 15:01:48
+ -0500 (Fri, 22 Apr 2011) | 4 lines Prevent the login thread and
+ the app threads from using the asterisk channel at the same time.
+ ABE-2756 ........ ................ ................
+
+2011-04-22 14:49 +0000 [r314824] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_unistim.c, /, res/res_fax_spandsp.c: Merged
+ revisions 314779 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r314779 | tzafrir | 2011-04-22 16:59:43 +0300 (ו', 22 אפר 2011) |
+ 2 lines Fix a few typos (shown by Lintian) ........
+
+2011-04-22 14:08 +0000 [r314781] Russell Bryant <russell@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 314780 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r314780 | russell | 2011-04-22 09:02:23 -0500
+ (Fri, 22 Apr 2011) | 18 lines Merged revisions 314778 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r314778 | russell | 2011-04-22 08:58:03 -0500 (Fri, 22 Apr 2011)
+ | 11 lines Initialize buffers in getvar and getvarfull.
+ Initialize the buffers used to hold the result from GET VARIABLE
+ or GET VARIABLE FULL. The bug report shows func_read returning
+ garbage in the result. It assumed that the buffer passed in was
+ initialized, like many other functions do. In the more common
+ code path (through the dialplan), it is initialized, so just
+ initialize it here too. (closes issue #19050) Reported by: johnz
+ ........ ................
+
+2011-04-21 22:53 +0000 [r314733-314735] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Implement AMI action PRIShowSpans. PRIShowSpans works like the
+ AMI action DAHDIShowChannels but for PRI spans. It is similar to
+ the CLI command "pri show spans". (closes issue #15980) Reported
+ by: dwery
+
+ * channels/sig_pri.c: Simplify sig_pri.c:build_status().
+
+ * channels/chan_dahdi.c, /: Merged revisions 314732 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r314732 | rmudgett | 2011-04-21 17:38:44 -0500 (Thu, 21
+ Apr 2011) | 1 line Correct DAHDIShowChannels XML documentation.
+ ........
+
+2011-04-21 18:32 +0000 [r314666] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c,
+ main/http.c, configs/sip.conf.sample, configs/skinny.conf.sample,
+ channels/sip/include/sip.h, configs/http.conf.sample: Merged
+ revisions 314628 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r314628 | mnicholson | 2011-04-21 13:24:05 -0500
+ (Thu, 21 Apr 2011) | 27 lines Merged revisions 314620 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r314620 | mnicholson | 2011-04-21 13:22:19 -0500
+ (Thu, 21 Apr 2011) | 20 lines Merged revisions 314607 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr
+ 2011) | 14 lines Added limits to the number of unauthenticated
+ sessions TCP based protocols are allowed to have open
+ simultaneously. Also added timeouts for unauthenticated sessions
+ where it made sense to do so. Unrelated, the manager interface
+ now properly checks if the user has the "system" privilege before
+ executing shell commands via the Originate action. AST-2011-005
+ AST-2011-006 (closes issue #18787) Reported by: kobaz (related to
+ issue #18996) Reported by: tzafrir ........ ................
+ ................
+
+2011-04-21 18:11 +0000 [r314598] David Vossel <dvossel@digium.com>
+
+ * configs/confbridge.conf.sample (added), apps/confbridge (added),
+ bridges/bridge_softmix.c, UPGRADE.txt,
+ include/asterisk/channel.h, res/res_musiconhold.c, CHANGES,
+ apps/confbridge/conf_config_parser.c (added), main/channel.c,
+ include/asterisk/bridging_technology.h,
+ bridges/bridge_builtin_features.c,
+ apps/confbridge/include/confbridge.h (added), apps/Makefile,
+ include/asterisk/bridging_features.h,
+ include/asterisk/bridging.h, include/asterisk/dsp.h,
+ apps/app_confbridge.c, apps/confbridge/include (added),
+ main/bridging.c, main/dsp.c: New HD ConfBridge conferencing
+ application. Includes a new highly optimized and customizable
+ ConfBridge application capable of mixing audio at sample rates
+ ranging from 8khz-192khz. Review:
+ https://reviewboard.asterisk.org/r/1147/
+
+2011-04-21 00:29 +0000 [r314551] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 314550 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r314550 | twilson | 2011-04-20 17:23:04 -0700
+ (Wed, 20 Apr 2011) | 13 lines Merged revisions 314549 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011)
+ | 6 lines Don't allocate more space than necessary for a sip_pkt
+ This extra allocation is a hold-over from when pkt->data was a
+ character array. Now that it is an allocated string, just
+ allocate enough for the sip_pkt. ........ ................
+
+2011-04-20 20:52 +0000 [r314509] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, main/abstract_jb.c, funcs/func_jitterbuffer.c
+ (added), include/asterisk/channel.h, CHANGES,
+ include/asterisk/abstract_jb.h: Introduction of the JITTERBUFFER
+ dialplan function. Review:
+ https://reviewboard.asterisk.org/r/1157/
+
+2011-04-20 19:56 +0000 [r314471] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c: codec_dahdi: DAHDI still advertises formats
+ using the old bitfields. Previously, the DAHDI format bit fields
+ matched up with the Asterisk bitfields. Since the Asterisk codec
+ bit fields were replaced in r306010, codec_dahdi needs to contain
+ the formats itself. In the future, the DAHDI formats should
+ either change to something other than bitfields, or the bitfields
+ need to move from include/dahdi/kernel.h to include/dahdi/user.h.
+ Signed-off-by: Shaun Ruffell <sruffell@digium.com>
+
+2011-04-20 16:55 +0000 [r314418] Richard Mudgett <rmudgett@digium.com>
+
+ * /, include/asterisk/frame.h: Merged revisions 314417 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r314417 | rmudgett | 2011-04-20 11:54:02 -0500 (Wed, 20
+ Apr 2011) | 1 line AST_CONTROL_XXX comment changes. ........
+
+2011-04-20 16:37 +0000 [r314415] David Vossel <dvossel@digium.com>
+
+ * codecs/codec_resample.c: Fixes error with frame datalen being
+ calculated from samples when this is not allwaya accurate.
+
+2011-04-20 05:28 +0000 [r314359] Terry Wilson <twilson@digium.com>
+
+ * main/lock.c, /: Merged revisions 314358 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r314358 | twilson | 2011-04-19 22:25:15 -0700 (Tue, 19 Apr 2011)
+ | 4 lines Initialize track pointer ast_reentrancy_init checks to
+ see if it is NULL before initializing with calloc ........
+
+2011-04-19 15:42 +0000 [r314204-314252] Leif Madsen <lmadsen@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 314251 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r314251 | lmadsen | 2011-04-19 10:42:10 -0500 (Tue, 19 Apr 2011)
+ | 8 lines Use SSLv23_client_method instead of old SSLv2 only.
+ (closes issue #19095) (closes issue #19138) Reported by: tzafrir
+ Patches: no_ssl2.diff uploaded by tzafrir (license 46) Tested by:
+ russell, chazzam ........
+
+ * /, funcs/func_channel.c: Merged revisions 314206 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r314206 | lmadsen | 2011-04-19 09:28:15 -0500
+ (Tue, 19 Apr 2011) | 14 lines Merged revisions 314205 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r314205 | lmadsen | 2011-04-19 09:27:50 -0500 (Tue, 19 Apr 2011)
+ | 6 lines Remove duplicate documentation from func_channel.c
+ (closes issue #18970) Reported by: IgorG Patches:
+ func_channel.c.doc.diff uploaded by IgorG (license 20) ........
+ ................
+
+ * apps/app_dial.c, /: Merged revisions 314203 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r314203 | lmadsen | 2011-04-19 09:24:25 -0500
+ (Tue, 19 Apr 2011) | 15 lines Merged revisions 314202 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r314202 | lmadsen | 2011-04-19 09:23:39 -0500 (Tue, 19 Apr 2011)
+ | 7 lines Update seconds to milliseconds in ast_verb output.
+ (closes issue #19084) Reported by: smurfix Patches:
+ app_dial.patch uploaded by smurfix (license 547) Tested by:
+ lmadsen, smurfix ........ ................
+
+2011-04-19 08:22 +0000 [r314158] Olle Johansson <oej@edvina.net>
+
+ * apps/app_meetme.c: Add explanation of strange flag setup in
+ app_meetme (stolen from Mark's message to asterisk-dev)
+
+2011-04-18 19:48 +0000 [r314079-314116] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sig_pri.c: Problems with ISDN MWI to phones. The
+ "controlling user number" is always the number of the voice mail
+ box which is identical with the subscriber number itself. This
+ number which is listed in the ISDN phone MWI menu cannot be
+ called back to contact the voice mail box. The controlling user
+ number should be made configurable. JIRA ABE-2738 JIRA SWP-2846
+
+ * /, res/res_agi.c: Merged revisions 314069 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r314069 | rmudgett | 2011-04-18 11:10:10 -0500 (Mon, 18 Apr 2011)
+ | 22 lines The AsyncAGI command loop is lax in the value it
+ returns for the return status. * Return correct status:
+ SUCCESS/FAILED/HANGUP. Previously, abnormal exits from the
+ command loop such as hangup would return SUCCESS. * The "asyncagi
+ break" command now returns SUCCESS and is now the only way to
+ break the command loop with that status. Previously, it returned
+ FAILED. * The AMI event AsyncAGI End is no longer sent if the
+ AsyncAGI Start event is not sent. Previously, this happened
+ because of an error setting up the AGI pipes. * All executed AGI
+ commands now get an AsyncAGI Exec result event. Previously, if
+ the command returned failure (because of hangup), the command
+ loop just exited with FAILURE and did not send the AsyncAGI Exec
+ result event. * Makes sure that the channel frame queue is empty
+ on hangup. Review: https://reviewboard.asterisk.org/r/1183/
+ ........
+
+ * apps/app_dial.c, /: Merged revisions 314068 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r314068 | rmudgett | 2011-04-18 11:02:12 -0500 (Mon, 18 Apr 2011)
+ | 7 lines Unclear code in app_dial.c. Make code formatting clear.
+ (closes issue #19134) Reported by: oej ........
+
+2011-04-18 16:22 +0000 [r314018-314078] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 314067 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r314067 | dvossel | 2011-04-18 10:23:45 -0500 (Mon, 18 Apr 2011)
+ | 22 lines Remove the need for deadlock avoidance in chan_sip
+ do_monitor. Deadlock avoidance between the sip pvt and the
+ pvt->owner is very difficult. Now that channel's are ao2 objects,
+ this complication is no longer necessary. It turns out the pvt's
+ msg queue only exists because of deadlock avoidance (when
+ deadlock avoidance fails msgs were added to a queue to be
+ processed later), so this goes away as well. The technique used
+ in the new sip_lock_pvt_full() function should be used as a
+ template for replacing all locations where deadlock avoidance
+ occurs between a channel tech_pvt and the pvt's owner. My hope is
+ that this will begin a reversal of the invalid channel driver
+ locking architecture we have been using for so long. This patch
+ also resolves an issue where the pvt->owner gets unlocked during
+ processing the msg queue. (closes issue #18690) Reported by:
+ dvossel Review: https://reviewboard.asterisk.org/r/1182/ ........
+
+ * main/rtp_engine.c, /, channels/chan_sip.c,
+ include/asterisk/rtp_engine.h: Merged revisions 314017 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011)
+ | 17 lines sip codec negotiation of dynamic rtp payloads error
+ fix This patch fixes how chan_sip handles dynamic rtp payload
+ types it does not understand. At the moment if a dynamic
+ payload's mime type does not match one we understand, the payload
+ does not get removed from our payload table. As a result of this,
+ the payload is set to whatever dynamic codec we use internally
+ for that payload number on outgoing INVITES. This is incorrect.
+ This patch fixes this by properly checking the rtpmap set
+ function's return code to make sure it was found. The function
+ can return both -1 and -2 depending on the source of the
+ mismatch. We were just checking -1 explicitly. Review:
+ https://reviewboard.asterisk.org/r/1169/ ........
+
+2011-04-17 09:28 +0000 [r313980] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Consolidate all new call calls to run
+ through new setsubstate_ringout. (closes issue #17907) Reported
+ by: wedhorn Patches: cleanup.stateringout.diff uploaded by
+ wedhorn (license 30) Tested by: salecha, wedhorn
+
+2011-04-17 01:28 +0000 [r313907-313944] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: fix compile error from r313907
+
+ * addons/chan_ooh323.c: fix trivial error with set_max_datagram on
+ pvt->udptl
+
+2011-04-15 15:20 +0000 [r313867] Jonathan Rose <jrose@digium.com>
+
+ * /, main/cli.c: Merged revisions 313860 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r313860 | jrose | 2011-04-15 10:08:05 -0500
+ (Fri, 15 Apr 2011) | 17 lines Merged revisions 313859 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r313859 | jrose | 2011-04-15 09:58:37 -0500 (Fri, 15 Apr 2011) |
+ 10 lines Fix a Tab Completion bug that occurs due to multiple
+ matches on a substring. Makes word_match function in cli.c repeat
+ a search for a command string until a proper match is found or
+ the string is searched to the last point. (closes issue #17494)
+ Reported by: ffossard Review:
+ https://reviewboard.asterisk.org/r/1180/ ........
+ ................
+
+2011-04-14 21:53 +0000 [r313822] Terry Wilson <twilson@digium.com>
+
+ * res/res_rtp_asterisk.c: Sets video mark bit on format field
+ correctly This fixes a regression in the media architecture
+ change where video frames did not have their video mark set
+ correctly. dvossel wrote this. twilson kindly committed this,
+ mmichelson found the bug.
+
+2011-04-14 21:02 +0000 [r313606-313781] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 313780 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r313780 | rmudgett | 2011-04-14 15:59:56 -0500 (Thu, 14
+ Apr 2011) | 20 lines Leftover debug messages unconditionally sent
+ to the console. Executing Dial(DAHDI/1/18475551212,300,) with the
+ echotraining config option enabled outputs the following debug
+ messages unconditionally: Dialing T1847555121 on 1 Dialing www2w
+ on 1 * Made debug messages in my_dial_digits() normal debug
+ messages that do not get output unless enabled. * Reworded some
+ debug messages in my_dial_digits() to be clearer. * Replace
+ strncpy() with ast_copy_string() in my_dial_digits() which does
+ the same job better. (closes issue #18847) Reported by:
+ vmikhelson Tested by: rmudgett ........
+
+ * CREDITS, main/ccss.c, configs/ccss.conf.sample: Add Device State
+ Information CCSS for Generic Devices. Add Asterisk Device State
+ information and callbacks to the Call Completion Supplemental
+ Services for generic agents. There are currently not many devices
+ that have native support for CCSS. Even as the devices become
+ available there may be other reasons why one may choose to not
+ take advantage of the native abilities and stick with the generic
+ implementation. The generic implementation is quite capable and
+ could be greatly enhanced by adding device state capabilities. A
+ phone could then subscribe to the device state with a BLF key in
+ conjunction with Asterisk hints. The advantages of the device
+ state information would allow a single button to: request CCSS,
+ cancel a CCSS request, and display the current state of a CCSS
+ request. For example, you may have a single button that when not
+ lit, there is no active CCSS request. When you press that button,
+ the dialplan can query the DEVICE_STATE() associated with that
+ caller to determine whether they should be calling
+ CallCompletionRequest() or CallCompletionCancel(). If there is
+ currently a pending request, then the dialplan would cancel it.
+ This also has the advantage of showing the true state of a
+ request, which is an asynchronous call, even when
+ CallCompletionRequest() thinks it was successful. The actual
+ request could ultimately fail. Once lit, further feedback can be
+ provided to the caller about the current state of their request
+ since it will be updated by the CCSS State Machine as
+ appropriate. The DEVICE_STATE mapping is configurable since the
+ BLF being used on a given phone type may vary. The idea is to
+ allow some level of customization as to the phone's behavior. As
+ an example, you may want the BLF key to go solid once you have
+ requested a callback. You may then want the LED to blink
+ (typically ringing) when either the callback is in process, which
+ is a visual indication that the incoming call is the desired
+ callback. You may want it to blink when the callee is ready but
+ you are busy, giving you a visual indication that the target is
+ available as you may want to get off the line so that the
+ callback can be successful. Device state information is sent back
+ via the ast_devstate_prov_add() callback for any generic CCSS
+ device as it traverses through the state machine. You simply
+ provide a map between CC_STATE values and the corresponding
+ AST_DEVICE state values. You could then generate hints against
+ these states similar to what is possible today with Custom
+ Devstates or MeetMe states. For example, you may have an
+ extension 3000 that is currently associated with device SIP/3000.
+ You could then create a feature code for that extension that may
+ look something like: exten => *823000,hint,ccss:sip/3000 You
+ would then subscribe a BLF button to *823000 which would point to
+ the dialplan that handled CCSS requests/cancels using the
+ available DEVICE_STATE() information about ccss:sip/3000 to make
+ the decision about what to do. (closes issue #18788) Reported by:
+ p_lindheimer Patches: ccss.trunk.18788.patch uploaded by p
+ lindheimer (license 558) Modified with final reviewboard
+ comments. Tested by: p_lindheimer, loloski Review:
+ https://reviewboard.asterisk.org/r/1105/
+
+ * /, res/res_agi.c: Merged revisions 313700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r313700 | rmudgett | 2011-04-13 17:52:47 -0500 (Wed, 13 Apr 2011)
+ | 5 lines Revert flushing stale AsyncAGI commands from -r313615.
+ It looks like it was intentional to leave any commands or
+ in-flight commands in the queue in case Async AGI is run again on
+ the call. ........
+
+ * /, res/res_agi.c: Merged revisions 313658 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r313658 | rmudgett | 2011-04-13 12:47:43 -0500 (Wed, 13 Apr 2011)
+ | 2 lines Miscellaneous AGI diagnostic message cleanup and code
+ optimization. ........
+
+ * /, res/res_agi.c: Merged revisions 313615 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r313615 | rmudgett | 2011-04-13 12:18:49 -0500 (Wed, 13 Apr 2011)
+ | 5 lines * Add missing channel lock to handle_cli_agi_add_cmd().
+ * Flush any Async AGI commands left over from earlier Async AGI
+ control of the call. ........
+
+ * main/channel.c, /, res/res_agi.c: Merged revisions 313588 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500
+ (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500
+ (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011)
+ | 41 lines Asterisk does not hangup a channel after endpoint
+ hangs up. If the call that the dialplan started an AGI script for
+ is hungup while the AGI script is in the middle of a command then
+ the AGI script is not notified of the hangup. There are many AGI
+ Exec commands that this can happen with. The reported
+ applications have been: Background, Wait, Read, and Dial. Also
+ the AGI Get Data command. * Don't wait on the Asterisk channel
+ after it has hung up. The channel is likely to never need
+ servicing again. * Restored the AGI script's ability to return
+ the AGI_RESULT_HANGUP value in run_agi(). It previously only
+ could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the
+ DeadAGI and AGI applications were merged. (closes issue #17954)
+ Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by
+ rmudgett (license 664) issue17954_v1.6.2.patch uploaded by
+ rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett
+ (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue
+ #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761
+ (closes issue #18935) Reported by: nvitaly Tested by: astmiv,
+ rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby
+ Tested by: rmudgett JIRA SWP-2727 Review:
+ https://reviewboard.asterisk.org/r/1165/ ........
+ ................ ................
+
+2011-04-13 15:49 +0000 [r313528] Leif Madsen <lmadsen@digium.com>
+
+ * configs/iax.conf.sample, configs/users.conf.sample,
+ channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
+ channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
+ channels/chan_iax2.c, channels/sip/include/sip.h: Add
+ 'description' field for CLI and Manager output (closes issue
+ #19076) Reported by: lmadsen Patches:
+ __20110408-channel-description.txt uploaded by lmadsen (license
+ 10) Tested by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/1163/
+
+2011-04-13 15:23 +0000 [r313527] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_dumpchan.c: Merged revisions 313517 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r313517 | rmudgett | 2011-04-12 17:35:53 -0500 (Tue, 12 Apr 2011)
+ | 12 lines Bring the dumpchan application inline with "core show
+ channel". * Added fields that are in "core show channel" to
+ dumpchan output. * Fixed reuse of formatbuf before the previous
+ string stored there was used by snprintf. All output strings now
+ have their own buffer. * Adjusted the buffer sizes to not be so
+ abusive of the stack now that there are more buffers. Change
+ requested by oej. ........
+
+2011-04-12 21:59 +0000 [r313482] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c,
+ addons/ooh323c/src/ooLogChan.h, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h,
+ addons/ooh323c/src/ooGkClient.h, addons/ooh323c/src/ooports.c,
+ addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ooh323ep.h,
+ addons/ooh323c/src/ootypes.h, addons/ooh323c/src/ooLogChan.c,
+ addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooSocket.h,
+ addons/ooh323c/src/ooq931.c: IPv6 support for chan_ooh323 IPv6
+ support for ooh323, bindaddr, peers and users ip can be IPv4 or
+ IPv6 addr correction for multi-homed mode (0.0.0.0 or ::
+ bindaddr) can work in dual 6/4 mode with :: bindaddr gatekeeper
+ mode isn't supported in v6 mode while (issue #18278) Reported by:
+ may213 Patches: ipv6-ooh323.patch uploaded by may213 (license
+ 454) Review: https://reviewboard.asterisk.org/r/1004/
+
+2011-04-12 18:53 +0000 [r313437-313438] Jonathan Rose <jrose@digium.com>
+
+ * /: blocking fix from 313436 that was already made in this commit
+
+ * channels/chan_dahdi.c, /: Merged revisions 313435 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 also
+ went ahead and fixed the problem it introduces before committing.
+ ........ r313435 | jrose | 2011-04-12 13:44:44 -0500 (Tue, 12 Apr
+ 2011) | 1 line fixing stupid mistake with putting code before
+ variable declaration ........ Merged revisions 313433 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) |
+ 14 lines reload Chan_dahdi memory leak caused by variables
+ chan_dahdi reloading with variables set via setvar in
+ chan_dahdi.conf would stay in the dahdi_pvt structs for
+ individual channels (causing them to just continue adding the new
+ ones to the list) and also there was a memory leak causes by the
+ conf objects. This patch resolves both of these by using
+ ast_variables_destroy during the loading process. (closes issue
+ #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by
+ jrose (license 1225) Tested by: tilghman, jrose Review:
+ https://reviewboard.asterisk.org/r/1170/ ........ ........
+ ........
+
+2011-04-11 23:20 +0000 [r313367-313383] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 313368-313369 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r313368 | rmudgett | 2011-04-11 18:03:02 -0500 (Mon, 11
+ Apr 2011) | 2 lines Backport a restructuring change from trunk to
+ make the next change stand out. ........ r313369 | rmudgett |
+ 2011-04-11 18:08:02 -0500 (Mon, 11 Apr 2011) | 13 lines Frames
+ from the inbound channel should go to all outbound channels in
+ app_dial.c. In app_dial.c:wait_for_answer() frames from the
+ inbound channel should be sent to all outbound channels instead
+ of only if there is just one outbound channel. Control frames
+ like AST_CONTROL_CONNECTED_LINE need to be passed to all of the
+ the outbound channels. This can happen if a blond transfer is
+ done by a remote switch on the inbound channel. JIRA AST-443 JIRA
+ SWP-2730 ........
+
+ * /, main/cli.c: Merged revisions 313366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r313366 | rmudgett | 2011-04-11 17:27:25 -0500 (Mon, 11 Apr 2011)
+ | 2 lines Added "Connected Line ID" and "Connected Line ID Name"
+ to "core show channel" output. ........
+
+2011-04-11 19:39 +0000 [r313280] Leif Madsen <lmadsen@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 313279 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r313279 | lmadsen | 2011-04-11 14:36:40 -0500
+ (Mon, 11 Apr 2011) | 21 lines Merged revisions 313278 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r313278 | lmadsen | 2011-04-11 14:33:03 -0500
+ (Mon, 11 Apr 2011) | 14 lines Merged revisions 313277 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011)
+ | 6 lines Fix detection of OpenSSL 1.0 (closes issue #19093)
+ Reported by: tzafrir Patches: detect_openssl_10.diff uploaded by
+ tzafrir (license 46) ........ ................ ................
+
+2011-04-11 15:47 +0000 [r313191] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 313190 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r313190 | rmudgett | 2011-04-11 10:40:30 -0500
+ (Mon, 11 Apr 2011) | 39 lines Merged revisions 313189 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r313189 | rmudgett | 2011-04-11 10:32:53 -0500
+ (Mon, 11 Apr 2011) | 32 lines Merged revisions 313188 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011)
+ | 25 lines Stuck channel using FEATD_MF if caller hangs up at the
+ right time. The cause was actually a caller hanging up just at
+ the end of the Feature Group D DTMF tones that setup the call.
+ The reason for this is a "guard timer" that's implemented using
+ ast_safe_sleep(100). If the caller happens to hang up AFTER the
+ final tone of the DTMF string but BEFORE the end of that
+ ast_safe_sleep(), then ast_safe_sleep() will return non-zero.
+ This causes the code to bounce to the end of ss_thread(), but it
+ does NOT tear down the call properly. This should be a rare
+ occurrence because the caller has to hang up at EXACTLY the right
+ time. Nonetheless, it was happening quite regularly on the
+ reporter's system. It's not easily reproducible, unless you
+ purposely increase the guard-time to 2000 or more. Once you do
+ that, you can reproduce it every time by watching the DTMF debug
+ and hanging up just as it ends. Simply add an ast_hangup() before
+ goto quit. (closes issue #15671) Reported by: jcromes Patches:
+ issue15671.patch uploaded by pabelanger (license 224) Tested by:
+ jcromes ........ ................ ................
+
+2011-04-09 21:00 +0000 [r313143] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: Merged revisions 313142 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r313142 | may | 2011-04-10 00:56:17 +0400 (Sun, 10 Apr
+ 2011) | 3 lines fix trivial bug in ooh323_indicate on
+ AST_CONTROL_SRC... check p->rtp is not null ........
+
+2011-04-08 16:17 +0000 [r313100] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_ss7.h, channels/sig_pri.c, channels/sig_ss7.c: Add
+ private lock deadlock avoidance callback to PRI and SS7. Factor
+ out the equivalent function for analog.
+
+2011-04-07 13:42 +0000 [r313049] Jonathan Rose <jrose@digium.com>
+
+ * /, main/features.c: Merged revisions 313048 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r313048 | jrose | 2011-04-07 08:35:33 -0500
+ (Thu, 07 Apr 2011) | 16 lines Merged revisions 313047 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r313047 | jrose | 2011-04-07 08:23:01 -0500 (Thu, 07 Apr 2011) |
+ 9 lines Makes parking lots clear and rebuild properly when
+ features reload is invoked from CLI Before, default parkinglot in
+ context parkedcalls with ext 700 would always be present and when
+ reload was invoked, the previous parkinglots would not be
+ cleared. (closes issue #18801) Reported by: mickecarlsson Review:
+ https://reviewboard.asterisk.org/r/1161/ ........
+ ................
+
+2011-04-07 10:30 +0000 [r313003-313005] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/sig_pri.c: Merged revisions 313001 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r313001 | alecdavis | 2011-04-07 22:19:31 +1200 (Thu, 07 Apr
+ 2011) | 13 lines Fix ISDN calling subaddr User Specified Odd/Even
+ Flag Calculation of the Odd/Even flag was wrong. Implement
+ correct algo, and set odd/even=0 if data would be truncated. Only
+ allow automatic calculation of the O/E flag, don't let dialplan
+ influence. (closes issue #19062) Reported by: festr Patches:
+ bug19062.diff2.txt uploaded by alecdavis (license 585) Tested by:
+ festr, alecdavis, rmudgett ........
+
+ * apps/app_voicemail.c: app_voicemail: close_mailbox change
+ LOG_WARNING to LOG_NOTICE
+
+2011-04-05 18:47 +0000 [r312868-312950] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c:
+ Merged revisions 312949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r312949 | rmudgett | 2011-04-05 13:45:24 -0500 (Tue, 05 Apr 2011)
+ | 6 lines Crash if ISDN span layer 1 is down on initial load.
+ Regression from -r312575 B channel shifting during negotiation. *
+ Also combine updating the alarm flag with clearing the resetting
+ flag. ........
+
+ * /, channels/chan_sip.c: Merged revisions 312889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r312889 | rmudgett | 2011-04-05 11:19:35 -0500 (Tue, 05 Apr 2011)
+ | 5 lines Add 416 response to OPTIONS packet. RFC3261 Section
+ 11.2 says the response code to an OPTIONS packet needs to be the
+ same as if it were an INVITE. ........
+
+ * /, channels/chan_sip.c: Merged revisions 312866 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r312866 | rmudgett | 2011-04-05 10:38:14 -0500 (Tue, 05 Apr 2011)
+ | 15 lines Responding to OPTIONS packet with 404 because Asterisk
+ not looking for "s" extension. The get_destination() function was
+ not using the "s" extension when the request URI did not specify
+ an extension. This is a regression caused when the URI parsing
+ code was extracted into parse_uri(). Made get_destination()
+ substitute the "s" extension when the parsed URI results in an
+ empty string. (closes issue #18348) Reported by: shmaize Patches:
+ issue18348_v1.8.patch uploaded by rmudgett (license 664) Tested
+ by: shmaize ........
+
+2011-04-05 14:16 +0000 [r312767] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /, configs/manager.conf.sample: Merged revisions
+ 312766 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r312766 | mnicholson | 2011-04-05 09:14:50 -0500
+ (Tue, 05 Apr 2011) | 22 lines Merged revisions 312764 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r312764 | mnicholson | 2011-04-05 09:13:07 -0500
+ (Tue, 05 Apr 2011) | 15 lines Merged revisions 312761 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r312761 | mnicholson | 2011-04-05 09:10:34 -0500 (Tue, 05 Apr
+ 2011) | 8 lines Limit the number of unauthenticated manager
+ sessions and also limit the time they have to authenticate.
+ AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested
+ by: mnicholson ........ ................ ................
+
+2011-04-05 13:55 +0000 [r312756] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_meetme.c: Minor change to 'L' option for meetme to
+ include some verb statements for the option.
+
+2011-04-04 19:31 +0000 [r312716] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Remove the channel parameter from
+ sig_pri_handle_subcmds(). It was only used in a debug message and
+ may not be correct anyway.
+
+2011-04-04 17:37 +0000 [r312678-312680] Jonathan Rose <jrose@digium.com>
+
+ * pbx/pbx_config.c: In handle_cli_dialplan_add_extension, const
+ char pointer *into_context is used instead of a->argv[5] to
+ improve readability.
+
+ * CHANGES, pbx/pbx_config.c: Makes 'dialplan add extension' create
+ the specified context if it does not already exist. If the user
+ invokes 'dialplan add extension' into a non-existing context, the
+ context will be created and a message informing the user of the
+ context being created will be issued in cli. (closes issue
+ #17431) Reported by: leearcher Patches: context_auto_create.diff
+ uploaded by kobaz (license 834) Tested by: leearcher, kobaz,
+ jrose
+
+2011-04-04 16:17 +0000 [r312579] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c:
+ Merged revisions 312575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r312575 | rmudgett | 2011-04-04 11:10:50 -0500
+ (Mon, 04 Apr 2011) | 52 lines Merged revisions 312574 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r312574 | rmudgett | 2011-04-04 11:00:02 -0500
+ (Mon, 04 Apr 2011) | 45 lines Merged revisions 312573 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011)
+ | 38 lines Issues with ISDN calls changing B channels during call
+ negotiations. The handling of the PROCEEDING message was not
+ using the correct call structure if the B channel was changed.
+ (The same for PROGRESS.) The call was also not hungup if the new
+ B channel is not provisioned or is busy. * Made all call
+ connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS,
+ ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are
+ using the correct structure and B channel. If there is any
+ problem with the operations then the call is now hungup with an
+ appropriate cause code. * Made miscellaneous messages
+ (INFORMATION, FACILITY, NOTIFY) find the correct structure by
+ looking for the call and not using the channel ID. NOTIFY is an
+ exception with versions of libpri before v1.4.11 because a call
+ pointer is not available for Asterisk to use. * Made all hangup
+ messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct
+ structure by looking for the call and not using the channel ID.
+ (closes issue #18313) Reported by: destiny6628 Tested by:
+ rmudgett JIRA SWP-2620 (closes issue #18231) Reported by:
+ destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue
+ #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The
+ issues fixed here are most likely causing this JIRA issue.) JIRA
+ DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed)
+ ........ ................ ................
+
+2011-04-01 23:17 +0000 [r312462-312510] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 312509 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r312509 | rmudgett | 2011-04-01 18:15:42 -0500 (Fri, 01
+ Apr 2011) | 22 lines When a call going out an NT-PTMP port gets
+ rejected, Asterisk crashes. If a call is sent to an ISDN phone
+ that rejects the call with RELEASE_COMPLETE(cause: call
+ reject(21), or busy(17)) Asterisk crashes. I could not get my
+ setup to crash. However, I could see the possibility from a race
+ condition between queuing an AST_CONTROL_BUSY to the core and
+ then queueing an AST_CONTROL_HANGUP. If the AST_CONTROL_BUSY is
+ processed before the AST_CONTROL_HANGUP is queued, the
+ ast_channel could be destroyed out from under chan_misdn. Avoid
+ this particular crash scenario by not queueing the
+ AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued. (closes
+ issue #18408) Reported by: wimpy Patches: issue18408_v1.8.patch
+ uploaded by rmudgett (license 664) Tested by: rmudgett, wimpy
+ JIRA SWP-2679 ........
+
+ * /, main/ccss.c: Merged revisions 312461 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011)
+ | 25 lines CallCompletionRequest()/CallCompletionCancel() exit
+ non-zero if fail. The
+ CallCompletionRequest()/CallCompletionCancel() dialplan
+ applications exit nonzero on normal failure conditions. The
+ nonzero exit causes the dialplan to hangup immediately. The
+ dialplan author has no opportunity to report success/failure to
+ the user. * Made always return zero so the dialplan can continue.
+ * Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and
+ CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively.
+ Also documented the values set. * Reduced the warning about no
+ core instance in CallCompletionCancel() to a debug message. It is
+ a normal event and should not be output at the WARNING level.
+ (closes issue #18763) Reported by: p_lindheimer Patches:
+ ccss.patch uploaded by p lindheimer (license 558) Modified Tested
+ by: p_lindheimer, rmudgett JIRA SWP-3042 ........
+
+2011-04-01 17:28 +0000 [r312384-312423] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_dahdi.c: Fixing bad line break from 312384
+
+ * channels/chan_dahdi.c, include/asterisk/dsp.h, CHANGES,
+ main/dsp.c: New Feature for chan_dahdi. 4 length pattern
+ matching. In chan_dahdi.conf, the user can now use length 4
+ patterns in addition to the usual length 2 patterns. The s ntax
+ remains the same and the method used to track the pattern history
+ will only change when using the length 4 patterns. (closes issue
+ SWP-3250) Code: jrose rmudgett
+
+2011-04-01 10:59 +0000 [r312289] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * include/asterisk/select.h, /, addons/cdr_mysql.c,
+ main/asterisk.c: Merged revisions 312286,312288 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r312286 | tilghman | 2011-04-01 05:44:33 -0500
+ (Fri, 01 Apr 2011) | 2 lines Reload must react correctly against
+ a possibly changed table, so dropping the conditional reload
+ flag. ................ r312288 | tilghman | 2011-04-01 05:58:45
+ -0500 (Fri, 01 Apr 2011) | 21 lines Merged revisions 312287 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r312287 | tilghman | 2011-04-01 05:51:24 -0500
+ (Fri, 01 Apr 2011) | 14 lines Merged revisions 312285 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011)
+ | 7 lines Found some leaking file descriptors while looking at
+ ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej
+ Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman
+ (license 14) ........ ................ ................
+
+2011-04-01 09:08 +0000 [r312118-312212] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, apps/app_voicemail.c: Merged revisions 312211 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r312211 | alecdavis | 2011-04-01 22:03:11 +1300
+ (Fri, 01 Apr 2011) | 36 lines Merged revisions 312210 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r312210 | alecdavis | 2011-04-01 21:47:29 +1300
+ (Fri, 01 Apr 2011) | 29 lines Merged revisions 312174 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr
+ 2011) | 23 lines voicemail: get real last_message_index and
+ count_messages, ODBC resequence change last_message_index to read
+ the max msgnum stored in the database change count_messages to
+ actually count the number of messages. last_message_index change:
+ This fixed overwriting of the last message if msgnum=0 was
+ missing. Previously every incoming message would overwrite
+ msgnum=1. count_messages change: allows us to detect when
+ requencing is required in opneA_mailbox. resequence enabled for
+ ODBC storage: Assists with fixing up corrupt databases with gaps,
+ but only when a user actively opens there mailboxes. (closes
+ issue #18692,#18582,#19032) Reported by: elguero Patches: based
+ on odbc_resequence_mailbox2.1.diff uploaded by elguero (license
+ 37) Tested by: elguero, nivek, alecdavis Review:
+ https://reviewboard.asterisk.org/r/1153/ ........
+ ................ ................
+
+ * /, apps/app_voicemail.c: Merged revisions 312117 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r312117 | alecdavis | 2011-04-01 20:32:12 +1300
+ (Fri, 01 Apr 2011) | 29 lines Merged revisions 312103 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r312103 | alecdavis | 2011-04-01 20:25:54 +1300
+ (Fri, 01 Apr 2011) | 22 lines Merged revisions 312070 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr
+ 2011) | 16 lines app_voicemail: close_mailbox needs to respect
+ additional messages while mailbox is open. close_mailbox leave
+ gaps in message sequence if messages are deleted and new messages
+ arrive during this time, this is because the shuffle down to slot
+ 0, only shuffles the number of pre-existing messages when mailbox
+ is opened, ignoring new arrivals. Fix: in close_mailbox
+ re-evaluate number of messages before the shuffle, this then
+ includes new arrivals. Happens on filebased or ODBC storage.
+ (issues #19032,#18582,#18692,#18998) Reported by:
+ alecdavis,tootai,afosorio Review:
+ https://reviewboard.asterisk.org/r/1153/ ........
+ ................ ................
+
+2011-03-31 20:12 +0000 [r311981-312023] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 312022 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r312022 | rmudgett | 2011-03-31 15:11:40 -0500 (Thu, 31
+ Mar 2011) | 14 lines chan_misdn segfaults when DEBUG_THREADS is
+ enabled. The segfault happens because jb->mutexjb is
+ uninitialized from the ast_malloc(). The internals of
+ ast_mutex_init() were assuming a nonzero value meant mutex
+ tracking initialization had already happened. Recent changes to
+ mutex tracking code to reduce excessive memory consumption
+ exposed this uninitialized value. Converted misdn_jb_init() to
+ use ast_calloc() instead of ast_malloc(). Also eliminated
+ redundant zero initialization code in the routine. (closes issue
+ #18975) Reported by: irroot ........
+
+ * include/asterisk/channel.h: Fix function reference in comment.
+
+2011-03-31 06:44 +0000 [r311931] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, configs/cdr_mysql.conf.sample: Merged revisions 311930 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r311930 | tilghman | 2011-03-31 01:43:18 -0500 (Thu, 31 Mar 2011)
+ | 6 lines Incorrect default example; the field is actually
+ internally named "clid", not "callerid". (closes issue #19040)
+ Reported by: wcselby Tested by: tilghman ........
+
+2011-03-30 01:57 +0000 [r311875] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 311874 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r311874 | rmudgett | 2011-03-29 20:56:05 -0500 (Tue, 29
+ Mar 2011) | 1 line Update some setup_dahdi_int() comments.
+ ........
+
+2011-03-29 08:33 +0000 [r311806] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * cel/cel_odbc.c, /: Merged revisions 311799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r311799 | tilghman | 2011-03-29 02:08:39 -0500 (Tue, 29 Mar 2011)
+ | 7 lines Remove extraneous check from integer-type fields.
+ (closes issue #19027) Reported by: mlehner Review:
+ https://reviewboard.asterisk.org/r/1149/ ........
+
+2011-03-28 22:00 +0000 [r311752] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 311751 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r311751 | russell | 2011-03-28 17:00:01 -0500 (Mon, 28
+ Mar 2011) | 2 lines Cross-reference VoiceMail() and
+ VoiceMailMain() in the xml docs. ........
+
+2011-03-27 21:49 +0000 [r311688] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: Merged revisions 311687 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r311687 | may | 2011-03-28 01:47:13 +0400 (Mon, 28 Mar
+ 2011) | 2 lines correct return values in ooh323_indicate for
+ AST_CONTROL_T38_PARAMETERS ........
+
+2011-03-23 21:55 +0000 [r311613-311616] Brett Bryant <bbryant@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 311615 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r311615 | bbryant | 2011-03-23 17:54:11 -0400 (Wed, 23 Mar 2011)
+ | 8 lines This patch fixes a bug with MeetMe behavior where the
+ 'P' option for always prompting for a pin is ignored for the
+ first caller. (closes issue #18070) Reported by: mav3rick Review:
+ https://reviewboard.asterisk.org/r/1132/ ........
+
+ * /, channels/sip/reqresp_parser.c: Merged revisions 311612 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r311612 | bbryant | 2011-03-23 17:45:46 -0400 (Wed, 23 Mar 2011)
+ | 9 lines Fix a possible crash in sip/reqresp_parser.c that is
+ caused by a possible null value. (closes issue #18821) Reported
+ by: cmaj Patches:
+ patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx
+ uploaded by cmaj (license 830) ........
+
+2011-03-23 02:51 +0000 [r311559] Terry Wilson <twilson@digium.com>
+
+ * /, channels/sip/reqresp_parser.c: Merged revisions 311558 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r311558 | twilson | 2011-03-22 19:24:53 -0700 (Tue, 22 Mar 2011)
+ | 5 lines Don't use static declared buf in parse_name_andor_addr
+ This function isn't used anywhere yet, but we definitely don't
+ want to keep the same value for buf between calls to the
+ function. ........
+
+2011-03-22 15:26 +0000 [r311498] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 311497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r311497 | dvossel | 2011-03-22 10:25:24 -0500
+ (Tue, 22 Mar 2011) | 9 lines Merged revisions 311496 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22
+ Mar 2011) | 2 lines Fixes memory leak in MeetMe AMI action
+ ........ ................
+
+2011-03-18 19:05 +0000 [r311427] Jonathan Rose <jrose@digium.com>
+
+ * CHANGES, apps/app_followme.c: Adds an option to FollowMe that
+ isn't useful for the bug it was made to solve. Still, due to the
+ nature of FollowMe, it makes sense to have this option since it
+ keeps apps bound to channels that would otherwise go away from
+ being lost.
+
+2011-03-18 16:27 +0000 [r311385] David Vossel <dvossel@digium.com>
+
+ * codecs/codec_resample.c: Remove libresample dependency from
+ codec_resample.c
+
+2011-03-18 16:24 +0000 [r311373] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c, res/res_fax.c, res/res_jabber.c: Merged
+ revisions 311352 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r311352 | jrose | 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) |
+ 10 lines Changes some print statements/events to use a blank
+ string in place of NULL if the string in question is NULL. This
+ is supposed to improve Solaris compatibility since Solaris goes
+ berserk when trying to output NULL strings. (closes issue #18759)
+ Reported by: bklang Patches: null-strings.patch uploaded by
+ bklang (license 919) ........
+
+2011-03-18 16:03 +0000 [r311343] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 311342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r311342 | mnicholson | 2011-03-18 11:02:50 -0500 (Fri, 18 Mar
+ 2011) | 2 lines Properly populate the LOCALSTATIONID channel
+ variable. ........
+
+2011-03-18 03:00 +0000 [r311296-311298] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 311297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r311297 | rmudgett | 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011)
+ | 12 lines Race condition when ISDN CallRerouting/CallDeflection
+ invoked. The queued AST_CONTROL_BUSY could sometimes be processed
+ before the call_forward dial string is recognized. * Moved
+ setting the call_forwarding dial string after sending a response
+ to the initiator and just queue an empty frame to wake up the
+ media thread instead of an AST_CONTROL_BUSY. * Added check for
+ empty rerouting/deflection number and respond with an error.
+ ........
+
+ * apps/app_dial.c, /: Merged revisions 311295 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r311295 | rmudgett | 2011-03-17 21:22:07 -0500
+ (Thu, 17 Mar 2011) | 35 lines Merged revision 310986 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed,
+ 16 Mar 2011) | 28 lines Dial() o option broke when connected line
+ feature added. The patch restores the o option behavior and adds
+ the ability to specify the CallerID. The Dial o and f options are
+ complementary to each other. The o option stores the CallerID on
+ the outgoing channel as the channel's CallerID. The f option
+ forces the CallerID sent by the outgoing channel. o(x) - The
+ argument 'x' is optional. If not present, then specify that the
+ CallerID that was present on the *calling* channel be stored as
+ the CallerID on the *called* channel. This was the behavior of
+ Asterisk 1.0 and earlier. If present, then specify the CallerID
+ stored on the *called* channel. Note that o(${CALLERID(all)}) is
+ similar to option o without parameters. f(x) - The argument 'x'
+ is optional and its presence changes the behavior of this option.
+ If not present, then force the outgoing CallerID on a
+ call-forward or deflection to the dialplan extension for this
+ Dial() using a dialplan 'hint'. For example, some PSTNs do not
+ allow CallerID to be set to anything other than the numbers
+ assigned to you. If present, then force the outgoing CallerID to
+ 'x'. Patches: jira_abe_2752_dial_fo_options.patch uploaded by
+ rmudgett (license 664) Tested by: rmudgett JIRA ABE-2752 JIRA
+ SWP-3096 .......... ................
+
+2011-03-17 19:05 +0000 [r311198] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 311197 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r311197 | jrose | 2011-03-17 14:03:34 -0500 (Thu, 17 Mar 2011) |
+ 11 lines This fixes a nasty chanspy bug which was causing a
+ channel leak every time a spied on channel made a call. In
+ addition to the above, it makes certain channel destruction
+ occurs so that applications don't get stuck waiting for datastore
+ destruction while monitored by chanspy. (closes issue #18742)
+ Reported by: jkister Tested by: jkister, jcovert, jrose Review:
+ http://reviewboard.digium.internal/r/106/ ........
+
+2011-03-17 15:02 +0000 [r311142] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /: Merged revisions 311141 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r311141 | mnicholson | 2011-03-17 10:00:33 -0500
+ (Thu, 17 Mar 2011) | 11 lines Merged revisions 311140 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r311140 | mnicholson | 2011-03-17 09:58:52 -0500 (Thu, 17 Mar
+ 2011) | 4 lines Don't write items to the manager socket twice.
+ AST-2011-003 (closes issue 0018987) Reported by: ks-steven
+ ........ ................
+
+2011-03-17 10:51 +0000 [r311051] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, configs/indications.conf.sample: Merged revisions 311050 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r311050 | alecdavis | 2011-03-17 23:49:41 +1300
+ (Thu, 17 Mar 2011) | 24 lines Merged revisions 311049 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r311049 | alecdavis | 2011-03-17 23:45:47 +1300
+ (Thu, 17 Mar 2011) | 17 lines Merged revisions 311048 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar
+ 2011) | 12 lines Remove extra quote in indications.conf Picking
+ low hanging fruit. (closes issue #18971) Reported by: IgorG
+ Patches: based on indications.conf.sample.diff uploaded by IgorG
+ (license 20) Tested by: IgorG ........ ................
+ ................
+
+2011-03-16 19:51 +0000 [r310941-311001] Terry Wilson <twilson@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 310999 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r310999 | twilson | 2011-03-16 14:47:59 -0500
+ (Wed, 16 Mar 2011) | 18 lines Merged revisions 310998 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r310998 | twilson | 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011)
+ | 11 lines Fix crash on fdopen failure See security advisory
+ AST-2011-004 (closes issue #18845) Reported by: cmaj Patches:
+ patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt
+ uploaded by cmaj (license 830)
+ patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt
+ uploaded by cmaj (license 830) Tested by: cmaj, twilson ........
+ ................
+
+ * main/manager.c, /: Merged revisions 310993 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r310993 | twilson | 2011-03-16 14:26:57 -0500
+ (Wed, 16 Mar 2011) | 11 lines Merged revisions 310992 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r310992 | twilson | 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011)
+ | 4 lines Don't keep trying to write to a closed connection See
+ security advisory AST-2011-003. ........ ................
+
+ * /, main/features.c: Merged revisions 310902 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r310902 | twilson | 2011-03-16 12:19:57 -0500
+ (Wed, 16 Mar 2011) | 43 lines Merged revisions 310889 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r310889 | twilson | 2011-03-16 12:03:27 -0500
+ (Wed, 16 Mar 2011) | 36 lines Merged revisions 310888 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011)
+ | 29 lines Don't delay DTMF in core bridge while listening for
+ DTMF features This patch is mostly the work of Olle Johansson. I
+ did some cleanup and added the silence generating code if
+ transmit_silence is set. When a channel listens for DTMF in the
+ core bridge, the outbound DTMF is not sent until we have received
+ DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds
+ of DTMF to Asterisk, which sends no audio for those 4 seconds.
+ Some products see this delay and the time skew on RTP packets
+ that results and start ignoring the audio that is sent afterward.
+ With this change, the DTMF_BEGIN frame is inspected and checked.
+ If it matches a feature code, we wait for DTMF_END and activate
+ the feature as before. If transmit_silence=yes in asterisk.conf,
+ silence is sent if we paritally match a multi-digit feature. If
+ it doesn't match a feature, the frame is forwarded along with the
+ DTMF_END without delay. By doing it this way, DTMF is not
+ delayed. (closes issue #15642) Reported by: jasonshugart Patches:
+ issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license
+ 396) Tested by: globalnetinc, jde (closes issue #16625) Reported
+ by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/
+ Review: https://reviewboard.asterisk.org/r/1125/ ........
+ ................ ................
+
+2011-03-15 01:49 +0000 [r310835] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * addons/chan_ooh323.c, /: Merged revisions 310834 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r310834 | tilghman | 2011-03-14 20:48:25 -0500 (Mon, 14
+ Mar 2011) | 2 lines Fix branch compile. ........
+
+2011-03-15 01:36 +0000 [r310833] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, main/utils.c: Merged revisions 310781 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r310781 | alecdavis | 2011-03-15 14:00:55 +1300 (Tue, 15 Mar
+ 2011) | 10 lines core show locks: display ThreadID in hexadecimal
+ Allow easier cross referencing of thread ID's with GDB backtraces
+ (closes issue #18968) Reported by: alecdavis Patches:
+ bug18968.diff.txt uploaded by alecdavis (license 585) ........
+
+2011-03-14 21:51 +0000 [r310735] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, addons/ooh323c/src/ooCapability.c, /,
+ addons/ooh323c/src/ooCalls.h: Merged revisions 310734 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 (closes
+ issue #18693) ........ r310734 | may | 2011-03-15 00:45:53 +0300
+ (Tue, 15 Mar 2011) | 12 lines Introduce t.38 parameters control
+ functionality not full but enough for Send/RcvFax support
+ Introduce t.38 controls between asterisk core and channel/proto
+ layers. Not all parameters are transferred from proto layers but
+ *Fax apps tested and work ok. (issue #18693) Reported by:
+ benngard2 Patches: issue-18693.patch uploaded by may213 (license
+ 454) ........
+
+2011-03-14 16:55 +0000 [r310637] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/callerid.c: Merged revisions 310636 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r310636 | rmudgett | 2011-03-14 11:50:59 -0500
+ (Mon, 14 Mar 2011) | 39 lines Merged revisions 310635 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r310635 | rmudgett | 2011-03-14 11:47:54 -0500
+ (Mon, 14 Mar 2011) | 32 lines Merged revisions 310633 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011)
+ | 25 lines "Caller*ID failed checksum" on Wildcard TDM2400P and
+ TDM410 The last character in the caller id message is getting a
+ framing error. The checksum is the last character in the message.
+ A framing error in the checksum could be because: 1) The sender
+ did not send a full stop bit. 2) The sender cut off the FSK
+ carrier too soon. 3) The sender opted to send zero of the
+ specified zero to 10 trailing mark bits and round-off errors in
+ the code resulted in the code not being where it thought it was
+ in the demodulated bit stream. Bit 8 of 'b' is set when parity
+ error. Bit 9 of 'b' is set when framing error. Made ignore the
+ framing and parity error bits if the errored character is the
+ checksum. We can tolerate a framing/parity error there. The
+ checksum character validates the message. (closes issue #18474)
+ Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek
+ (license 636) (with modifications) Tested by: nivek ........
+ ................ ................
+
+2011-03-14 15:40 +0000 [r310547-310588] Jonathan Rose <jrose@digium.com>
+
+ * /, funcs/func_volume.c: Merged revisions 310587 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r310587 | jrose | 2011-03-14 10:27:57 -0500
+ (Mon, 14 Mar 2011) | 15 lines Merged revisions 310585 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) |
+ 8 lines Adds 'p' as an option to func_volume. When it is on, the
+ old behavior with DTMF controlling volume adjustment will be
+ enforced. When it is off, DTMF will not be processed by the
+ function. Programmed by Jonathan Rose Reviewed by David Vossel,
+ Leif Madsen, and Russell Bryant
+ http://reviewboard.digium.internal/r/93/ ........
+ ................
+
+ * main/audiohook.c: Fixes null reference bug introduced by audio
+ hook changes that affects various OS distributions. Thanks David.
+
+2011-03-12 20:42 +0000 [r310416-310500] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, pbx/pbx_ael.c: Merged revisions 310462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r310462 | tilghman | 2011-03-12 14:27:54 -0600
+ (Sat, 12 Mar 2011) | 45 lines Merged revisions 310448 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r310448 | tilghman | 2011-03-12 14:24:54 -0600
+ (Sat, 12 Mar 2011) | 38 lines Recorded merge of revisions 310435
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12 Mar 2011)
+ | 31 lines Add AELSub, which provides a stable entry point into
+ AEL subroutines. This commit needs some explanation, given that
+ we're adding a new application into an existing release branch.
+ This is generally a violation of our release policy, except in
+ very limited circumstances, and I believe this is one of those
+ circumstances. The problem that this solves is one of the sanity
+ of using multiple dialplan languages to define a dialplan. In the
+ case of the reporter, he or she is using AEL is define
+ subroutines, while using Realtime extensions to invoke those
+ subroutines. While you can do this, it's based upon the reality
+ of AEL using actual dialplan extensions; however, there is no
+ guarantee that the details of _how_ AEL is compiled into
+ extensions will remain stable. In fact, at the time of this
+ commit, it has already changed twice, once in a fundamental way.
+ Now normally, a new application would only be added to trunk.
+ However, this application is explicitly to create a stable
+ user-level API between versions, and adding it to trunk only will
+ not solve the user's problem of switching between 1.6.2 and 1.8,
+ nor will it help anybody switching from 1.8 to 1.10. Therefore,
+ it needs to go into existing release branches. For the sake of
+ consistency, and also because one of the changes was between 1.4
+ and 1.6.x, I am also electing to commit this to 1.4. (closes
+ issue #18910) Reported by: alexandrekeller Patches:
+ 20110304__issue18919__1.6.2.diff.txt uploaded by tilghman
+ (license 14) 20110304__issue18919__1.4.diff.txt uploaded by
+ tilghman (license 14) Tested by: alexandrekeller ........
+ ................ ................
+
+ * /, funcs/func_odbc.c: Merged revisions 310415 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r310415 | tilghman | 2011-03-12 14:05:46 -0600
+ (Sat, 12 Mar 2011) | 14 lines Merged revisions 310414 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r310414 | tilghman | 2011-03-12 13:51:23 -0600 (Sat, 12 Mar 2011)
+ | 7 lines Transactional handles should be used for the insertbuf,
+ if available. Also, fix a possible resource leak. (closes issue
+ #18943) Reported by: irroot ........ ................
+
+2011-03-11 18:54 +0000 [r310373] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/audiohook.h, main/audiohook.c, CHANGES,
+ apps/app_mixmonitor.c: Mix Monitor: Now with r and t options.
+
+2011-03-11 15:09 +0000 [r310332] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, configure, codecs/gsm/Makefile, configure.ac,
+ makeopts.in, codecs/lpc10/Makefile: Use "-march=native" when
+ possible. Recent versions of GCC have a tuning option value of
+ 'native', which causes the compiler to optimize the build for the
+ CPU the compile is performed on. Since most people are building
+ Asterisk on the machine they plan to run it on, the configure
+ script and build system will now use this value unless a
+ different value is specified by the user in CFLAGS when the
+ configure script is executed. In addition, this value will be
+ used for building the GSM and LPC10 codecs as well, in preference
+ to the logic that has been in their Makefiles forever to optimize
+ for certain types of CPUs.
+
+2011-03-11 06:56 +0000 [r310288] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/rtp_engine.c, /: Merged revisions 310287 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r310287 | alecdavis | 2011-03-11 19:47:44 +1300 (Fri, 11 Mar
+ 2011) | 17 lines remote_bridge_loop: prevent segfault when after
+ transfer of IAX2 of DAHDI call If the channel condition is one of
+ the following after breaking out of the loop, don't try to
+ update_peer (where x = 0/1) 1). ZOMBIE 2). cx->tech_pvt != pvtx
+ 3). gluex != ast_rtp_instance_get_glue(cx->tech->type)) (closes
+ issue #18781) Reported by: alecdavis Patches: bug18781.diff3.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis, ZX81
+ Review: https://reviewboard.asterisk.org/r/1128/ ........
+
+2011-03-10 16:09 +0000 [r310241] Terry Wilson <twilson@digium.com>
+
+ * main/manager.c, /, res/res_phoneprov.c: Merged revisions 310240
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r310240 | twilson | 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011)
+ | 13 lines Add \r\n to remaining http headers passed to
+ ast_http_send r309204 changed the behavior of ast_http_send. It
+ now requires headers to be passed with trailing \r\n. This change
+ updates the remaining instances in the code that did not pass the
+ \r\n. (closes issue #18186) Reported by: nivaldomjunior Patches:
+ res_phoneprov.c.diff uploaded by lathama (license 1028)
+ manager.diff.txt uploaded by twilson (license 396) Tested by:
+ lathama ........
+
+2011-03-10 15:28 +0000 [r310238] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 310231 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r310231 | mmichelson | 2011-03-10 09:17:04 -0600 (Thu, 10 Mar
+ 2011) | 9 lines Be more tolerant of what URI we accept for call
+ completion PUBLISH requests. (closes issue #18946) Reported by:
+ GeorgeKonopacki Patches: 18946.patch uploaded by mmichelson
+ (license 60) Tested by: GeorgeKonopacki ........
+
+2011-03-10 05:54 +0000 [r310143] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * res/res_config_odbc.c, /, funcs/func_odbc.c,
+ apps/app_voicemail.c: Merged revisions 310142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r310142 | tilghman | 2011-03-09 23:53:29 -0600
+ (Wed, 09 Mar 2011) | 19 lines Merged revisions 310141 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r310141 | tilghman | 2011-03-09 23:51:37 -0600
+ (Wed, 09 Mar 2011) | 12 lines Merged revisions 310140 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011)
+ | 5 lines Initialize column size to 0 to deal with a potential
+ UnixODBC bug on 64-bit systems. (closes issue #18295) Reported
+ by: pruiz ........ ................ ................
+
+2011-03-08 20:34 +0000 [r310089] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/sip/dialplan_functions.c: Merged revisions 310088 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r310088 | jrose | 2011-03-08 14:19:32 -0600 (Tue, 08 Mar 2011) |
+ 9 lines Returns with an error notice if CHANNEL function of SIP
+ channel is read without arguments. (Closes issue #18653) Reported
+ by: wuwu Patches: diff.patch uploaded by jrose (license 1225)
+ Tested by: jrose ........
+
+2011-03-08 18:19 +0000 [r310045] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_calendar.c: Merged revisions 310039 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r310039 | twilson | 2011-03-08 10:10:50 -0800 (Tue, 08 Mar 2011)
+ | 11 lines Spelling fix in "calendar show calendar"
+ s/Cartegories/Catagories/ (closes issue #18931) Reported by:
+ pdugas Patches: res_calendar.c.patch uploaded by pdugas (license
+ 1222) Review: [full review board URL with trailing slash]
+ ........
+
+2011-03-08 16:46 +0000 [r309996] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 309994 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r309994 | rmudgett | 2011-03-08 10:37:02 -0600 (Tue, 08 Mar 2011)
+ | 1 line Make pri parameter description consistent. ........
+
+2011-03-07 22:16 +0000 [r309859] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_mixmonitor.c: Merged revisions 309858 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r309858 | jrose | 2011-03-07 16:07:25 -0600
+ (Mon, 07 Mar 2011) | 22 lines Merged revisions 309857 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r309857 | jrose | 2011-03-07 16:04:44 -0600
+ (Mon, 07 Mar 2011) | 15 lines Merged revisions 309856 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) |
+ 8 lines Bug fix for MixMonitor involving filenames with '.' not
+ in the extension Closes issue #18391) Reported by: pabelanger
+ Patches: bugfix.patch uploaded by jrose (license 1225) Tested by:
+ jrose ........ ................ ................
+
+2011-03-07 01:01 +0000 [r309809] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * channels/chan_dahdi.c, /, configure,
+ include/asterisk/autoconfig.h.in, main/ast_expr2f.c,
+ configure.ac, main/ast_expr2.fl: Merged revisions 309808 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r309808 | tilghman | 2011-03-06 18:54:42 -0600
+ (Sun, 06 Mar 2011) | 14 lines Merged revisions 309251 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011)
+ | 7 lines Revert previous 2 commits, and instead conditionally
+ redefine the same macro used in flex 2.5.35 that clashed with our
+ workaround. Not surprisingly, the workaround was exactly the same
+ code as was provided by the Flex maintainers, albeit in two
+ different places, in different macros. This should fix the
+ FreeBSD builds, which have an older version of Flex. ........
+ ................
+
+2011-03-07 00:14 +0000 [r309766] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/sip.conf.sample: Merged revisions 309765 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r309765 | mmichelson | 2011-03-06 18:13:36 -0600 (Sun,
+ 06 Mar 2011) | 3 lines Indicate that Asterisk uses the Allow
+ header to determine if MESSAGE requests should be sent. ........
+
+2011-03-05 17:53 +0000 [r309721] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 309720 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r309720 | moy | 2011-03-05 12:44:30 -0500 (Sat, 05 Mar
+ 2011) | 6 lines Fix caller id passed to openr2_chan_make_call
+ (closes issue #18894) Reported by: malufrj Tested by: moy
+ ........
+
+2011-03-05 10:30 +0000 [r309679] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/asterisk.c: Merged revisions 309678 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r309678 | tilghman | 2011-03-05 04:29:30 -0600
+ (Sat, 05 Mar 2011) | 14 lines Merged revisions 309677 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011)
+ | 7 lines Missed part of the conversion when we started passing
+ ppid to astcanary. (closes issue #18850) Reported by: viraptor
+ Patches: canary_ppid.patch uploaded by viraptor (license 543)
+ ........ ................
+
+2011-03-04 23:22 +0000 [r309640] Terry Wilson <twilson@digium.com>
+
+ * configs/calendar.conf.sample, include/asterisk/calendar.h,
+ CHANGES, res/res_calendar.c: Add setvar option to calendaring
+ Adding the setvar option with variable substitution on the value
+ allows things like setting the outbound caller id name to the
+ summary of a calendar event, etc. Values could be chained
+ together as they are appended in order to do some scripting if
+ necessary. Review: https://reviewboard.asterisk.org/r/1134/
+
+2011-03-04 19:38 +0000 [r309493-309587] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, pbx/pbx_lua.c: Merged revisions 309585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r309585 | mnicholson | 2011-03-04 13:38:25 -0600
+ (Fri, 04 Mar 2011) | 9 lines Merged revisions 309584 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r309584 | mnicholson | 2011-03-04 13:37:13 -0600 (Fri,
+ 04 Mar 2011) | 2 lines Restore mysterious lua_pushvalue() call
+ removed in r309494. The mystery has been solved. ........
+ ................
+
+ * /, pbx/pbx_lua.c: Merged revisions 309542 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r309542 | mnicholson | 2011-03-04 13:00:33 -0600
+ (Fri, 04 Mar 2011) | 11 lines Merged revisions 309541 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar
+ 2011) | 4 lines Check for errors from fseek() when loading config
+ file, properly abort on errors from fread(), and supply a
+ traceback for errors generated when loading the config file.
+ Also, prepend a newline to traceback output so that the main
+ error message is on it's own line. ........ ................
+
+ * /, pbx/pbx_lua.c: Merged revisions 309495 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r309495 | mnicholson | 2011-03-04 12:10:23 -0600
+ (Fri, 04 Mar 2011) | 9 lines Merged revisions 309494 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r309494 | mnicholson | 2011-03-04 11:55:57 -0600 (Fri,
+ 04 Mar 2011) | 2 lines remove mysterious lua_pushvalue() that is
+ never used ........ ................
+
+ * pbx/pbx_lua.c, configs/extensions.lua.sample: Add support for
+ defining hints from pbx_lua (closes issue #16024) Reported by:
+ mnicholson
+
+2011-03-04 17:40 +0000 [r309491] Russell Bryant <russell@digium.com>
+
+ * channels/chan_nbs.c: Fix a buglet that prevented chan_nbs from
+ loading (and subsequently stopped Asterisk). In passing, convert
+ the return codes to be the proper AST_MODULE_LOAD_* constants.
+
+2011-03-04 16:00 +0000 [r309449] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, pbx/pbx_lua.c: Merged revisions 309448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r309448 | mnicholson | 2011-03-04 09:59:25 -0600 (Fri, 04 Mar
+ 2011) | 8 lines Export global symbols from pbx_lua to allow
+ modules to be loaded. Fixes a regression introduced in r278132.
+ (closes issue #18671) Reported by: Igels Patches:
+ pbx_lua_global_symbols1.diff uploaded by mnicholson (license 96)
+ Tested by: Igels ........
+
+2011-03-04 15:28 +0000 [r309446] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, /,
+ funcs/func_channel.c, channels/sig_pri.c, UPGRADE-1.8.txt: Merged
+ revisions 309445 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011)
+ | 46 lines Get real channel of a DAHDI call. Starting with
+ Asterisk v1.8, the DAHDI channel name format was changed for ISDN
+ calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
+ There were several reasons that the channel name had to change.
+ 1) Call completion requires a device state for ISDN phones. The
+ generic device state uses the channel name. 2) Calls do not
+ necessarily have B channels. Calls placed on hold by an ISDN
+ phone do not have B channels. 3) The B channel a call initially
+ requests may not be the B channel the call ultimately uses.
+ Changes to the internal implementation of the Asterisk master
+ channel list caused deadlock problems for chan_dahdi if it needed
+ to change the channel name. Chan_dahdi no longer changes the
+ channel name. 4) DTMF attended transfers now work with ISDN
+ phones because the channel name is "dialable" like the chan_sip
+ channel names. For various reasons, some people need to know
+ which B channel a DAHDI call is using. * Added
+ CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and
+ CHANNEL(dahdi_type) so the dialplan can determine the B channel
+ currently in use by the channel. Use CHANNEL(no_media_path) to
+ determine if the channel even has a B channel. * Added AMI event
+ DAHDIChannel to associate a DAHDI channel with an Asterisk
+ channel so AMI applications can passively determine the B channel
+ currently in use. Calls with "no-media" as the DAHDIChannel do
+ not have an associated B channel. No-media calls are either on
+ hold or call-waiting. (closes issue #17683) Reported by: mrwho
+ Tested by: rmudgett (closes issue #18603) Reported by: arjankroon
+ Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett
+ (license 664) Tested by: stever28, rmudgett ........
+
+2011-03-04 01:52 +0000 [r309404] David Ruggles <thedavidfactor@gmail.com>
+
+ * /, apps/app_externalivr.c: Merged revisions 309403 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r309403 | diruggles | 2011-03-03 20:50:44 -0500
+ (Thu, 03 Mar 2011) | 23 lines Merged revisions 309356 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r309356 | diruggles | 2011-03-03 19:42:28 -0500
+ (Thu, 03 Mar 2011) | 16 lines Merged revisions 309355 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar
+ 2011) | 9 lines fix small memory leak fix small memory leak
+ caused by a string allocation that wasn't freed (closes issue
+ #18907) Reported by: andy11 Patches:
+ asterisk_trunk-app_externalivr-leak.patch uploaded by andy11
+ (license 1224) ........ ................ ................
+
+2011-03-02 21:08 +0000 [r309209-309300] Jason Parker <jparker@digium.com>
+
+ * main/channel.c: Add HangupRequest manager event, to specify
+ when/where a channel gets hung up. (closes issue #18226) Reported
+ by: clegall_proformatique Patches:
+ asterisk_1.8_293157_hanguprequests.svn.patch uploaded by clegall
+ proformatique (license 1139)
+
+ * /, channels/chan_sip.c: Merged revisions 309256 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r309256 | qwell | 2011-03-02 13:54:20 -0600
+ (Wed, 02 Mar 2011) | 15 lines Merged revisions 309255 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) |
+ 8 lines Fix usage of "hasvoicemail=yes" and "mailbox=" in
+ users.conf for SIP. Since it's a duplicate, nothing is going to
+ be done, so delme doesn't need to be set at all. Strangely, when
+ this was added, this was being set to 1 in 1.6, and 0 in trunk.
+ (issue AST-439) ........ ................
+
+ * /, main/http.c: Merged revisions 309204 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r309204 | qwell | 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) |
+ 7 lines Fix consistency of CRLFs on HTTP headers that get sent
+ out. (closes issue #18186) Reported by: nivaldomjunior Patches:
+ 18186-httpheadernewline.diff uploaded by qwell (license 4)
+ ........
+
+2011-03-01 21:57 +0000 [r309127-309171] Richard Mudgett <rmudgett@digium.com>
+
+ * /, funcs/func_channel.c: Merged revisions 309170 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r309170 | rmudgett | 2011-03-01 15:57:26 -0600 (Tue, 01
+ Mar 2011) | 7 lines Document CHANNEL(keypad_digits) and
+ CHANNEL(no_media_path). * Added XML documentation for
+ CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Tweaked XML
+ documentation for CHANNEL(reversecharge). ........
+
+ * channels/sig_analog.c, /: Merged revisions 309126 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r309126 | rmudgett | 2011-03-01 12:44:05 -0600 (Tue, 01
+ Mar 2011) | 16 lines Chan_dahdi does not retain CID when
+ detecting DTMF CID without polarity reversal. Looks like an
+ unintended change when sig_analog.c was extracted from
+ chan_dahdi.c. Removed useless conditional around needed code and
+ fixed resulting compiler warning. (closes issue #18667) Reported
+ by: enegaard Patches: issue18667.patch uploaded by enegaard
+ (license 1197) Tested by: enegaard JIRA SWP-2965 ........
+
+2011-03-01 16:22 +0000 [r309090] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 309084 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r309084 | dvossel | 2011-03-01 10:09:11 -0600
+ (Tue, 01 Mar 2011) | 15 lines Merged revisions 309083 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011)
+ | 9 lines Fixes thread blocking issue in the sip TCP/TLS
+ implementation. (closes issue #18497) Reported by: vois Patches:
+ issues_18497.diff uploaded by dvossel (license 671) Tested by:
+ vois, rossbeer, kowalma, Freddi_Fonet ........ ................
+
+2011-02-28 11:16 +0000 [r308992-309036] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ main/ast_expr2f.c, configure.ac, main/ast_expr2.fl: Merged
+ revisions 309035 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r309035 | tilghman | 2011-02-28 05:10:28 -0600
+ (Mon, 28 Feb 2011) | 15 lines Merged revisions 309033-309034 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011)
+ | 4 lines A later version of flex already includes the fwrite
+ workaround code, which if used twice causes a compilation error.
+ Detect whether Flex will compile without the workaround; if so,
+ suppress our workaround code. ........ r309034 | tilghman |
+ 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines Clarify
+ meaning, removing double negative (stupid!) ........
+ ................
+
+ * /, funcs/func_odbc.c: Merged revisions 308991 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r308991 | tilghman | 2011-02-28 03:33:22 -0600
+ (Mon, 28 Feb 2011) | 14 lines Merged revisions 308990 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011)
+ | 7 lines Statements updating zero rows may return SQL_NO_DATA.
+ This is fine; it's handled. (closes issue #18815) Reported by:
+ irroot Patches: func_odbc.insert_nodata.patch uploaded by irroot
+ (license 52) ........ ................
+
+2011-02-25 18:58 +0000 [r308946] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_sip.c: Merged revisions 308945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb
+ 2011) | 21 lines Fix Deadlock with attended transfer of SIP call
+ Call path sip_set_rtp_peer (locks chan then pvt)
+ transmit_reinvite_with_sdp try_suggested_sip_codec
+ pbx_builtin_getvar_helper (locks p->owner) But by the time
+ p->owner lock was attempted, seems as though chan and p->owner
+ were different. So in sip_set_rtp_peer, lock pvt first then lock
+ p->owner using deadlocking methods. (closes issue #18837)
+ Reported by: alecdavis Patches: bug18837-trunk.diff3.txt uploaded
+ by alecdavis (license 585) Tested by: alecdavis, Irontec, ZX81,
+ cmaj Review: [https://reviewboard.asterisk.org/r/1126/] ........
+
+2011-02-24 21:43 +0000 [r308904] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Merged revisions 308903 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011)
+ | 9 lines Invalid read in ast_channel_set_caller_event().
+ Valgrind reported that ast_channel_set_caller_event() was reading
+ data from a freed buffer when using the pre_set structure.
+ Rearange things to pre-calculate the name and number pointer
+ before updating the caller party structure to see if the name or
+ number was changed. ........
+
+2011-02-24 17:59 +0000 [r308816] Terry Wilson <twilson@digium.com>
+
+ * main/manager.c, /: Merged revisions 308815 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r308815 | twilson | 2011-02-24 11:57:18 -0600
+ (Thu, 24 Feb 2011) | 26 lines Merged revisions 308814 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r308814 | twilson | 2011-02-24 11:54:49 -0600
+ (Thu, 24 Feb 2011) | 19 lines Merged revisions 308813 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011)
+ | 12 lines Don't broadcast FullyBooted to every AMI connection
+ The FullyBooted event should not be sent to every AMI connection
+ every time someone connects via AMI. It should only be sent to
+ the user who just connected. (closes issue #18168) Reported by:
+ FeyFre Patches: bug0018168.patch uploaded by FeyFre (license
+ 1142) Tested by: FeyFre, twilson ........ ................
+ ................
+
+2011-02-24 15:10 +0000 [r308724] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/udptl.c, /: Merged revisions 308723 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r308723 | mnicholson | 2011-02-24 09:06:14 -0600
+ (Thu, 24 Feb 2011) | 16 lines Merged revisions 308722 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r308722 | mnicholson | 2011-02-24 08:59:41 -0600
+ (Thu, 24 Feb 2011) | 9 lines Merged revisions 308721 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu,
+ 24 Feb 2011) | 2 lines silence gcc 4.2 compiler warning ........
+ ................ ................
+
+2011-02-24 03:49 +0000 [r308680] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 308679 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r308679 | twilson | 2011-02-23 21:41:34 -0600
+ (Wed, 23 Feb 2011) | 15 lines Merged revisions 308678 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011)
+ | 8 lines Use remotesecret to authenticate with a remote party
+ The remotesecret option was only being used for outbound
+ registration and not for placing calls. This patch uses
+ remotesecret on outbound calls if it is set, otherwise secret is
+ still used. Review: https://reviewboard.asterisk.org/r/1107/
+ ........ ................
+
+2011-02-23 23:55 +0000 [r308623-308624] Richard Mudgett <rmudgett@digium.com>
+
+ * main/translate.c: Fix compiler warning.
+
+ * /, channels/sig_pri.c: Merged revisions 308622 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011)
+ | 9 lines sig_pri_new_ast_channel() should return NULL when
+ new_ast_channel() fails. (closes issue #18874) Reported by: cmaj
+ Patches:
+ patch-sig_pri-crash-possible-null-channel-pointer.diff.txt
+ uploaded by cmaj (license 830) JIRA SWP-3172 ........
+
+2011-02-22 23:04 +0000 [r308582] David Vossel <dvossel@digium.com>
+
+ * main/format.c, funcs/func_speex.c, main/frame.c,
+ main/rtp_engine.c, include/asterisk/silk.h (added),
+ codecs/speex/fixed_generic.h (added), bridges/bridge_softmix.c,
+ channels/chan_gtalk.c, bridges/bridge_multiplexed.c,
+ channels/chan_iax2.c, main/format_pref.c, codecs/speex/resample.c
+ (added), main/channel.c, funcs/func_pitchshift.c,
+ include/asterisk/audiohook.h, channels/chan_skinny.c,
+ main/format_cap.c, funcs/func_volume.c, codecs/speex (added),
+ codecs/codec_resample.c, include/asterisk/format.h,
+ codecs/speex/arch.h (added), include/asterisk/frame.h,
+ include/asterisk/rtp_engine.h, codecs/speex/stack_alloc.h
+ (added), main/bridging.c, apps/app_jack.c,
+ configs/codecs.conf.sample, res/res_rtp_asterisk.c,
+ formats/format_attr_silk.c (added), channels/chan_sip.c,
+ main/translate.c, main/slinfactory.c, codecs/codec_speex.c,
+ include/asterisk/_private.h, CHANGES,
+ codecs/speex/speex_resampler.h (added), res/res_mutestream.c,
+ include/asterisk/format_cap.h, codecs/Makefile,
+ channels/chan_jingle.c, main/data.c, channels/iax2.h,
+ main/audiohook.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
+ main/asterisk.c, include/asterisk/slinfactory.h,
+ include/asterisk/translate.h, codecs/speex/resample_sse.h
+ (added), include/asterisk/time.h: Media Project Phase2: SILK
+ 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio
+ ConfBridge, and other stuff -Functional changes 1. Dynamic global
+ format list build by codecs defined in codecs.conf 2. SILK 8khz,
+ 12khz, 16khz, and 24khz with custom attributes defined in
+ codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4.
+ SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz,
+ 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using
+ codec_resample.c 6. Various changes to RTP code required to
+ properly handle the dynamic format list and formats with
+ attributes. 7. ConfBridge now dynamically jumps to the best
+ possible sample rate. This allows for conferences to take
+ advantage of HD audio (Which sounds awesome) 8. Audiohooks are no
+ longer limited to 8khz audio, and most effects have been updated
+ to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
+ 9. codec_resample now uses its own code rather than depending on
+ libresample. -Organizational changes Global format list is moved
+ from frame.c to format.c Various format specific functions moved
+ from frame.c to format.c Review:
+ https://reviewboard.asterisk.org/r/1104/
+
+2011-02-22 15:33 +0000 [r308527] Andrew Latham <lathama@gmail.com>
+
+ * main/http.c: Use ast_debug for console logging Guessed the log
+ levels based on info that level 3 is the soft roof. Can we create
+ a page / document to define the levels?
+
+2011-02-21 15:04 +0000 [r308417] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/udptl.c, /: Merged revisions 308416 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r308416 | mnicholson | 2011-02-21 09:02:20 -0600
+ (Mon, 21 Feb 2011) | 19 lines Merged revisions 308414 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r308414 | mnicholson | 2011-02-21 09:00:22 -0600
+ (Mon, 21 Feb 2011) | 12 lines Merged revisions 308413 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb
+ 2011) | 5 lines Properly check the bounds of arrays when decoding
+ UDPTL packets. Also, remove broken support for receiving UDPTL
+ packets larger than 16k. That shouldn't ever happen anyway.
+ AST-2011-002 FAX-281 ........ ................ ................
+
+2011-02-21 14:14 +0000 [r308372] Andrew Latham <lathama@gmail.com>
+
+ * main/http.c: Add HTTP URI Debug logging and update notice enable
+ reporting of the request URI / URL in debugging change funny
+ debug note to a serious note.
+
+2011-02-21 13:58 +0000 [r308371] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * main/pbx.c: fix a memory leak in device state The callback
+ handle_statechange (pbx.c) fails to release its data pointer,
+ leaking memory in the process. Reported by: tzafrir Patches:
+ 18735_pbx_free_callback.diff uploaded by tzafrir (license 46)
+ Review: https://reviewboard.asterisk.org/r/1110/
+
+2011-02-19 14:07 +0000 [r308331] Andrew Latham <lathama@gmail.com>
+
+ * main/http.c: Add CSS MIME Type Modern browsers are checking for
+ the MIME Type of pages and in some cases will not load a file if
+ the type is wrong.
+
+2011-02-19 11:03 +0000 [r308289] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * utils, /: Merged revisions 308288 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r308288 | tilghman | 2011-02-19 05:02:49 -0600 (Sat, 19 Feb 2011)
+ | 2 lines A few more (copies of) files to ignore in this
+ directory. ........
+
+2011-02-18 00:11 +0000 [r308243] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, /, addons/ooh323cDriver.c,
+ addons/ooh323cDriver.h: Merged revisions 308242 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r308242 | may | 2011-02-18 03:07:20 +0300 (Fri, 18 Feb 2011) | 3
+ lines added g729onlyA option for announce only AnnexA g.729 codec
+ in h.323 capabilities. Option can be global or per user/peer.
+ ........
+
+2011-02-17 20:21 +0000 [r308205] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Add more verbage to CLI command 'pri show
+ channels' usage.
+
+2011-02-16 22:02 +0000 [r308157] Paul Belanger <pabelanger@digium.com>
+
+ * /, addons/ooh323c/src/ooSocket.c: Merged revisions 308150 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r308150 | pabelanger | 2011-02-16 15:21:17 -0500 (Wed, 16 Feb
+ 2011) | 2 lines Fix FreeBSD builds. ........
+
+2011-02-16 08:06 +0000 [r308099] Alexandr Anikin <may@telecom-service.ru>
+
+ * /, addons/ooh323c/src/ooSocket.c: Merged revisions 308098 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r308098 | may | 2011-02-16 10:57:22 +0300 (Wed, 16 Feb 2011) | 2
+ lines ifdef __linux__ keepalive variables also ........
+
+2011-02-15 23:34 +0000 [r308013] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 308010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r308010 | qwell | 2011-02-15 17:34:03 -0600
+ (Tue, 15 Feb 2011) | 24 lines Merged revisions 308007 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r308007 | qwell | 2011-02-15 17:33:24 -0600
+ (Tue, 15 Feb 2011) | 17 lines Merged revisions 308002 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) |
+ 10 lines Fix regression that changed behavior of queues when
+ ringing a queue member. This reverts r298596, which was to fix a
+ highly bizarre and contrived issue with a queue member that
+ called into his own queue being transferred back into his own
+ queue. I couldn't reproduce that issue in any way. I think one of
+ the other recent transfer fixes actually fixed this. (closes
+ issue #18747) Reported by: vrban ........ ................
+ ................
+
+2011-02-15 23:07 +0000 [r307969] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooSocket.c: include tcp keepalive socket calls
+ only on linux, freebsd and others don't have these options on
+ sockets.
+
+2011-02-15 21:42 +0000 [r307963-307964] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Add CLI "pri show channels" command. List the current mapping of
+ DAHDI B channels to Asterisk channel names and which calls are on
+ hold or call-waiting. Calls on hold or call-waiting are not
+ associated with any B channel. JIRA LIBPRI-27 JIRA SWP-2547
+
+ * apps/app_dial.c, /: Merged revisions 307962 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r307962 | rmudgett | 2011-02-15 13:52:45 -0600 (Tue, 15 Feb 2011)
+ | 1 line Don't crash when forcing caller id. ........
+
+2011-02-15 18:09 +0000 [r307927] David Vossel <dvossel@digium.com>
+
+ * channels/chan_phone.c: Fixes compile error in chan_phone for big
+ endian
+
+2011-02-15 16:18 +0000 [r307883] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, /,
+ channels/chan_sip.c, main/ccss.c, channels/sig_pri.c,
+ include/asterisk/ccss.h: Merged revisions 307879 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15
+ Feb 2011) | 37 lines No response sent for SIP CC
+ subscribe/resubscribe request. Asterisk does not send a response
+ if we try to subscribe for call completion after we have received
+ a 180 Ringing. You can only subscribe for call completion when
+ the call has been cleared. When we receive the 180 Ringing, for
+ this call, its call-completion state is 'CC_AVAILABLE'. If we
+ then send a subscribe message to Asterisk, it trys to change the
+ call-completion state to 'CC_CALLER_REQUESTED'. Because this is
+ an invalid state change, it just ignores the message. The only
+ state Asterisk will accept our subscribe message is in the
+ 'CC_CALLER_OFFERED' state. Asterisk will go into the
+ 'CC_CALLER_OFFERED' when the SIP client clears the call by
+ sending a CANCEL. Asterisk should always send a response. Even if
+ its a negative one. The fix is to allow for the CCSS core to
+ notify a CC agent that a failure has occurred when CC is
+ requested. The "ack" callback is replaced with a "respond"
+ callback. The "respond" callback has a parameter indicating
+ either a successful response or a specific type of failure that
+ may need to be communicated to the requester. (closes issue
+ #18336) Reported by: GeorgeKonopacki Tested by: mmichelson,
+ rmudgett JIRA SWP-2633 (closes issue #18337) Reported by:
+ GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 ........
+
+2011-02-15 07:03 +0000 [r307751-307838] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, funcs/func_odbc.c: Merged revisions 307837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r307837 | tilghman | 2011-02-15 01:02:45 -0600
+ (Tue, 15 Feb 2011) | 15 lines Merged revisions 307836 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011)
+ | 8 lines Need to retrieve the rows affected before using the
+ associated variable. (closes issue #18795) Reported by: irroot
+ Patches: 20110211__issue18795.diff.txt uploaded by tilghman
+ (license 14) Tested by: tilghman ........ ................
+
+ * /, res/res_odbc.c: Merged revisions 307793 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r307793 | tilghman | 2011-02-14 14:16:55 -0600
+ (Mon, 14 Feb 2011) | 15 lines Merged revisions 307792 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r307792 | tilghman | 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011)
+ | 8 lines Increment usage count at first reference, to avoid a
+ race condition with many threads creating connections all at
+ once. (issue #18156) Reported by: asgaroth Patches:
+ 20110214__issue18156.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........ ................
+
+ * addons/chan_ooh323.c, addons/ooh323c/src/ooCmdChannel.c: Making
+ trunk compile again.
+
+ * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 307750 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011)
+ | 23 lines Calling a gosub routine defined in AEL from Dial/Queue
+ ceased to work. A bug in AEL did not distinguish between the "s"
+ extension generated by AEL and an "s" extension that was required
+ to exist by the chan_dahdi (or another channel) that was not
+ supplied with a starting extension. Therefore, AEL made incorrect
+ assumptions about what commands were permissable in the context.
+ This was fixed by making AEL generate a different extension name.
+ However, Dial and Queue make additional assumptions about the
+ name of the default gosub extension. Therefore, they needed to be
+ brought into line with a "macro" rendered by AEL (as a gosub),
+ without breaking traditional dialplans written without the aid of
+ AEL. Related to (issue #18480) Reported by: nivek (closes issue
+ #18729) Reported by: kkm Patches: 20110209__issue18729.diff.txt
+ uploaded by tilghman (license 14)
+ 018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888)
+ Tested by: kkm ........
+
+2011-02-13 10:50 +0000 [r307677-307713] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooLogChan.c,
+ addons/ooh323c/src/ooCmdChannel.c: lc not found - it's warning,
+ not error, change malloc to ast_calloc again
+
+ * addons/chan_ooh323.c, addons/ooh323cDriver.c: change malloc to
+ ast_calloc calls to prevent crash of asterisk
+
+2011-02-10 22:43 +0000 [r307537] Jason Parker <jparker@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /, main/asterisk.c: Merged
+ revisions 307536 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r307536 | qwell | 2011-02-10 16:39:30 -0600
+ (Thu, 10 Feb 2011) | 22 lines Merged revisions 307535 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r307535 | qwell | 2011-02-10 16:35:49 -0600
+ (Thu, 10 Feb 2011) | 15 lines Merged revisions 307534 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) |
+ 8 lines Remove color when executing commands via a remote
+ console. Essentially this makes '-x' imply '-n' on rasterisk.
+ This was done in a different and incomplete way previously, which
+ I'm reverting here. (issue #18776) Reported by: alecdavis
+ ........ ................ ................
+
+2011-02-10 17:45 +0000 [r307468] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/ccss.conf.sample: Merged revisions 307467 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r307467 | mmichelson | 2011-02-10 11:44:42 -0600 (Thu,
+ 10 Feb 2011) | 5 lines Fix a gaffe in the CCSS sample
+ configuration. Discovered by Philippe Lindheimer and pointed out
+ on #asterisk-dev ........
+
+2011-02-10 17:12 +0000 [r307433] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, main/format_cap.c,
+ include/asterisk/format_cap.h: Fixes bug in chan_sip where
+ nativeformats are not set correctly. The nativeformats field was
+ being overwritten when it should have been appended too. This
+ caused some format capabilities to be lost briefly and some log
+ warnings to be output.
+
+2011-02-10 13:29 +0000 [r307396] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h,
+ addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c:
+ Corrections for properly work with H.323v2 (older) endpoints and
+ other small fixes. Interpret remote side H.225 version.
+ Corrections for H.323v2 endpoints: don't start TCS and MSD before
+ connect, don't start TCS and MSD by accepting H.245 connection,
+ start TCS and MSD by StartH245 facility message. Other fixes: fix
+ non zeroended remoteDisplayName issue, small fixes in call
+ clearing by closing H.245 connection, tcp keepalive introduced on
+ TCP connections (now is hardcoded, will be configurable in the
+ future), don't force H.245tunneling if FastStart is active, don't
+ send Alerting singal more than once per call. (closes issue
+ #18542) Reported by: vmikhelson Patches: issue18542-final-3.patch
+ uploaded by may213 (license 454) Tested by: vmikhelson
+
+2011-02-09 22:48 +0000 [r307359] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c, CHANGES: Add new manager action
+ MeetmeListRooms. From the submitter: I've added a new manager
+ action to list only the active conferences on an Asterisk system.
+ It shows the same data displayed when you run a 'meetme list' on
+ the Asterisk CLI. (closes issue #17905) Reported by: rcasas
+ Patches: app_meetme.c.patch uploaded by rcasas (license 641)
+ Review: https://reviewboard.asterisk.org/r/874/
+
+2011-02-09 21:46 +0000 [r307315] Andrew Latham <lathama@gmail.com>
+
+ * contrib/init.d/rc.debian.asterisk: Disable color during running
+ test (closes issue #18776) Reported by: alecdavis Patches:
+ ast_deb_init.diff uploaded by lathama (license 1028) Tested by:
+ andrel, lathama
+
+2011-02-09 21:08 +0000 [r307229-307274] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/astobj2.c: Merged revisions 307273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r307273 | jpeeler | 2011-02-09 15:06:33 -0600 (Wed, 09 Feb 2011)
+ | 8 lines Add missing debug info for ao2_link for use with
+ REF_DEBUG in ao2 callback. (closes issue #18758) Reported by:
+ rgagnon Patches: branch-1.8-r306540-astobj-fix.diff uploaded by
+ rgagnon (license 1202) trunk-r306540-astobj-fix.diff uploaded by
+ rgagnon (license 1202) ........
+
+ * main/features.c, CHANGES: Allow parkedmusicclass to be settable
+ for non-default parking lots. (closes issue #17946) Reported by:
+ bluecrow76 Patches:
+ asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff
+
+ * /, main/features.c: Merged revisions 307228 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r307228 | jpeeler | 2011-02-09 13:52:51 -0600
+ (Wed, 09 Feb 2011) | 17 lines Merged revisions 307227 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011)
+ | 11 lines Make sure to set parking dial context for non-default
+ parking lots. Since parking_con_dial isn't settable, set all
+ parking lots to "park-dial". (closes issue #17946) Reported by:
+ bluecrow76 Patches:
+ asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by
+ bluecrow76 (license 270) modified by me ........ ................
+
+2011-02-09 19:17 +0000 [r307192] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * main/loader.c: clarify warning when no loadable module support
+ Clarify warning message when LOADABLE_MODULES is disabled but we
+ still try to load a module.
+
+2011-02-09 05:53 +0000 [r307143] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/lock.c, /: Merged revisions 307142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r307142 | tilghman | 2011-02-08 23:39:39 -0600 (Tue, 08 Feb 2011)
+ | 3 lines Initialize tracking variable in structure properly.
+ Fixes a memory leak. (Reported by The_Boy_Wonder on IRC, fixed by
+ me.) ........
+
+2011-02-08 21:24 +0000 [r307097] Jason Parker <jparker@digium.com>
+
+ * /, main/logger.c: Merged revisions 307092 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r307092 | qwell | 2011-02-08 15:24:01 -0600 (Tue, 08 Feb 2011) |
+ 9 lines Fix issue with verbose messages not showing on remote
+ console. This code was reworked recently, and since the
+ logchannel list hadn't been created yet at this point, and it was
+ a verbose message, it was being dropped on the floor. Now it'll
+ continue on to where it should be handled. (closes issue #18580)
+ Reported by: pabelanger ........
+
+2011-02-08 21:18 +0000 [r307071] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/ccss.c: Merged revisions 307065 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r307065 | mmichelson | 2011-02-08 15:13:08 -0600 (Tue, 08 Feb
+ 2011) | 6 lines Add a couple of useful channel variables for the
+ CC recall macro. CC_EXTEN and CC_CONTEXT will allow you to
+ determine the channel and context that will be called when the
+ recall occurs. ........
+
+2011-02-08 20:42 +0000 [r307061] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 306979 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r306979 | twilson | 2011-02-08 12:18:08 -0800
+ (Tue, 08 Feb 2011) | 16 lines Merged revisions 306973 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306973 | twilson | 2011-02-08 12:14:09 -0800
+ (Tue, 08 Feb 2011) | 9 lines Merged revisions 306972 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08
+ Feb 2011) | 2 lines Fix comparison for REFER Replaces tags with
+ pedantic=yes ........ ................ ................
+
+2011-02-08 20:31 +0000 [r307041] Andrew Latham <lathama@gmail.com>
+
+ * /, doc/asterisk.8, configs/asterisk.conf.sample,
+ configs/voicemail.conf.sample, doc/asterisk.sgml: Documentation
+ Updates Note default polling setting in voicemail.conf Add
+ missing config to asterisk.conf Update manpage (issue #16505)
+ Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff
+ uploaded by tzafrir (license 46) Tested by: lathama, tzafrir
+
+2011-02-08 19:42 +0000 [r306867-306968] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 306967 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r306967 | jpeeler | 2011-02-08 13:41:42 -0600
+ (Tue, 08 Feb 2011) | 16 lines Merged revisions 306966 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306966 | jpeeler | 2011-02-08 13:41:21 -0600
+ (Tue, 08 Feb 2011) | 9 lines Merged revisions 306965 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08
+ Feb 2011) | 1 line fix this line again ........ ................
+ ................
+
+ * /, apps/app_voicemail.c: Merged revisions 306962 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r306962 | jpeeler | 2011-02-08 13:25:38 -0600
+ (Tue, 08 Feb 2011) | 22 lines Merged revisions 306961 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306961 | jpeeler | 2011-02-08 13:25:10 -0600
+ (Tue, 08 Feb 2011) | 15 lines Merged revisions 306960 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011)
+ | 9 lines Backup file storing message duration is not used with
+ IMAP_STORAGE, remove code. The message duration is stored in the
+ body of the email when using IMAP_STORAGE, so nothing needs to
+ happen with the backup file. (closes issue #18718) Reported by:
+ kerframil ........ ................ ................
+
+ * /, apps/app_voicemail.c: Merged revisions 306866 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r306866 | jpeeler | 2011-02-08 10:21:45 -0600
+ (Tue, 08 Feb 2011) | 16 lines Merged revisions 306865 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306865 | jpeeler | 2011-02-08 10:21:25 -0600
+ (Tue, 08 Feb 2011) | 9 lines Merged revisions 306864 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08
+ Feb 2011) | 1 line make this safer and fully correct, pointed out
+ by Steve Davis ........ ................ ................
+
+2011-02-08 02:05 +0000 [r306827] Andrew Latham <lathama@gmail.com>
+
+ * doc/asterisk.sgml: Documentation Updates. Start updates to the
+ man pages. (issue #16505) Reported by: tzafrir Tested by: lathama
+
+2011-02-08 00:43 +0000 [r306755-306793] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/chan_dahdi.conf.sample: Define the MCID acronym in
+ chan_dahdi.conf.sample.
+
+ * channels/sig_pri.h: Use correct conditional for MCID send.
+
+ * channels/sig_pri.h, main/channel.c, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, funcs/func_frame_trace.c,
+ main/features.c, CHANGES, channels/sig_pri.c,
+ include/asterisk/frame.h: Pass a MCID request to the bridged
+ channel. Pass a MCID request to the bridged channel so the
+ bridged channel can send it to the network. The ability to send
+ the MCID request on an ISDN span is enabled with the new
+ chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736
+
+2011-02-07 22:46 +0000 [r306670-306675] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 306674 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r306674 | twilson | 2011-02-07 14:43:22 -0800
+ (Mon, 07 Feb 2011) | 24 lines Merged revisions 306673 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306673 | twilson | 2011-02-07 14:40:20 -0800
+ (Mon, 07 Feb 2011) | 17 lines Merged revisions 306672 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011)
+ | 10 lines Don't try to pickup a call in the middle of a
+ masquerade If A calls B which doesn't answer and C & D both try
+ to do a call pickup, it is possible for ast_pickup_call to answer
+ the call, then fail to masquerade one of the calls because the
+ other one is already in the process of masquerading. This patch
+ checks to see if the channel is in the process of masquerading
+ before call before selecting it for a pickup. Review:
+ https://reviewboard.asterisk.org/r/1094/ ........
+ ................ ................
+
+ * /, channels/chan_sip.c: Merged revisions 306619 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r306619 | twilson | 2011-02-07 14:15:27 -0800
+ (Mon, 07 Feb 2011) | 24 lines Merged revisions 306618 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306618 | twilson | 2011-02-07 13:59:54 -0800
+ (Mon, 07 Feb 2011) | 17 lines Merged revisions 306617 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011)
+ | 10 lines Don't allow a REFER w/replaces to replace its own
+ dialog Asterisk currently accepts a REFER with a Refer-To with an
+ embedded Replaces header that matches the dialog of the REFER.
+ This would be a situation like A calls B, A calls C, A transfers
+ B to A, which is just silly. This patch makes the transfer fail
+ instead of making Asterisk freak out and forget to hang other
+ channels up. Review: https://reviewboard.asterisk.org/r/1093/
+ ........ ................ ................
+
+2011-02-07 17:55 +0000 [r306576] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/ccss.c: Merged revisions 306575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r306575 | mmichelson | 2011-02-07 11:36:56 -0600 (Mon, 07 Feb
+ 2011) | 9 lines Rearrange a bit of code in the generic CC recall
+ operation. By waiting to call the callback macro after the
+ CC_INTERFACES, extension, priority, and context have been set,
+ this information can be accessed more easily within the callback
+ macro. Reported by Philippe Lindheimer. ........
+
+2011-02-07 16:33 +0000 [r306541] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes use of ast_format_cap_append where
+ ast_format_cap_copy is necessary.
+
+2011-02-05 22:16 +0000 [r306499] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: fix trivial issue after dvossel patch,
+ initial zero fill user and peer structure before cap structure
+ allocated.
+
+2011-02-05 02:55 +0000 [r306464] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Ignore voice frames in chan_dahdi native
+ bridging. Hardware is handling them.
+
+2011-02-04 22:37 +0000 [r306432] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c: Send manager event for blackfilter only if it
+ DOES NOT match. The logic got reversed, oops. Works properly now
+ when multiple blackfilters are present. (closes issue #18283)
+ Reported by: telecos82 Patches: ast_managereventfilter.patch
+ uploaded by telecos82 (license 687)
+
+2011-02-04 20:30 +0000 [r306396] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
+ channels/sig_pri.c: Add ISDN display ie text handling options to
+ chan_dahdi.conf. The display ie handling can be controlled
+ independently in the send and receive directions with the
+ following options: * Block display text data. * Use display text
+ in SETUP/CONNECT messages for name. * Use display text for COLP
+ name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary
+ display text during a call. Sent in INFORMATION messages.
+ Received from any message that the display text was not used as a
+ name. If the display options are not set then the options default
+ to legacy behavior. The arbitrary display text is exchanged
+ between bridged channels using the AST_FRAME_TEXT frame type. To
+ send display text from the dialplan use the SendText()
+ application when the arbitrary display text option is enabled.
+ JIRA SWP-2688 JIRA ABE-2693
+
+2011-02-04 19:24 +0000 [r306359] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 306356 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r306356 | qwell | 2011-02-04 13:24:29 -0600
+ (Fri, 04 Feb 2011) | 16 lines Merged revisions 306346 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) |
+ 9 lines Don't fallthrough to 'unknown' in the 'ringing' case.
+ This could cause improper exits from the queue. (closes issue
+ #18499) Reported by: zaltar Patches: app_queue.patch uploaded by
+ zaltar (license 1148) ........ ................
+
+2011-02-04 19:09 +0000 [r306325-306326] Richard Mudgett <rmudgett@digium.com>
+
+ * tests/test_format_api.c: Fix compiler warning.
+
+ * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 306324 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r306324 | rmudgett | 2011-02-04 12:53:06 -0600 (Fri, 04 Feb 2011)
+ | 9 lines Don't send redirecting updates to the caller if the
+ dialplan forked the call. Each fork in the dial could be
+ redirected and confuse the caller. For ISDN the DivLeg1 and
+ DivLeg3 messages would get confused because ISDN redirects calls
+ in sequence not in parallel. * Also fixed a formatting
+ inconsistency in app_dial.c and make a warning message more
+ useful about what frame type could not be written. ........
+
+2011-02-04 18:16 +0000 [r306258-306292] Paul Belanger <pabelanger@digium.com>
+
+ * utils/extconf.c: Revert changes to extconf.c It seems extconf.c
+ already defines some local ast_debug() functions. Theses should
+ be removed and replaced with logger.h. A patch will be added to
+ reviewboard shortly.
+
+ * cel/cel_radius.c, addons/chan_ooh323.c, apps/app_meetme.c,
+ main/say.c, channels/chan_gtalk.c, main/taskprocessor.c,
+ res/res_http_post.c, res/res_musiconhold.c, channels/chan_iax2.c,
+ res/res_jabber.c, pbx/pbx_loopback.c, main/channel.c,
+ channels/chan_dahdi.c, pbx/pbx_spool.c, main/manager.c,
+ res/res_smdi.c, channels/chan_skinny.c, main/features.c,
+ res/res_agi.c, main/http.c, main/logger.c, res/ais/evt.c,
+ main/app.c, res/res_config_ldap.c, apps/app_rpt.c,
+ res/res_rtp_asterisk.c, main/pbx.c, channels/chan_sip.c,
+ apps/app_fax.c, include/asterisk/channel.h, channels/sig_pri.c,
+ channels/chan_misdn.c, include/asterisk/sched.h, utils/extconf.c,
+ codecs/codec_ilbc.c, main/audiohook.c, res/res_odbc.c,
+ main/xmldoc.c, apps/app_voicemail.c: Replace ast_log(LOG_DEBUG,
+ ...) with ast_debug() (closes issue #18556) Reported by: kkm
+ Review: https://reviewboard.asterisk.org/r/1071/
+
+2011-02-04 16:42 +0000 [r306257] David Vossel <dvossel@digium.com>
+
+ * codecs/codec_ilbc.c, codecs/ex_ilbc.h: Fix compile error in codec
+ ilbc translator.
+
+2011-02-03 23:50 +0000 [r306216] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 306215 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r306215 | jpeeler | 2011-02-03 17:49:28 -0600 (Thu, 03 Feb 2011)
+ | 20 lines Fix SIP deadlock involving state changes. Once again a
+ call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper)
+ has caused locking problems. Both of these functions lock the
+ channel when the channel argument is passed in! In this case, the
+ suspected problem (the backtrace makes it impossible to tell) was
+ the private being locked in sip_set_rtp_peer and then:
+ transmit_reinvite_with_sdp try_suggested_sip_codec
+ pbx_builtin_getvar_helper (Traced to verify that the fix was only
+ required in 1.8 and later.) (closes issue #18491) Reported by:
+ cmaj Patches: chan_sip_fix_deadlocks_bug_18491.txt uploaded by
+ cmaj (license 830) Tested by: cmaj ........
+
+2011-02-03 21:13 +0000 [r306128] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 306127 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r306127 | twilson | 2011-02-03 13:03:26 -0800
+ (Thu, 03 Feb 2011) | 23 lines Merged revisions 306126 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306126 | twilson | 2011-02-03 12:56:00 -0800
+ (Thu, 03 Feb 2011) | 16 lines Merged revisions 306119 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011)
+ | 9 lines Set hangup cause in local_hangup When a call involves a
+ local channel (like SIP -> Local -> SIP), the hangup cause was
+ not being set. This resulted in SIP channels sometimes getting a
+ 503 error instead of a 486 when the far side sent a busy. In
+ Asterisk 1.8+ this also can cause issues with CCSS that involve a
+ local channel. This patch sets the hangupcause for one side of
+ the local channel to the other in local_hangup for outbound
+ calls. ........ ................ ................
+
+2011-02-03 20:51 +0000 [r306125] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 306124 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r306124 | jpeeler | 2011-02-03 14:50:48 -0600
+ (Thu, 03 Feb 2011) | 17 lines Merged revisions 306123 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011)
+ | 10 lines Set exception on channel in parking thread when
+ POLLPRI event detected. This is done just to make the code be
+ equivalent to the old select code. As noted in 303106 the same
+ issue was already fixed in this branch, but the exception was not
+ set on the channel in the case of POLLPRI. The reason that this
+ did not cause a problem here is because in 122923 the check in
+ __ast_read to check the exception flag was removed. (related to
+ #18637) ........ ................
+
+2011-02-03 18:37 +0000 [r306086] Jason Parker <jparker@digium.com>
+
+ * main/frame.c: Modify alignment of 'core show codecs', since the
+ ID is no longer a huge int.
+
+2011-02-03 18:12 +0000 [r306010-306053] David Vossel <dvossel@digium.com>
+
+ * main/frame.c: Fixes output of "core show codecs" to display image
+ types correctly.
+
+ * apps/app_dahdibarge.c, channels/chan_local.c, main/frame.c,
+ apps/app_record.c, apps/app_alarmreceiver.c,
+ bridges/bridge_softmix.c, formats/format_sln16.c,
+ apps/app_ices.c, bridges/bridge_multiplexed.c,
+ channels/chan_iax2.c, main/astobj2.c, res/res_rtp_multicast.c,
+ channels/chan_dahdi.c, include/asterisk/bridging_technology.h,
+ funcs/func_pitchshift.c, pbx/pbx_spool.c,
+ include/asterisk/audiohook.h, channels/chan_skinny.c,
+ channels/sip/include/globals.h, apps/app_dumpchan.c,
+ formats/format_pcm.c, formats/format_h263.c, main/bridging.c,
+ codecs/ex_ulaw.h, channels/sip/include/sip.h, main/pbx.c,
+ codecs/codec_g722.c, formats/format_wav.c, codecs/codec_g726.c,
+ formats/format_ogg_vorbis.c, bridges/bridge_simple.c,
+ include/asterisk/channel.h, apps/app_talkdetect.c,
+ channels/iax2-parser.c, include/asterisk/format_cap.h (added),
+ apps/app_speech_utils.c, channels/iax2-parser.h, main/data.c,
+ funcs/func_channel.c, main/audiohook.c, codecs/codec_dahdi.c,
+ include/asterisk/frame_defs.h, formats/format_g726.c,
+ apps/app_mixmonitor.c, main/asterisk.c, res/res_calendar.c,
+ apps/app_voicemail.c, channels/chan_vpb.cc, addons/format_mp3.c,
+ formats/format_sln.c, apps/app_dictate.c, codecs/ex_g722.h,
+ codecs/codec_gsm.c, codecs/ex_g726.h, channels/chan_gtalk.c,
+ include/asterisk/abstract_jb.h, main/channel.c, apps/app_mp3.c,
+ codecs/codec_resample.c, formats/format_h264.c,
+ formats/format_siren14.c, apps/app_rpt.c, channels/chan_mgcp.c,
+ codecs/codec_lpc10.c, channels/chan_sip.c, codecs/ex_lpc10.h,
+ include/asterisk/format_pref.h (added), codecs/codec_alaw.c,
+ res/res_adsi.c, tests/test_format_api.c (added),
+ apps/app_originate.c, channels/chan_jingle.c,
+ formats/format_vox.c, main/abstract_jb.c,
+ include/asterisk/bridging.h, main/callerid.c, main/file.c,
+ apps/app_sms.c, formats/format_g723.c, main/dsp.c, main/format.c
+ (added), main/udptl.c, main/rtp_engine.c, addons/chan_ooh323.c,
+ codecs/codec_adpcm.c, apps/app_test.c, addons/chan_ooh323.h,
+ include/asterisk/speech.h, codecs/ex_adpcm.h, codecs/ex_alaw.h,
+ formats/format_wav_gsm.c, include/asterisk/data.h,
+ codecs/ex_gsm.h, main/indications.c, main/format_pref.c (added),
+ main/cli.c, main/features.c, include/asterisk/mod_format.h,
+ apps/app_amd.c, addons/ooh323cDriver.c, channels/chan_alsa.c,
+ formats/format_jpeg.c, addons/ooh323cDriver.h,
+ formats/format_gsm.c, apps/app_milliwatt.c, res/res_speech.c,
+ formats/format_g719.c, channels/h323/ast_h323.cxx,
+ channels/chan_bridge.c, apps/app_echo.c, apps/app_fax.c,
+ codecs/codec_speex.c, include/asterisk/slin.h,
+ channels/chan_agent.c, channels/iax2-provision.c,
+ codecs/ex_speex.h, channels/chan_misdn.c,
+ include/asterisk/image.h, channels/iax2.h, codecs/codec_ilbc.c,
+ apps/app_chanspy.c, res/res_fax_spandsp.c,
+ include/asterisk/slinfactory.h, include/asterisk/translate.h,
+ channels/chan_unistim.c, channels/chan_multicast_rtp.c,
+ main/ccss.c, apps/app_meetme.c, res/res_musiconhold.c,
+ apps/app_followme.c, formats/format_siren7.c,
+ formats/format_ilbc.c, include/asterisk/file.h,
+ include/asterisk/callerid.h, channels/chan_phone.c, main/dial.c,
+ main/manager.c, main/format_cap.c (added),
+ funcs/func_frame_trace.c, res/res_agi.c, main/app.c,
+ apps/app_confbridge.c, include/asterisk/format.h (added),
+ main/image.c, include/asterisk/rtp_engine.h,
+ include/asterisk/frame.h, addons/chan_mobile.c,
+ apps/app_parkandannounce.c, apps/app_jack.c,
+ res/res_clioriginate.c, res/res_rtp_asterisk.c,
+ apps/app_nbscat.c, codecs/codec_a_mu.c, res/res_fax.c,
+ apps/app_festival.c, apps/app_waitforsilence.c,
+ include/asterisk/astobj2.h, main/slinfactory.c, main/translate.c,
+ channels/chan_console.c, channels/h323/chan_h323.h,
+ channels/chan_oss.c, channels/chan_usbradio.c,
+ channels/chan_h323.c, codecs/codec_ulaw.c,
+ include/asterisk/pbx.h, channels/chan_nbs.c,
+ formats/format_g729.c: Asterisk media architecture conversion -
+ no more format bitfields This patch is the foundation of an
+ entire new way of looking at media in Asterisk. The code present
+ in this patch is everything required to complete phase1 of my
+ Media Architecture proposal. For more information about this
+ project visit the link below.
+ https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
+ The primary function of this patch is to convert all the usages
+ of format bitfields in Asterisk to use the new format and
+ format_cap APIs. Functionally no change in behavior should be
+ present in this patch. Thanks to twilson and russell for all the
+ time they spent reviewing these changes. Review:
+ https://reviewboard.asterisk.org/r/1083/
+
+2011-02-03 16:13 +0000 [r305988] Andrew Latham <lathama@gmail.com>
+
+ * phoneprov/snom-mac.xml (added), configs/phoneprov.conf.sample:
+ res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support
+ (issue #18713) Reported by: lathama Patches: snom_dir.diff
+ uploaded by lathama (license 1028) Tested by: lathama
+
+2011-02-03 00:29 +0000 [r305939] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, main/manager.c, /, channels/chan_sip.c,
+ apps/app_sendtext.c: Merged revisions 305923 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r305923 | rmudgett | 2011-02-02 18:24:40 -0600
+ (Wed, 02 Feb 2011) | 24 lines Merged revisions 305889 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600
+ (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011)
+ | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null
+ terminator in the buffer length. When the frame is queued it is
+ copied. If the null terminator is not part of the frame buffer
+ length, the receiver could see garbage appended onto it. * Add
+ channel lock protection with ast_sendtext(). * Fixed AMI SendText
+ action ast_sendtext() return value check. ........
+ ................ ................
+
+2011-02-02 20:06 +0000 [r305845] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, funcs/func_env.c: Merged revisions 305844 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r305844 | tilghman | 2011-02-02 14:05:43 -0600 (Wed, 02 Feb 2011)
+ | 5 lines Eliminate a file descriptor leak when using the FILE()
+ dialplan function. (closes issue #18731) Reported by: marioabajo
+ ........
+
+2011-02-02 19:30 +0000 [r305759-305843] Andrew Latham <lathama@gmail.com>
+
+ * configs/iax.conf.sample, funcs/func_enum.c,
+ configs/dundi.conf.sample, funcs/func_callcompletion.c, /,
+ configs/mgcp.conf.sample, configs/iaxprov.conf.sample,
+ configs/unistim.conf.sample, apps/app_externalivr.c,
+ configs/sip.conf.sample, configs/skinny.conf.sample,
+ configs/h323.conf.sample, configs/sla.conf.sample,
+ apps/app_voicemail.c: Replacing doc/* and asterisk.pdf with wiki
+ links Adding links to http(s)://wiki.asterisk.org
+
+ * configs/chan_dahdi.conf.sample, /, configs/extconfig.conf.sample,
+ configs/res_snmp.conf.sample, main/ast_expr2f.c,
+ res/res_timing_dahdi.c, configs/ccss.conf.sample,
+ configs/sip.conf.sample, configs/skinny.conf.sample,
+ main/config.c, configs/h323.conf.sample, configs/sla.conf.sample,
+ main/ast_expr2.fl, res/res_srtp.c: Replacing doc/* with wiki
+ links Adding links to http(s)://wiki.asterisk.org
+
+ * /, channels/chan_sip.c: Replace link to old doc with new wiki
+ page. Link to
+ https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
+
+2011-02-01 22:48 +0000 [r305693] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 305692 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r305692 | qwell | 2011-02-01 16:48:16 -0600 (Tue, 01 Feb
+ 2011) | 7 lines Reverse sense of an error test when reading from
+ astdb. (closes issue #18545) Reported by: jcovert Patches:
+ chan_iax2.c.patch uploaded by jcovert (license 551) ........
+
+2011-02-01 21:16 +0000 [r305650] Andrew Latham <lathama@gmail.com>
+
+ * configs/sip.conf.sample: SIP Configuration Documentation sip show
+ settings reports qualifyfreq in milliseconds. sip.conf configures
+ qualifyfreg in seconds.
+
+2011-02-01 19:27 +0000 [r305604] Brett Bryant <bbryant@digium.com>
+
+ * cel/cel_pgsql.c, /: Merged revisions 305603 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r305603 | bbryant | 2011-02-01 14:23:20 -0500 (Tue, 01 Feb 2011)
+ | 4 lines Add a possible solution to a customer problem with
+ reloading cel_pgsql.so quickly. ........
+
+2011-02-01 18:03 +0000 [r305561] Andrew Latham <lathama@gmail.com>
+
+ * /: doc/tex dir removed, but corresponding entries still exists
+ Update README, CHANGES, and Makefile. Direct users to
+ http://wiki.asterisk.org for documentation or to the AST.txt and
+ AST.pdf included in the tarball. (closes issue #18443) Reported
+ by: bas Patches: changes.diff uploaded by lathama (license 1028)
+ readme.diff uploaded by lathama (license 1028) Tested by: lathama
+ bas
+
+2011-02-01 17:05 +0000 [r305474] Jason Parker <jparker@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 305473 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r305473 | qwell | 2011-02-01 11:04:23 -0600
+ (Tue, 01 Feb 2011) | 23 lines Merged revisions 305472 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305472 | qwell | 2011-02-01 11:02:09 -0600
+ (Tue, 01 Feb 2011) | 16 lines Merged revisions 305471 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305471 | qwell | 2011-02-01 11:00:55 -0600 (Tue, 01 Feb 2011) |
+ 9 lines Close file descriptor for timing source when a MOH class
+ gets destroyed. (closes issue #18457) Reported by: mcallist
+ Patches: 18457-closetimer.diff uploaded by qwell (license 4)
+ 18457-closetimer_trunk.diff uploaded by qwell (license 4) Tested
+ by: qwell, loloski ........ ................ ................
+
+2011-02-01 16:05 +0000 [r305433] Brett Bryant <bbryant@digium.com>
+
+ * apps/app_confbridge.c: Add's two features to confbridge:
+ confbridge kick, and confbridge list. (closes issue #14389)
+ (closes issue #18007) Reported by: jcollie Patches:
+ 0001-Fix-up-bridging-module-so-that-menuselect-works.patch
+ uploaded by jcollie (license 412)
+ 0002-Add-confbridge-list-and-confbridge-kick-CLI-comm.patch
+ uploaded by jcollie (license 412) Tested by: file Review:
+ https://reviewboard.asterisk.org/r/1084/
+
+2011-02-01 00:07 +0000 [r305344] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 305343 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r305343 | rmudgett | 2011-01-31 18:01:09 -0600
+ (Mon, 31 Jan 2011) | 21 lines Merged revisions 305342 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305342 | rmudgett | 2011-01-31 17:50:10 -0600
+ (Mon, 31 Jan 2011) | 14 lines Merged revisions 305341 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011)
+ | 7 lines Obtain the pri lock for PRI queue counters. Need to
+ obtain the pri lock when calling pri_dump_info_str() to avoid a
+ reentrancy problem when calculating the Q.921 Q count statistic.
+ JIRA AST-484 ........ ................ ................
+
+2011-01-31 23:08 +0000 [r305132-305255] Jason Parker <jparker@digium.com>
+
+ * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 305254
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r305254 | qwell | 2011-01-31 17:07:00 -0600
+ (Mon, 31 Jan 2011) | 24 lines Merged revisions 305253 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305253 | qwell | 2011-01-31 16:59:34 -0600
+ (Mon, 31 Jan 2011) | 17 lines Merged revisions 305252 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) |
+ 10 lines Prevent a crash when dialing a technology with no
+ destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers
+ already had code to prevent this. The attempt that app_dial was
+ making to prevent it was not correct, so I fixed that. (closes
+ issue #18371) Reported by: gbour Patches: 18371.patch uploaded by
+ gbour (license 1162) ........ ................ ................
+
+ * main/tcptls.c, /, configs/sip.conf.sample: Merged revisions
+ 305247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r305247 | qwell | 2011-01-31 16:25:23 -0600 (Mon, 31 Jan 2011) |
+ 7 lines Add alternative name for config option. The SIP sample
+ configuration had "tlscadir" as the option name, but chan_sip
+ used the more correct "tlscapath". Now both are accepted.
+ Discovered (sort of) by a user on IRC in #asterisk ........
+
+ * /, res/res_musiconhold.c: Merged revisions 305198 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r305198 | qwell | 2011-01-31 15:30:44 -0600 (Mon, 31 Jan
+ 2011) | 2 lines Fix compile error. pseudofd no longer exists.
+ ........
+
+ * /, res/res_musiconhold.c: Merged revisions 305131 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r305131 | qwell | 2011-01-31 15:00:25 -0600
+ (Mon, 31 Jan 2011) | 16 lines Merged revisions 305130 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305130 | qwell | 2011-01-31 14:59:37 -0600
+ (Mon, 31 Jan 2011) | 9 lines Merged revisions 305129 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r305129 | qwell | 2011-01-31 14:56:25 -0600 (Mon, 31 Jan
+ 2011) | 2 lines Set file descriptors to -1 on creation, so that
+ we don't see weirdness later. ........ ................
+ ................
+
+2011-01-31 13:57 +0000 [r305084] Andrew Latham <lathama@gmail.com>
+
+ * main/http.c: Asterisk HTTP response Content-type Address content
+ type for BSD and other platforms (closes issue #18456) Reported
+ by: alexo Patches: asterisk18_http.patch uploaded by alexo
+ (license 1175) Tested by: alexo
+
+2011-01-31 07:52 +0000 [r304951-305041] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, include/asterisk/lock.h: Merged revisions 305040 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r305040 | tilghman | 2011-01-31 01:51:40 -0600 (Mon, 31
+ Jan 2011) | 2 lines Use the non-specific API aliases, to avoid a
+ problem with building the utils directory. ........
+
+ * /, apps/app_voicemail.c: Merged revisions 304985 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304985 | tilghman | 2011-01-31 01:27:13 -0600
+ (Mon, 31 Jan 2011) | 16 lines Merged revisions 304978 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304978 | tilghman | 2011-01-31 01:25:14 -0600
+ (Mon, 31 Jan 2011) | 9 lines Merged revisions 304952 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31
+ Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined.
+ ........ ................ ................
+
+ * main/lock.c, /, main/heap.c, main/utils.c,
+ include/asterisk/lock.h, .cleancount: Merged revisions 304950 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r304950 | tilghman | 2011-01-31 00:41:36 -0600 (Mon, 31 Jan 2011)
+ | 18 lines Change mutex tracking so that it only consumes memory
+ in the core mutex object when it's actually being used. This
+ reduces the overall size of a mutex which was 3016 bytes before
+ this back down to 216 bytes (this is on 64-bit Linux with a
+ glibc-implemented mutex). The exactness of the numbers here may
+ vary slightly based upon how mutexes are implemented on a
+ platform, but the long and short of it is that prior to this
+ commit, chan_iax2 held down 98MB of memory on a 64-bit system for
+ nothing more than a table of 32767 locks. After this commit, the
+ same table occupies a mere 7MB of memory. (closes issue #18194)
+ Reported by: job Patches: 20110124__issue18194.diff.txt uploaded
+ by tilghman (license 14) Tested by: tilghman Review:
+ https://reviewboard.asterisk.org/r/1066 ........
+
+2011-01-30 00:22 +0000 [r304913] Andrew Latham <lathama@gmail.com>
+
+ * funcs/func_callcompletion.c, /, apps/app_externalivr.c,
+ apps/app_queue.c, apps/app_voicemail.c, funcs/func_realtime.c,
+ res/res_calendar.c: Add Function and Application Relationships to
+ documentation Add and extend the see-also sections to the
+ documentation for applications and functions in an effort to
+ expand the online documentation of the wiki. Also check for and
+ update any links to moved documentation in the doc folder.
+
+2011-01-29 23:10 +0000 [r304639-304867] Sean Bright <sean@malleable.com>
+
+ * /, res/res_config_ldap.c: Merged revisions 304866 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304866 | seanbright | 2011-01-29 18:07:18 -0500
+ (Sat, 29 Jan 2011) | 14 lines Merged revisions 304865 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304865 | seanbright | 2011-01-29 18:05:25 -0500 (Sat, 29 Jan
+ 2011) | 7 lines Plug some memory leaks in the LDAP realtime
+ driver. (closes issue #18435) Reported by: zaltar Patches:
+ res_config_ldap.patch uploaded by zaltar (license 1148) ........
+ ................
+
+ * /, apps/app_meetme.c: Merged revisions 304777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304777 | seanbright | 2011-01-29 13:09:37 -0500
+ (Sat, 29 Jan 2011) | 22 lines Merged revisions 304776 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304776 | seanbright | 2011-01-29 13:08:14 -0500 (Sat, 29 Jan
+ 2011) | 15 lines If we fail to allocate our announcement objects,
+ make sure we don't leak objects. The majority of this patch was
+ committed already in r304726 and r304729. (issue #18225) Reported
+ by: kenji (issue #18444) Reported by: junky (closes issue #18343)
+ Reported by: kobaz Patches: meetme-refs.diff uploaded by kobaz
+ (license 834) ........ ................
+
+ * /, apps/app_meetme.c: Merged revisions 304774 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304774 | seanbright | 2011-01-29 12:54:43 -0500
+ (Sat, 29 Jan 2011) | 16 lines Merged revisions 304773 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan
+ 2011) | 9 lines When we pass the S() or L() options to MeetMe,
+ make sure that we honor C as well. Without this patch, if the
+ user was kicked from the conference via the S() or L() mechanism,
+ we would just hang up on them even if we also passed C (continue
+ in dialplan when kicked). With this patch we honor the C flag in
+ those cases. (closes issue #17317) Reported by: var ........
+ ................
+
+ * /, apps/app_meetme.c: Merged revisions 304730 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304730 | seanbright | 2011-01-29 12:15:27 -0500
+ (Sat, 29 Jan 2011) | 22 lines Merged revisions 304729 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan
+ 2011) | 15 lines Make sure that we unref the correct object when
+ ejecting the most recent caller. Currently, when we kick the last
+ user to enter, we decrement our own reference count which results
+ in a crash when we kick another user or when we exit the
+ conference ourselves. This will fix #18225 in 1.8 and trunk, but
+ that particular bug does not exist in 1.6.2. (closes issue
+ #18225) Reported by: kenji Patches: issue18225.patch uploaded by
+ seanbright (license 71) Tested by: seanbright ........
+ ................
+
+ * /, apps/app_meetme.c: Merged revisions 304727 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304727 | seanbright | 2011-01-29 11:28:27 -0500
+ (Sat, 29 Jan 2011) | 16 lines Merged revisions 304726 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan
+ 2011) | 9 lines Fix user reference leak in MeetMe. We were
+ unlinking the user from the conferences user container, but not
+ decrementing the reference count of the user as well, resulting
+ in a leak. (closes issue #18444) Reported by: junky Tested by:
+ seanbright ........ ................
+
+ * /, apps/app_meetme.c: Merged revisions 304683 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304683 | seanbright | 2011-01-28 17:54:23 -0500
+ (Fri, 28 Jan 2011) | 16 lines Merged revisions 304659,304682 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri, 28 Jan
+ 2011) | 5 lines Don't leak references if we can't create a pseudo
+ channel for mixing in MeetMe. If there was a problem allocating a
+ pseudo channel when building our meetme, we weren't destroying
+ our user container or destroying the mutexes that we created.
+ ........ r304682 | seanbright | 2011-01-28 17:38:05 -0500 (Fri,
+ 28 Jan 2011) | 2 lines Revert part of the previous commit that
+ snuck in. ........ ................
+
+ * /, main/acl.c: Merged revisions 304638 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r304638 | seanbright | 2011-01-28 15:19:08 -0500 (Fri, 28 Jan
+ 2011) | 11 lines Restore some conditionals that we lost in
+ r277814. There are some cases where ast_append_ha() is called
+ with a NULL instead of a valid int pointer. So if we get a NULL,
+ don't try to dereference it. (closes issue #18162) Reported by:
+ imcdona Patches: issue0018162.patch uploaded by pabelanger
+ (license 224) Tested by: enegaard ........
+
+2011-01-27 20:09 +0000 [r304600] Brett Bryant <bbryant@digium.com>
+
+ * res/res_config_pgsql.c: Patch that fixes the "realtime show pgsql
+ cache" command crash when giving a table name, because of the use
+ of an uninitialized variable. Fixes an error introduced in
+ r300882. (closes issue #18605) Reported by: romain_proformatique
+ Patches: res_config_pgsql_fix.patch uploaded by romain
+ proformatique (license 975) Tested by: romain_proformatique
+
+2011-01-27 20:07 +0000 [r304599] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_fax.c: Fix bug with 'F' option for ReceiveFAX and
+ SendFAX. Skipping the call to set_t38_fax_caps() caused the FAX
+ session details to not be marked as supporting audio FAX
+ either... the function's name is a bit misleading. This patch
+ restores the single bit of non-T.38 behavior from that function
+ when audio mode is forced.
+
+2011-01-27 19:12 +0000 [r304555] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/ccss.c: Merged revisions 304554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r304554 | rmudgett | 2011-01-27 13:08:14 -0600 (Thu, 27 Jan 2011)
+ | 4 lines Warning message if CALLCOMPLETION(cc_callback_macro or
+ cc_agent_dialstring) are empty. Test if the value pointer is not
+ NULL instead of not ast_strlen_zero(). ........
+
+2011-01-27 17:03 +0000 [r304463-304467] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 304466 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304466 | qwell | 2011-01-27 11:03:01 -0600
+ (Thu, 27 Jan 2011) | 23 lines Merged revisions 304465 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304465 | qwell | 2011-01-27 11:01:24 -0600
+ (Thu, 27 Jan 2011) | 16 lines Merged revisions 304464 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan 2011) |
+ 9 lines Fix default prefix=/usr regression on non-Linux systems.
+ This partially reverts a change made in branches/1.4/ r267759,
+ which will cause issue #17013 to be reopened. This issue was
+ pointed out by a user on #asterisk, who helpfully discovered that
+ paths were being set incorrectly. To truly understand what was
+ wrong, one should run: svn diff --force -c<this revision>
+ configure ........ ................ ................
+
+ * /, configure: Merged revisions 304462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304462 | qwell | 2011-01-27 10:48:44 -0600
+ (Thu, 27 Jan 2011) | 16 lines Merged revisions 304461 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304461 | qwell | 2011-01-27 10:48:00 -0600
+ (Thu, 27 Jan 2011) | 9 lines Merged revisions 304460 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r304460 | qwell | 2011-01-27 10:47:03 -0600 (Thu, 27 Jan
+ 2011) | 1 line Rerun bootstrap.sh with no changes, so that it is
+ more obvious what my next commit changes. ........
+ ................ ................
+
+2011-01-27 15:57 +0000 [r304422] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_fax.c: Rename the SendFAX/ReceiveFAX 'force audio'
+ option. The recently added option to disable usage of T.38 for a
+ single session should have been named 'F' for 'force audio',
+ since that is really what the user is asking to happen (and it's
+ a positive option instead of a negative option that way).
+
+2011-01-27 00:06 +0000 [r304385] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, channels/sig_pri.c: Merged from
+ revision 304341
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r304341 | rmudgett | 2011-01-26 16:38:39 -0600 (Wed,
+ 26 Jan 2011) | 7 lines Add connected line chan_dahdi.conf
+ pricpndialplan option. * Added from_channel value to
+ prilocaldialplan option. JIRA ABE-2731 JIRA SWP-2842 ..........
+
+2011-01-26 23:41 +0000 [r304384] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_followme.c: Add option to followme to delay answer until
+ ready to bridge call. Followme answers an incoming call if it
+ hasn't already been answered and starts MOH. Some poorly designed
+ autodialers see the answer and start playing their message to the
+ hold music. The 'N' option has been added to indicate ringing and
+ not answer until the call is accepted. (closes issue #18479)
+ Reported by: ianc Patches: trunk_followme.diff uploaded by ianc
+ (license 998)
+
+2011-01-26 22:39 +0000 [r304342] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_fax.c: Add ability to disable T.38 usage for specific
+ SendFAX/ReceiveFAX sessions. Sometimes during troubleshooting it
+ can be useful to disable T.38 usage in order to narrow down a
+ problem. This patch adds an 'n' option to SendFAX and ReceiveFAX
+ so that can be done without having to disable T.38 usage entirely
+ for the peer that Asterisk is communicating with. (inspired by
+ trying to assist Bryant Zimmerman on asterisk-users)
+
+2011-01-26 22:27 +0000 [r304340] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 304339 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304339 | jpeeler | 2011-01-26 16:27:30 -0600
+ (Wed, 26 Jan 2011) | 9 lines Merged revisions 304338 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r304338 | jpeeler | 2011-01-26 16:26:37 -0600 (Wed, 26
+ Jan 2011) | 2 lines Change delimiter used internally for
+ GOTO_ON_BLINDXFR to commas to match 76703. ........
+ ................
+
+2011-01-26 21:03 +0000 [r304252] Mark Michelson <mmichelson@digium.com>
+
+ * main/udptl.c, /: Merged revisions 304250 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304250 | mmichelson | 2011-01-26 15:02:10 -0600
+ (Wed, 26 Jan 2011) | 9 lines Merged revisions 304242 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed,
+ 26 Jan 2011) | 3 lines Get rid of unused 'verbose' field in
+ ast_udptl ........ ................
+
+2011-01-26 20:44 +0000 [r304246] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/netsock2.c, /, channels/chan_sip.c,
+ channels/sip/reqresp_parser.c, include/asterisk/netsock2.h,
+ channels/sip/include/sip.h,
+ channels/sip/include/reqresp_parser.h: Merged revisions 304245
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304245 | mnicholson | 2011-01-26 14:43:27 -0600
+ (Wed, 26 Jan 2011) | 20 lines Merged revisions 304244 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304244 | mnicholson | 2011-01-26 14:42:16 -0600
+ (Wed, 26 Jan 2011) | 13 lines Merged revisions 304241 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan
+ 2011) | 6 lines This patch modifies chan_sip to route responses
+ to the address the request came from. It also modifies chan_sip
+ to respect the maddr parameter in the Via header. ABE-2664
+ Review: https://reviewboard.asterisk.org/r/1059/ ........
+ ................ ................
+
+2011-01-26 20:25 +0000 [r304195] Sean Bright <sean@malleable.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 304186 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304186 | seanbright | 2011-01-26 15:23:48 -0500
+ (Wed, 26 Jan 2011) | 16 lines Merged revisions 304181 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304181 | seanbright | 2011-01-26 15:22:47 -0500
+ (Wed, 26 Jan 2011) | 9 lines Merged revisions 304159 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r304159 | seanbright | 2011-01-26 15:18:29 -0500 (Wed,
+ 26 Jan 2011) | 1 line Make sure the sample queues.conf is
+ properly commented. ........ ................ ................
+
+2011-01-26 19:58 +0000 [r304152] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions
+ 303907 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r303907 | mnicholson | 2011-01-25 14:56:12 -0600 (Tue, 25 Jan
+ 2011) | 2 lines Reimplemented fax session reservation to reverse
+ the ABI breakage introduced in r297486. ........
+
+2011-01-26 19:40 +0000 [r304151] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 304150 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304150 | rmudgett | 2011-01-26 13:39:35 -0600
+ (Wed, 26 Jan 2011) | 16 lines Merged revisions 304149 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304149 | rmudgett | 2011-01-26 13:38:38 -0600
+ (Wed, 26 Jan 2011) | 9 lines Merged revisions 304148 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed,
+ 26 Jan 2011) | 2 lines Update documentation for
+ DAHDISendCallreroutingFacility() application. ..........
+ ................ ................
+
+2011-01-26 01:27 +0000 [r304098] Sean Bright <sean@malleable.com>
+
+ * /, main/file.c: Merged revisions 304097 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304097 | seanbright | 2011-01-25 20:26:26 -0500
+ (Tue, 25 Jan 2011) | 19 lines Merged revisions 304096 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan
+ 2011) | 12 lines Per the man page, setvbuf() must be called
+ before any other operation on an open file. We use setvbuf() to
+ associate a buffer with a stream, but we have already written to
+ the open file. This works (by chance) on Linux, but fails on
+ other platforms, such as OpenSolaris. (closes issue #16610)
+ Reported by: bklang Patches: setvbuf.patch uploaded by crjw
+ (license 963) Tested by: bklang, asgaroth, efutch ........
+ ................
+
+2011-01-25 23:31 +0000 [r304008] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 304007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r304007 | rmudgett | 2011-01-25 17:28:25 -0600
+ (Tue, 25 Jan 2011) | 22 lines Merged revisions 304006 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304006 | rmudgett | 2011-01-25 17:25:32 -0600
+ (Tue, 25 Jan 2011) | 15 lines Merged revisions 304005 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011)
+ | 8 lines DTMF attended transfers sometimes fail for no apparent
+ reason. The loop in feature_request_and_dial() can exit when
+ Party C has answered without processing an AST_CONTROL_ANSWER.
+ Also sometimes an AST_CONTROL_ANSWER never happens even though
+ Party C has answered. Don't hangup Party C if he is up or we
+ receive an AST_CONTROL_ANSWER. ........ ................
+ ................
+
+2011-01-25 22:15 +0000 [r303963] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 303962 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r303962 | twilson | 2011-01-25 16:09:01 -0600
+ (Tue, 25 Jan 2011) | 30 lines Merged revisions 303960 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303960 | twilson | 2011-01-25 16:02:42 -0600
+ (Tue, 25 Jan 2011) | 23 lines Merged revisions 303906 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011)
+ | 16 lines Guard against retransmitting BYEs indefinitely In the
+ case of an attended transfer (A calls B, A atxfers to C) where A
+ becomes unreachable before replying to Asterisk's BYE, Asterisk
+ can sometimes retransmit the BYE indefinitely. This is because
+ __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0],
+ SIP_ALREADYGONE and will then transmit a BYE. When this BYE times
+ out, it will not ever be marked as ALREADYGONE, so when
+ __sip_autodestruct is called again, we end up starting the cycle
+ over. This patch adds a call to sip_alreadygone(pkt->owner) in
+ retrans_pkt in the case of a BYE that has timed out. This should
+ prevent Asterisk from trying to transmit new BYE messages in the
+ future. Review: https://reviewboard.asterisk.org/r/1077/ ........
+ ................ ................
+
+2011-01-25 18:56 +0000 [r303861] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, channels/chan_sip.c: Merged revisions 303860 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r303860 | tilghman | 2011-01-25 12:55:27 -0600
+ (Tue, 25 Jan 2011) | 12 lines Merged revisions 303858 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r303858 | tilghman | 2011-01-25 12:41:26 -0600 (Tue, 25 Jan 2011)
+ | 5 lines Fix "sip show user <tab>", so that it actually shows
+ results, instead of just completing the last entry. (closes issue
+ #16675) Reported by: pj ........ ................
+
+2011-01-25 17:58 +0000 [r303772] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_ss7.h, /,
+ channels/sig_pri.c, channels/sig_ss7.c: Merged revisions 303771
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600
+ (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600
+ (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011)
+ | 40 lines Sending out unnecessary PROCEEDING messages breaks
+ overlap dialing. Issue #16789 was a good idea. Unfortunately, it
+ breaks overlap dialing through Asterisk. There is not enough
+ information available at this point to know if dialing is
+ complete. The ast_exists_extension(), ast_matchmore_extension(),
+ and ast_canmatch_extension() calls are not adequate to detect a
+ dial through extension pattern of "_9!". Workaround is to use the
+ dialplan Proceeding() application early in non-dial through
+ extensions. * Effectively revert issue #16789. * Allow outgoing
+ overlap dialing to hear dialtone and other early media. A
+ PROGRESS "inband-information is now available" message is now
+ sent after the SETUP_ACKNOWLEDGE message for non-digital calls.
+ An AST_CONTROL_PROGRESS is now generated for incoming
+ SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of
+ the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent
+ with the cause codes. * Added better protection from sending out
+ of sequence messages by combining several flags into a single
+ enum value representing call progress level. * Added diagnostic
+ messages for deferred overlap digits handling corner cases.
+ (closes issue #17085) Reported by: shawkris (closes issue #18509)
+ Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch
+ uploaded by rmudgett (license 664) Expanded upon
+ issue18509_early_media_v1.8_v3.patch to include analog and SS7
+ because of backporting requirements. Tested by: wimpy, rmudgett
+ ........ ................ ................
+
+2011-01-25 17:05 +0000 [r303679] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 303678 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r303678 | jpeeler | 2011-01-25 11:02:38 -0600
+ (Tue, 25 Jan 2011) | 33 lines Merged revisions 303677 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303677 | jpeeler | 2011-01-25 10:59:28 -0600
+ (Tue, 25 Jan 2011) | 26 lines Merged revisions 303676 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011)
+ | 20 lines Fix voicemail sequencing for file based storage. A
+ previous change was made to account for when the number of
+ voicemail messages exceeds the max limit to be handled properly,
+ but it caused gaps in the messages to not be properly handled.
+ This has now been resolved. In later non 1.4 branches, it appears
+ that resequencing wasn't even occurring due from what appears and
+ accidental code removal. (closes issue #18498) Reported by:
+ JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license
+ 325) (closes issue #18486) Reported by: bluefox Patches:
+ bug18486.patch uploaded by jpeeler (license 325) ........
+ ................ ................
+
+2011-01-25 15:52 +0000 [r303638] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/utils.c: Use unsigned char in comparison for UTF8 check to
+ quiet a compiler warning.
+
+2011-01-24 20:57 +0000 [r303547-303551] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, main/pbx.c, /, apps/app_meetme.c,
+ main/features.c, include/asterisk/channel.h: Merged revisions
+ 303549 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r303549 | russell | 2011-01-24 14:51:37 -0600
+ (Mon, 24 Jan 2011) | 45 lines Merged revisions 303548 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303548 | russell | 2011-01-24 14:49:53 -0600
+ (Mon, 24 Jan 2011) | 38 lines Merged revisions 303546 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011)
+ | 31 lines Fix channel redirect out of MeetMe() and other issues
+ with channel softhangup. Mantis issue #18585 reports that a
+ channel redirect out of MeetMe() stopped working properly. This
+ issue includes a patch that resolves the issue by removing a call
+ to ast_check_hangup() from app_meetme.c. I left that in my patch,
+ as it doesn't need to be there. However, the rest of the patch
+ fixes this problem with or without the change to app_meetme. The
+ key difference between what happens before and after this patch
+ is the effect of the END_OF_Q control frame. After END_OF_Q is
+ hit in ast_read(), ast_read() will return NULL. With the
+ ast_check_hangup() removed, app_meetme sees this which causes it
+ to exit as intended. Checking ast_check_hangup() caused
+ app_meetme to exit earlier in the process, and the target of the
+ redirect saw the condition where ast_read() returned NULL.
+ Removing ast_check_hangup() works around the issue in app_meetme,
+ but doesn't solve the issue if another application did the same
+ thing. There are also other edge cases where if an application
+ finishes at the same time that a redirect happens, the target of
+ the redirect will think that the channel hung up. So, I made some
+ changes in pbx.c to resolve it at a deeper level. There are
+ already places that unset the SOFTHANGUP_ASYNCGOTO flag in an
+ attempt to abort the hangup process. My patch extends this to
+ remove the END_OF_Q frame from the channel's read queue, making
+ the "abort hangup" more complete. This same technique was used in
+ every place where a softhangup flag was cleared. (closes issue
+ #18585) Reported by: oej Tested by: oej, wedhorn, russell Review:
+ https://reviewboard.asterisk.org/r/1082/ ........
+ ................ ................
+
+ * contrib/scripts/install_prereq: Add gsm-devel as a package to
+ install on redhat based systems.
+
+2011-01-24 18:59 +0000 [r303509] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_config_curl.c, include/asterisk/utils.h,
+ funcs/func_curl.c, channels/chan_sip.c, tests/test_utils.c,
+ res/res_agi.c, channels/sip/reqresp_parser.c, main/http.c,
+ main/utils.c, funcs/func_uri.c: According to section 19.1.2 of
+ RFC 3261: For each component, the set of valid BNF expansions
+ defines exactly which characters may appear unescaped. All other
+ characters MUST be escaped. This patch modifies ast_uri_encode()
+ to encode strings in line with this recommendation. This patch
+ also adds an ast_escape_quoted() function which escapes '"' and
+ '\' characters in quoted strings in accordance with section 25.1
+ of RFC 3261. The ast_uri_encode() function has also been modified
+ to take an ast_flags struct describing the set of rules it should
+ use when escaping characters to allow for it to escape SIP URIs
+ in addition to HTTP URIs and other types of URIs or variations of
+ those two URI types in the future. The ast_uri_decode() function
+ has also been modified to accept an ast_flags struct describing
+ the set of rules to use when decoding to enable decoding '+' as '
+ ' in legacy http URLs. The unit tests for these functions have
+ also been updated. ABE-2705 Review:
+ https://reviewboard.asterisk.org/r/1081/
+
+2011-01-24 17:21 +0000 [r303468] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 303467 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r303467 | qwell | 2011-01-24 11:20:03 -0600
+ (Mon, 24 Jan 2011) | 22 lines Merged revisions 303285 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303285 | qwell | 2011-01-21 15:48:09 -0600
+ (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) |
+ 8 lines Reset configuration before parsing users.conf. Some
+ values configured in chan_dahdi.conf were able to leak in to
+ users.conf configuration. This was surprising users, and
+ potentially setting non-sane "defaults". ASTNOW-125 ........
+ ................ ................
+
+2011-01-22 04:13 +0000 [r303418] Russell Bryant <russell@digium.com>
+
+ * configure, configure.ac: Revert default compiler change. If
+ someone wishes to do so, it is trivial to set your own default
+ when running the configure script.
+
+2011-01-21 23:11 +0000 [r303288-303376] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c, /: Temporarily revert r303288
+
+ * channels/chan_dahdi.c, /: Merged revisions 303286 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r303286 | qwell | 2011-01-21 15:50:11 -0600
+ (Fri, 21 Jan 2011) | 22 lines Merged revisions 303285 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303285 | qwell | 2011-01-21 15:48:09 -0600
+ (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) |
+ 8 lines Reset configuration before parsing users.conf. Some
+ values configured in chan_dahdi.conf were able to leak in to
+ users.conf configuration. This was surprising users, and
+ potentially setting non-sane "defaults". ASTNOW-125 ........
+ ................ ................
+
+2011-01-21 09:09 +0000 [r303198-303235] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * configure, configure.ac: Really use llvm-gcc, when available.
+
+ * funcs/func_db.c, CHANGES: Add DB_KEYS. Discussion on #asterisk on
+ 2011-01-19: (02:07:03 PM) boch: i wonder how to cycle all entries
+ in a tree (02:07:11 PM) leifmadsen: use While() (02:07:17 PM)
+ leifmadsen: you need to know the tree structure already though
+ (02:07:36 PM) boch: what you mean? (02:09:02 PM) leifmadsen: you
+ need to know the structure prior to looping, because you can't
+ just return the structure from the dialplan (02:09:43 PM)
+ leifmadsen: the only way I can think of doing that is via
+ something like writing the output of: asterisk -rx "database
+ show" to a file, then looping through that to know the structure
+ of the database and check everything (02:09:59 PM) leifmadsen:
+ but at that point you're better off just using either a
+ relational database or an external script (02:10:13 PM) boch: for
+ example i need to know all entries in the tree (02:10:15 PM)
+ boch: got it (02:10:20 PM) leifmadsen: exactly (02:10:22 PM)
+ leifmadsen: that's the problem (02:10:22 PM) boch: thank you
+ (02:13:09 PM) mateu: yeah, i'm surprised there isn't something
+ from the dialplan like 'database show family' so one can get all
+ keys in a family to loop over. (02:15:35 PM) leifmadsen: database
+ shows everything (02:16:22 PM) mateu: i mean something from the
+ dial plan that mimics 'database show <family>' (02:16:41 PM)
+ leifmadsen: guess no one has found that important enough to
+ program :) (02:16:52 PM) leifmadsen: at that point you should
+ probably just use a relational database... (02:17:10 PM) mateu: i
+ dunno (02:17:16 PM) mateu: seems pretty basic to me. (02:17:16
+ PM) leifmadsen: me either (02:17:19 PM) leifmadsen: sure does
+ (02:17:24 PM) leifmadsen: no one has programmed it though
+ (02:17:28 PM) ***leifmadsen shrugs (02:17:43 PM) mateu: ok, well
+ at least we know how it currently stands. thanks leifmadsen
+ (02:28:52 PM) Corydon76-home: leifmadsen: something like
+ HASHKEYS() ? (02:30:11 PM) leifmadsen: Corydon76-home: ummm, I
+ was thinking more like DUNDI_QUERY() and DUNDI_RESULT() (02:30:31
+ PM) leifmadsen: although HASHKEYS() might work (02:30:58 PM)
+ leifmadsen: actually ya, looking at it, similar to HASHKEYS()
+ (02:31:01 PM) leifmadsen: DBKEYS() I guess? (02:31:45 PM)
+ Corydon76-home: So with no argument, retrieves families, with an
+ argument, retrieves keys of that family? (02:34:02 PM)
+ leifmadsen: ya (02:34:16 PM) leifmadsen: how would you iterate
+ through layers of them? (02:34:30 PM) leifmadsen: i.e.
+ family/key/key/key ? (02:34:43 PM) Corydon76-home: Essentially,
+ yes
+
+2011-01-20 20:35 +0000 [r303154] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/ccss.c: Merged revisions 303153 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r303153 | rmudgett | 2011-01-20 14:31:20 -0600
+ (Thu, 20 Jan 2011) | 22 lines Merged revision 303098 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu,
+ 20 Jan 2011) | 15 lines CC_INTERFACES does not get built
+ correctly with local channels. If local channels are used with
+ CCSS, CC_INTERFACES gets garbage prepended to it so the CC recall
+ fails. Also CC_INTERFACES gets "&(null)" appended to it. *
+ Initialize the buffer to eliminate the prepended garbage. *
+ Filter out the empty interface strings to eliminate the latter. *
+ Added a diagnostic message if the CC_INTERFACES is ever empty.
+ JIRA ABE-2740 JIRA SWP-2848 .......... ................
+
+2011-01-20 19:58 +0000 [r303108] Shaun Ruffell <sruffell@digium.com>
+
+ * /, main/features.c: Merged revisions 303107 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r303107 | sruffell | 2011-01-20 13:57:31 -0600
+ (Thu, 20 Jan 2011) | 23 lines Merged revisions 303106 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r303106 | sruffell | 2011-01-20 13:56:34 -0600 (Thu, 20 Jan 2011)
+ | 15 lines main/features: Use POLLPRI when waiting for events on
+ parked channels. This change resolves a regression in the 1.6.2
+ when converting from select to poll. The DAHDI timers use POLLPRI
+ to indicate that the timer fired, but features was not waiting
+ for that flag. The result was no audio for MOH when a call was
+ parked and res_timing_dahdi was in use. This patch is slightly
+ modified from the one on the mantis issue. It does not set an
+ exception on the channel if the POLLPRI flag is set. (closes
+ issue #18262) Reported by: francesco_r Patches:
+ patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
+ Tested by: francesco_r, rfrantik, one47 ........ ................
+
+2011-01-20 17:14 +0000 [r303011] Jeff Peeler <jpeeler@digium.com>
+
+ * /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions
+ 303009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r303009 | jpeeler | 2011-01-20 11:10:32 -0600
+ (Thu, 20 Jan 2011) | 21 lines Merged revisions 303008 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303008 | jpeeler | 2011-01-20 11:07:44 -0600
+ (Thu, 20 Jan 2011) | 14 lines Merged revisions 303007 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011)
+ | 8 lines Add new queue strategy to preserve behavior for when
+ queue members moved to ao2. Add queue strategy called "rrordered"
+ to mimic old behavior from when queue members were stored in a
+ linked list. ABE-2707 ........ ................ ................
+
+2011-01-20 16:12 +0000 [r302922] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_privacy.c: Merged revisions 302921 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302921 | russell | 2011-01-20 10:12:15 -0600
+ (Thu, 20 Jan 2011) | 9 lines Merged revisions 302920 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r302920 | russell | 2011-01-20 10:11:58 -0600 (Thu, 20
+ Jan 2011) | 2 lines Resolve a compiler warning. ........
+ ................
+
+2011-01-20 15:46 +0000 [r302919] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 302918 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302918 | lmadsen | 2011-01-20 09:45:39 -0600
+ (Thu, 20 Jan 2011) | 16 lines Merged revisions 302917 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011)
+ | 8 lines Option L() is milliseconds, not seconds. > Change the
+ verbose output of option L() to say milliseconds and not seconds
+ > as the value is in milliseconds. > > (closes issue #18264) >
+ Reported by: jacco > Patches: > app_dial_patch.txt uploaded by
+ lmadsen (license 10) ........ ................
+
+2011-01-20 09:07 +0000 [r302879] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * configure, configure.ac: On systems which have LLVM, use that
+ compiler. Should result in a massive speed increase.
+
+2011-01-19 23:57 +0000 [r302838] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /: Merged revisions 302837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r302837 | russell | 2011-01-19 17:56:48 -0600 (Wed, 19 Jan 2011)
+ | 2 lines Only check container count if it exists. ........
+
+2011-01-19 23:53 +0000 [r302835-302836] Sean Bright <sean@malleable.com>
+
+ * main/config.c: Clarify a source comment about configuration
+ template categories. (closes issue #18578) Reported by: astmiv
+ Patches: asterisk.main.config.2.patch uploaded by astmiv (license
+ 1189)
+
+ * /, apps/app_voicemail.c: Merged revisions 302834 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302834 | seanbright | 2011-01-19 18:49:00 -0500
+ (Wed, 19 Jan 2011) | 14 lines Merged revisions 302833 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302833 | seanbright | 2011-01-19 18:47:22 -0500 (Wed, 19 Jan
+ 2011) | 7 lines Support greetingsfolder as documented in
+ voicemail.conf.sample. (closes issue #17870) Reported by:
+ edhorton Patches:
+ __20100816-app_voicemail-greetingsfolder-support.txt uploaded by
+ lmadsen (license 10) ........ ................
+
+2011-01-19 23:33 +0000 [r302832] Paul Belanger <pabelanger@digium.com>
+
+ * /, contrib/scripts/install_prereq: Merged revisions 302831 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r302831 | pabelanger | 2011-01-19 18:29:45 -0500 (Wed, 19 Jan
+ 2011) | 2 lines Add binutils-dev for BETTER_BACKTRACES ........
+
+2011-01-19 23:07 +0000 [r302786-302790] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /: Merged revisions 302789 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302789 | russell | 2011-01-19 17:06:46 -0600
+ (Wed, 19 Jan 2011) | 11 lines Merged revisions 302788 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302788 | russell | 2011-01-19 17:06:14 -0600 (Wed, 19 Jan 2011)
+ | 4 lines Turn a noisy verbose message into a debug message. This
+ can drown your console if you're using the AMI over HTTP.
+ ........ ................
+
+ * main/manager.c, /: Merged revisions 302785 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r302785 | russell | 2011-01-19 16:35:15 -0600 (Wed, 19 Jan 2011)
+ | 15 lines Resolve a memory leak with the manager interface is
+ disabled. The intent of this check as it stands in previous
+ versions of Asterisk was to check if there are any active
+ sessions. If there were no sessions, then the function would
+ return immediately and not bother with queueing up the manager
+ event to be processed. Since the conversion of storing sessions
+ in an astobj2 container, this check will always pass. I changed
+ it to go back to checking what was intended. The side effect of
+ this was that if the AMI is disabled, the manager event queue is
+ populated anyway, but the code that runs to clear out the queue
+ never runs. A producer with no consumer is a bad thing. Reported
+ internally by kmorgan. ........
+
+2011-01-19 21:35 +0000 [r302732] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 302713 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302713 | rmudgett | 2011-01-19 15:29:22 -0600
+ (Wed, 19 Jan 2011) | 29 lines Merged revisions 302693 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r302693 | rmudgett | 2011-01-19 15:25:41 -0600
+ (Wed, 19 Jan 2011) | 22 lines Merged revisions 302671 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011)
+ | 15 lines DTMF transfer plays the wrong sounds for wrong number
+ or other call failure. * Set the default for features.conf.sample
+ xferfailsound option to "beeperr" as documented instead of
+ "pbx-invalid" and corrected the use of it in DTMF blind transfer
+ (#1). * Improved DTMF blind transfer handling of wrong numbers.
+ Most of the concerns in this issue were taken care of by the
+ patch for issue 17999: Issues with DTMF triggered attended
+ transfers. (closes issue #18379) Reported by: gincantalupo Tested
+ by: rmudgett ........ ................ ................
+
+2011-01-19 21:24 +0000 [r302644-302686] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, include/asterisk/astdb.h: Merged revisions 302680 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302680 | tilghman | 2011-01-19 15:23:31 -0600
+ (Wed, 19 Jan 2011) | 16 lines Merged revisions 302675 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r302675 | tilghman | 2011-01-19 15:22:45 -0600
+ (Wed, 19 Jan 2011) | 9 lines Merged revisions 302663 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19
+ Jan 2011) | 2 lines Add some API documentation ........
+ ................ ................
+
+ * /, main/app.c: Merged revisions 302634 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302634 | tilghman | 2011-01-19 14:24:57 -0600
+ (Wed, 19 Jan 2011) | 22 lines Merged revisions 302599 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011)
+ | 15 lines Kill zombies. When we ast_safe_fork() with a non-zero
+ argument, we're expected to reap our own zombies. On a zero
+ argument, however, the zombies are only reaped when there aren't
+ any non-zero forked children alive. At other times, we accumulate
+ zombies. This code is forward ported from res_agi in 1.4, so that
+ forked children are always reaped, thus preventing an
+ accumulation of zombie processes. (closes issue #18515) Reported
+ by: ernied Patches: 20101221__issue18515.diff.txt uploaded by
+ tilghman (license 14) Tested by: ernied ........ ................
+
+2011-01-19 20:15 +0000 [r302601] Jason Parker <jparker@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 302600 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r302600 | qwell | 2011-01-19 14:14:40 -0600 (Wed, 19 Jan 2011) |
+ 1 line Fix typo pointed out on asterisk-users list. ........
+
+2011-01-19 19:04 +0000 [r302507-302556] Sean Bright <sean@malleable.com>
+
+ * /, main/utils.c: Merged revisions 302555 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302555 | seanbright | 2011-01-19 14:03:32 -0500
+ (Wed, 19 Jan 2011) | 14 lines Merged revisions 302554 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302554 | seanbright | 2011-01-19 14:02:29 -0500 (Wed, 19 Jan
+ 2011) | 7 lines Don't call strlen() when we only need to look at
+ the next character or two. (closes issue #18042) Reported by:
+ wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded
+ by wdoekes (license 717) ........ ................
+
+ * /, main/features.c: Merged revisions 302552 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302552 | seanbright | 2011-01-19 13:54:47 -0500
+ (Wed, 19 Jan 2011) | 14 lines Merged revisions 302551 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302551 | seanbright | 2011-01-19 13:54:03 -0500 (Wed, 19 Jan
+ 2011) | 7 lines Remove an extraneous \r\n at the end of a parking
+ manager events. (closes issue #18363) Reported by:
+ clegall_proformatique Patches:
+ asterisk_1.8_295998_parking_manager_events_format.patch uploaded
+ by clegall proformatique (license 1139) ........ ................
+
+ * /, res/res_agi.c: Merged revisions 302549 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302549 | seanbright | 2011-01-19 13:43:11 -0500
+ (Wed, 19 Jan 2011) | 17 lines Merged revisions 302548 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan
+ 2011) | 10 lines Properly handle partial reads from fgets() when
+ handling AGIs. When fgets() failed with EAGAIN, we were
+ continually decrementing the available space left in our buffer,
+ resulting in botched command handling. (closes issue #16032)
+ Reported by: notahat Patches: agi_buffer_patch2.diff uploaded by
+ fnordian (license 110) ........ ................
+
+ * /, main/utils.c: Merged revisions 302505 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302505 | seanbright | 2011-01-19 12:58:11 -0500
+ (Wed, 19 Jan 2011) | 14 lines Merged revisions 302504 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302504 | seanbright | 2011-01-19 12:56:32 -0500 (Wed, 19 Jan
+ 2011) | 7 lines Make sure that h_length is set when we
+ short-circuit out of ast_gethostbyname. (closes issue #16135)
+ Reported by: thedavidfactor Patches: utils.patch uploaded by
+ thedavidfactor (license 903) ........ ................
+
+2011-01-19 17:15 +0000 [r302463] Paul Belanger <pabelanger@digium.com>
+
+ * /, res/res_timing_timerfd.c: Merged revisions 302462 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302462 | pabelanger | 2011-01-19 12:09:35 -0500
+ (Wed, 19 Jan 2011) | 9 lines Merged revisions 302461 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r302461 | pabelanger | 2011-01-19 12:08:01 -0500 (Wed,
+ 19 Jan 2011) | 2 lines Handle 'Resource temporarily unavailable'
+ error more gracefully. ........ ................
+
+2011-01-19 15:54 +0000 [r302413-302418] Sean Bright <sean@malleable.com>
+
+ * /, configs/extensions.conf.sample: Merged revisions 302417 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302417 | seanbright | 2011-01-19 10:53:20 -0500
+ (Wed, 19 Jan 2011) | 16 lines Merged revisions 302416 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302416 | seanbright | 2011-01-19 10:52:44 -0500 (Wed, 19 Jan
+ 2011) | 9 lines Remove references to priorityjumping from the
+ sample extensions.conf. Priority jumping was removed from
+ pbx_config in r68970. (closes issue #18622) Reported by: kshumard
+ Patches: extensions.conf.sample.patch uploaded by kshumard
+ (license 92) ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 302414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r302414 | seanbright | 2011-01-19 10:45:17 -0500 (Wed, 19 Jan
+ 2011) | 7 lines Initialize an uninitialized variable. (closes
+ issue #18640) Reported by: jcovert Patches: chan_sip.c.patch
+ uploaded by jcovert (license 551) ........
+
+ * channels/chan_local.c, /: Merged revisions 302412 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r302412 | seanbright | 2011-01-19 10:31:39 -0500 (Wed,
+ 19 Jan 2011) | 10 lines Use appropriate type for requested format
+ in chan_local. We were passing and storing the requested format
+ as an int instead of format_t resulting in truncation. (closes
+ issue #18238) Reported by: whizemen Patches:
+ 0018238_speex16.patch uploaded by whizemen (license 1143)
+ ........
+
+2011-01-18 22:06 +0000 [r302319] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 302318 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r302318 | rmudgett | 2011-01-18 16:04:14 -0600 (Tue, 18 Jan 2011)
+ | 1 line Use the expanded format type instead of plain int.
+ ........
+
+2011-01-18 21:44 +0000 [r302315] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 302314 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302314 | mnicholson | 2011-01-18 15:43:21 -0600
+ (Tue, 18 Jan 2011) | 18 lines Merged revisions 302313 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r302313 | mnicholson | 2011-01-18 15:40:03 -0600
+ (Tue, 18 Jan 2011) | 11 lines Merged revisions 302311 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan
+ 2011) | 4 lines URI encode the user part of the contact header.
+ ABE-2705 ........ ................ ................
+
+2011-01-18 20:40 +0000 [r302270] Jeff Peeler <jpeeler@digium.com>
+
+ * main/pbx.c, /: Merged revisions 302266 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302266 | jpeeler | 2011-01-18 14:19:57 -0600
+ (Tue, 18 Jan 2011) | 34 lines Merged revisions 302265 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011)
+ | 27 lines Convert device state callbacks to ao2 objects to fix a
+ deadlock in chan_sip. Lock scenario presented here: Thread 1
+ holds ast_rdlock_contexts &conlock holds handle_statechange hints
+ holds handle_statechange hint waiting for cb_extensionstate
+ Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds
+ handle_request_do &netlock holds find_call sip_pvt_ptr waiting
+ for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911
+ (ast_rdlock_contexts) Chan_sip has an established locking order
+ of locking the sip_pvt and then getting the context lock. So the
+ as stated by the summary, the operations in thread 2 have been
+ modified to no longer require the context lock. (closes issue
+ #18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch
+ uploaded by one47 (license 23), modified by me Review:
+ https://reviewboard.asterisk.org/r/1072/ ........
+ ................
+
+2011-01-18 20:21 +0000 [r302268] Russell Bryant <russell@digium.com>
+
+ * /, main/astobj2.c: Merged revisions 302267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r302267 | russell | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011)
+ | 5 lines Don't enable AO2_DEBUG by default if AST_DEVMODE is on.
+ AO2_DEBUG is not important and is causing a false compiler
+ warning to be generated on my Ubuntu Natty dev box. ........
+
+2011-01-18 18:17 +0000 [r302178] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 302174 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r302174 | rmudgett | 2011-01-18 12:11:43 -0600
+ (Tue, 18 Jan 2011) | 102 lines Merged revisions 302173 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600
+ (Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011)
+ | 88 lines Issues with DTMF triggered attended transfers. Issue
+ #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in
+ features.conf for attended transfer). 3) A hears MOH. B dial
+ number C 4) C ringing. A hears MOH. 5) B hangup. A still hears
+ MOH. C ringing. 6) A hangup. C still ringing until
+ "atxfernoanswertimeout" expires. For v1.4 C will ring forever
+ until C answers the dead line. (Issue #17096) Problem: When A and
+ B hangup, C is still ringing. Issue #18395 SIP call limit of B is
+ 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C
+ ringing 4. Timeout waiting for C to answer 5. Recall to B fails
+ because B has reached its call limit. Because B reached its call
+ limit, it cannot do anything until the transfer it started
+ completes. Issue #17273 Same scenario as issue 18395 but party B
+ is an FXS port. Party B cannot do anything until the transfer it
+ started completes. If B goes back off hook before C answers, B
+ hears ringback instead of the expected dialtone. ********** Note
+ for the issue #17273 and #18395 fix: DTMF attended transfer works
+ within the channel bridge. Unfortunately, when either party A or
+ B in the channel bridge hangs up, that channel is not completely
+ hung up until the transfer completes. This is a real problem
+ depending upon the channel technology involved. For chan_dahdi,
+ the channel is crippled until the hangup is complete. Either the
+ channel is not useable (analog) or the protocol disconnect
+ messages are held up (PRI/BRI/SS7) and the media is not released.
+ For chan_sip, a call limit of one is going to block that endpoint
+ from any further calls until the hangup is complete. For party A
+ this is a minor problem. The party A channel will only be in this
+ condition while party B is dialing and when party B and C are
+ conferring. The conversation between party B and C is expected to
+ be a short one. Party B is either asking a question of party C or
+ announcing party A. Also party A does not have much incentive to
+ hangup at this point. For party B this can be a major problem
+ during a blonde transfer. (A blonde transfer is our term for an
+ attended transfer that is converted into a blind transfer. :))
+ Party B could be the operator. When party B hangs up, he assumes
+ that he is out of the original call entirely. The party B channel
+ will be in this condition while party C is ringing, while
+ attempting to recall party B, and while waiting between call
+ attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to
+ fix the problem. It will replace the party B channel technology
+ with a NULL channel driver to complete hanging up the party B
+ channel technology. The consequences of this code is that the 'h'
+ extension will not be able to access any channel technology
+ specific information like SIP statistics for the call.
+ ATXFER_NULL_TECH is not defined by default. ********** (closes
+ issue #17999) Reported by: iskatel Tested by: rmudgett JIRA
+ SWP-2246 (closes issue #17096) Reported by: gelo Tested by:
+ rmudgett JIRA SWP-1192 (closes issue #18395) Reported by:
+ shihchuan Tested by: rmudgett (closes issue #17273) Reported by:
+ grecco Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1047/ ........
+ ................ ................
+
+2011-01-17 16:38 +0000 [r302006-302048] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 293493 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010)
+ | 14 lines Only offer codecs both sides support for directmedia
+ When using directmedia, Asterisk needs to limit the codecs
+ offered to just the ones that both sides recognize, otherwise
+ they may end up sending audio that the other side doesn't
+ understand. (closes issue #17403) Reported by: one47 Patches:
+ sip_codecs_simplified4 uploaded by one47 (license 23) Tested by:
+ one47, falves11 Review: https://reviewboard.asterisk.org/r/967/
+ ........
+
+ * /, configs/sip.conf.sample: Merged revisions 302005 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r302005 | twilson | 2011-01-17 09:04:59 -0600 (Mon, 17
+ Jan 2011) | 2 lines Document "encryption" option in
+ sip.conf.sample ........
+
+2011-01-14 21:13 +0000 [r301947] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 301946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r301946 | rmudgett | 2011-01-14 15:09:57 -0600 (Fri, 14 Jan 2011)
+ | 13 lines Deadlock between dahdi_request() and pri_dchannel()
+ processing an incomming call. The sig_pri_new_ast_channel() is
+ called with the channel private lock held when pri_dchannel()
+ calls it and no channel private lock held when dahdi_request()
+ calls it. The use of pri_grab() in sig_pri_new_ast_channel()
+ could leave the channel private lock held when it returns if the
+ lock was not held before calling it. Make
+ sig_pri_new_ast_channel() just lock the PRI span lock instead of
+ using pri_grab(). It is safe to do this because dahdi_request()
+ does not have the channel private lock and the deadlock potential
+ with the PRI span lock is only between pri_dchannel() and other
+ threads. ........
+
+2011-01-14 20:18 +0000 [r301858] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_multicast_rtp.c, /: Merged revisions 301851 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r301851 | bbryant | 2011-01-14 15:11:55 -0500 (Fri, 14 Jan 2011)
+ | 6 lines Changing previous revisions 301845/301847 to use
+ ast_sockaddr_setnull() instead of setting the field manually to
+ avoid uninitialized data. Review:
+ https://reviewboard.asterisk.org/r/1076/ ........
+
+2011-01-14 20:07 +0000 [r301850] Andrew Latham <lathama@gmail.com>
+
+ * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to
+ function documentation. Fix amatuer type mistake
+
+2011-01-14 19:44 +0000 [r301847] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_multicast_rtp.c, /: Merged revisions 301845 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r301845 | bbryant | 2011-01-14 14:35:23 -0500 (Fri, 14 Jan 2011)
+ | 9 lines Fix for a consistent MulticastRTP channel driver crash
+ due to use of unitilized data. (closes issue #18290) (closes
+ issue #18602) Reported by: voipgate, wybecom Review:
+ https://reviewboard.asterisk.org/r/1076/ ........
+
+2011-01-14 19:39 +0000 [r301846] Andrew Latham <lathama@gmail.com>
+
+ * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to
+ function documentation.
+
+2011-01-14 17:34 +0000 [r301791] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 301790 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r301790 | jpeeler | 2011-01-14 11:32:52 -0600 (Fri, 14 Jan 2011)
+ | 42 lines Resolve deadlock involving REFER. Two fixes: 1) One
+ must always have the private unlocked before calling
+ pbx_builtin_setvar_helper to not invalidate locking order since
+ it locks the channel. 2) Unlock the channel before calling
+ pbx_find_extension, which starts and stops autoservice during the
+ lookup. The problem scenario as illustrated by the reporter:
+ Thread: do_monitor ----------------------- handle_request_do
+ handle_incoming handle_request_refer ast_parking_ext_valid
+ pbx_find_extension ast_autoservice_stop while (chan_list_state ==
+ as_chan_list_state) { usleep(1000); } Thread: autoservice_run
+ ----------------------- autoservice_run chan = ast_waitfor_n
+ ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple /
+ complex (depending on your system) ast_channel_lock(c[x]);
+ handle_request_do and schedule_process_request_queue locks the
+ owner if it exists. The autoservice thread is waiting for the
+ channel lock, which wasn't ever released since the do_monitor
+ thread was waiting for autoservice operations to complete. Solved
+ by unlocking the channel but keeping a reference to guarantee
+ safety. (closes issue #18403) Reported by: jthurman Patches:
+ 20110103-blind_deadlock.diff uploaded by jthurman (license 614)
+ issue18403.patch uploaded by jpeeler (license 325) Tested by:
+ jthurman ........
+
+2011-01-13 17:02 +0000 [r301732] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/phoneprov.conf.sample: Merged revisions 301731 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r301731 | lmadsen | 2011-01-13 11:01:43 -0600
+ (Thu, 13 Jan 2011) | 15 lines Merged revisions 301730 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011)
+ | 7 lines Add static entry for split Polycom 332 firmware.
+ (closes issue #18607) Reported by: cjacobsen Patches:
+ polycom_331.diff uploaded by cjacobsen (license 1029) Tested by:
+ lathama ........ ................
+
+2011-01-13 16:27 +0000 [r301729] Paul Belanger <pabelanger@digium.com>
+
+ * main/pbx.c, CHANGES: Add dialplan variables for asterisk.conf
+ directories Review: https://reviewboard.asterisk.org/r/1075/
+
+2011-01-12 21:24 +0000 [r301684] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 301683 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r301683 | twilson | 2011-01-12 15:19:48 -0600
+ (Wed, 12 Jan 2011) | 15 lines Merged revisions 301682 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011)
+ | 9 lines Don't reject all SUBSCRIBE auth requests When merging
+ another SUBSCRIBE fix from 1.4, some braces were put in the wrong
+ place. This patch fixes that. (closes issue #18597) Reported by:
+ thsgmbh ........ ................
+
+2011-01-12 18:52 +0000 [r301596] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /: Merged revisions 301595 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r301595 | mnicholson | 2011-01-12 12:51:37 -0600
+ (Wed, 12 Jan 2011) | 22 lines Merged revisions 301594 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r301594 | mnicholson | 2011-01-12 12:50:31 -0600
+ (Wed, 12 Jan 2011) | 15 lines Removed a usleep(1) that shouldn't
+ be necessary in session_do, and removed the ms_t member from the
+ mansession_session structure. Merged revisions 301591 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan
+ 2011) | 5 lines Don't store the thread id for the manager session
+ in the structure we pass to the thread for the manager session.
+ ABE-2543 ........ ................ ................
+
+2011-01-12 18:12 +0000 [r301505] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 301504 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r301504 | jpeeler | 2011-01-12 12:12:08 -0600
+ (Wed, 12 Jan 2011) | 26 lines Merged revisions 301503 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600
+ (Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011)
+ | 12 lines Fix CPU spike when pressing DTMF after agent login.
+ The problem here is that DTMF was being continuously deferred and
+ requeued since ast_safe_sleep is called in a loop. There are
+ serveral other places in the code that sleeps and then loops in a
+ similar fashion. Because of this fact I opted to not defer DTMF
+ any more, which will not affect the original fix:
+ https://reviewboard.asterisk.org/r/674 (closes issue #18130)
+ Reported by: rgj ........ ................ ................
+
+2011-01-12 16:05 +0000 [r301447] David Vossel <dvossel@digium.com>
+
+ * /, main/file.c: Merged revisions 301446 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r301446 | dvossel | 2011-01-12 10:05:12 -0600 (Wed, 12 Jan 2011)
+ | 2 lines Removal of unused variables so Asterisk will compile.
+ ........
+
+2011-01-12 15:59 +0000 [r301445] Stefan Schmidt <sst@sil.at>
+
+ * Makefile: fix wrong text of rerun menuselect after user interface
+ warning the warning, if no user interface for menuselect warning
+ was found is not right. you have to rerun configure before make
+ menuselect after installing a proper user interface. (closes
+ issue 0018594) Reported by: Dovid
+
+2011-01-12 00:27 +0000 [r301403] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/file.c: Merged revisions 301402 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r301402 | tilghman | 2011-01-11 18:26:39 -0600 (Tue, 11 Jan 2011)
+ | 7 lines Call execl() directly for a better solution for paths
+ with spaces. (closes issue #18600) Reported by: ebroad Patches:
+ 20110111__issue18600__2.diff.txt uploaded by tilghman (license
+ 14) ........
+
+2011-01-11 19:19 +0000 [r301319] Paul Belanger <pabelanger@digium.com>
+
+ * /, configs/extensions.conf.sample: Merged revisions 301311 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r301311 | pabelanger | 2011-01-11 14:16:06 -0500
+ (Tue, 11 Jan 2011) | 9 lines Merged revisions 301310 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r301310 | pabelanger | 2011-01-11 14:14:31 -0500 (Tue,
+ 11 Jan 2011) | 2 lines Fix a logic issue when passing context ARG
+ ........ ................
+
+2011-01-11 18:55 +0000 [r301309] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/utils.c: Merged revisions 301308 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r301308 | mnicholson | 2011-01-11 12:51:40 -0600
+ (Tue, 11 Jan 2011) | 18 lines Merged revisions 301307 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r301307 | mnicholson | 2011-01-11 12:42:05 -0600
+ (Tue, 11 Jan 2011) | 11 lines Merged revisions 301305 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan
+ 2011) | 4 lines Prevent buffer overflows in ast_uri_encode()
+ ABE-2705 ........ ................ ................
+
+2011-01-10 22:40 +0000 [r301264] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/strcompat.c: Merged revisions 301263 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r301263 | tilghman | 2011-01-10 16:39:31 -0600 (Mon, 10 Jan 2011)
+ | 8 lines Little endian machines were not converted properly.
+ (closes issue #18583) Reported by: jcovert Patches:
+ 20110110__issue18583.diff.txt uploaded by tilghman (license 14)
+ Tested by: jcovert ........
+
+2011-01-09 21:42 +0000 [r301178-301222] Paul Belanger <pabelanger@digium.com>
+
+ * /, configure, configure.ac, autoconf/ast_ext_lib.m4: Merged
+ revisions 301221 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r301221 | pabelanger | 2011-01-09 16:40:34 -0500
+ (Sun, 09 Jan 2011) | 21 lines Merged revisions 301220 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan
+ 2011) | 14 lines SOUND_CACHE_DIR now defaults to empty Sounds
+ files included in the Asterisk tarball were being ignored and
+ re-downloaded. Users wanting to cache the files can still
+ override the setting using the --with-sounds-cache option.
+ (closes issue #18589) Reported by: pabelanger Patches:
+ issue18589.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger Review: https://reviewboard.asterisk.org/r/1074/
+ ........ ................
+
+ * /, apps/app_verbose.c: Merged revisions 301177 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r301177 | pabelanger | 2011-01-08 17:00:12 -0500
+ (Sat, 08 Jan 2011) | 14 lines Merged revisions 301176 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301176 | pabelanger | 2011-01-08 16:58:24 -0500 (Sat, 08 Jan
+ 2011) | 7 lines Indicate log level argument for Log() is not
+ optional (closes issue #18586) Reported by: kshumard Patches:
+ app_verbose.c.patch uploaded by kshumard (license 92) ........
+ ................
+
+2011-01-08 01:13 +0000 [r301135] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 301134 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r301134 | rmudgett | 2011-01-07 19:11:31 -0600 (Fri, 07
+ Jan 2011) | 7 lines The DTMF attended transfer feature cannot
+ callback a chan_dahdi BRI phone. The DAHDI ISDN channel name is
+ not dialable. Make a channel name like DAHDI/i3/400-12 dialable
+ when the sequence number is stripped off of the name. ........
+
+2011-01-07 20:53 +0000 [r301091] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 301090 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r301090 | qwell | 2011-01-07 14:53:02 -0600
+ (Fri, 07 Jan 2011) | 15 lines Merged revisions 301089 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) |
+ 8 lines Initialize useropts/adminopts in case there is no column
+ in the realtime DB. (closes issue #18182) Reported by: dimas
+ Patches: v1-18182.patch uploaded by dimas (license 88) Tested by:
+ dimas ........ ................
+
+2011-01-07 19:58 +0000 [r301048] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 301047 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r301047 | jpeeler | 2011-01-07 13:58:30 -0600
+ (Fri, 07 Jan 2011) | 15 lines Merged revisions 301046 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07 Jan 2011)
+ | 8 lines Fix regression causing forwarding voicemails to not
+ work with file storage. I had actually already fixed this in
+ 295200 in 1.4 and thought it wasn't missing in the other branches
+ for some reason. (closes issue #18358) Reported by: cabal95
+ ........ ................
+
+2011-01-07 18:23 +0000 [r301008] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * funcs/func_curl.c: Oops, missed the actual decoding part. (closes
+ issue #18046) Reported by: wdoekes
+
+2011-01-07 17:24 +0000 [r300959] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 300955 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r300955 | jpeeler | 2011-01-07 11:24:14 -0600
+ (Fri, 07 Jan 2011) | 21 lines Merged revisions 300951 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r300951 | jpeeler | 2011-01-07 11:23:37 -0600
+ (Fri, 07 Jan 2011) | 14 lines Merged revisions 300918 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011)
+ | 7 lines Ensure good bye prompt in voicemail is played at the
+ correct time. Specifically in the case of timing out but not
+ leaving voicemail nothing should be heard. And when leaving
+ voicemail it should be heard. ABE-2647 ........ ................
+ ................
+
+2011-01-07 07:47 +0000 [r300882] Mark Murawki <markm@intellasoft.net>
+
+ * res/res_config_pgsql.c: Added support for postgres database retry
+ query on disconnection to res_config_pgsql If your postgres
+ connection died suddenly in between res_config_pgsql queries, the
+ next query will fail because the query is executed on a
+ disconnected/disconnecting handle. The query is abandoned and is
+ returned from in error. Now we will reconnect and try again if a
+ query was run on a disconnected connection. (closes issue #18071)
+
+2011-01-06 17:50 +0000 [r300799-300841] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * Makefile, funcs/func_curl.c: XML validation
+
+ * funcs/func_curl.c: Add a hashcompat mode called "legacy", which
+ translates a literal plus sign to a space. (closes issue #18046)
+ Reported by: wdoekes Patches: 20100930__issue18046.diff.txt
+ uploaded by tilghman (license 14)
+
+ * /, addons/res_config_mysql.c: Merged revisions 300798 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r300798 | tilghman | 2011-01-06 00:28:18 -0600 (Thu, 06 Jan 2011)
+ | 8 lines Don't destroy handle not created by use (because the
+ caller will). (closes issue #18526) Reported by: makoto Patches:
+ res-config-mysql-include.patch uploaded by makoto (license 38)
+ Tested by: makoto ........
+
+2011-01-06 01:41 +0000 [r300761] David Ruggles <thedavidfactor@gmail.com>
+
+ * Makefile, contrib/scripts/safe_asterisk: update safe_asterisk
+ script change defaults to make a little more sense. Default log
+ location is now asterisk log location and default email
+ notification has been changed to root on the local machine
+ Review: https://reviewboard.asterisk.org/r/1067/
+
+2011-01-05 21:07 +0000 [r300716] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 300714 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r300714 | rmudgett | 2011-01-05 14:54:21 -0600
+ (Wed, 05 Jan 2011) | 21 lines Merged revision 300711 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed,
+ 05 Jan 2011) | 14 lines A call retrieved from hold may wind up
+ with no audio. If the retrieved call is natively bridged then the
+ call may not have any audio path. The following warning message
+ is given: "Failed to add <dfd> to conference <chan>/<chan>:
+ Invalid argument". * Open the media on a B channel when
+ pri_fixup_principle() moves the call from a no_b_channel channel
+ to a real channel. * Added lock protection while
+ pri_fixup_principle() moves a call from one private structure to
+ another. * Made some pri_fixup_principle() messages more
+ meaningful. .......... ................
+
+2011-01-05 18:57 +0000 [r300624] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, res/res_odbc.c: Merged revisions 300623 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r300623 | tilghman | 2011-01-05 12:56:12 -0600
+ (Wed, 05 Jan 2011) | 24 lines Merged revisions 300622 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r300622 | tilghman | 2011-01-05 12:54:58 -0600
+ (Wed, 05 Jan 2011) | 17 lines Merged revisions 300621 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011)
+ | 10 lines Use the sanity check in place of the
+ disconnect/connect cycle. The disconnect/connect cycle has the
+ potential to cause random crashes. (closes issue #18243) Reported
+ by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147)
+ Tested by: ks3 ........ ................ ................
+
+2011-01-05 16:30 +0000 [r300576] Paul Belanger <pabelanger@digium.com>
+
+ * /, cdr/cdr_sqlite.c: Merged revisions 300575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r300575 | pabelanger | 2011-01-05 11:29:19 -0500
+ (Wed, 05 Jan 2011) | 13 lines Merged revisions 300574 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r300574 | pabelanger | 2011-01-05 11:28:07 -0500 (Wed, 05 Jan
+ 2011) | 6 lines Change deprecated message to LOG_WARNING Also
+ removed latter part of message Discussed on #asterisk-dev
+ ........ ................
+
+2011-01-04 21:54 +0000 [r300434-300522] Leif Madsen <lmadsen@digium.com>
+
+ * /, channels/chan_sip.c, channels/chan_agent.c,
+ channels/chan_iax2.c, main/xmldoc.c: Merged revisions 300521 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r300521 | lmadsen | 2011-01-04 15:53:27 -0600
+ (Tue, 04 Jan 2011) | 17 lines Merged revisions 300520 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011)
+ | 9 lines Fix backwards and broken XML documentation. (closes
+ issue #18547) Reported by: jcovert Patches: xmldoc.c.patch
+ uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded
+ by jcovert (license 551) chan_sip.c.patch uploaded by jcovert
+ (license 551) chan_agent.c.patch uploaded by jcovert (license
+ 551) ........ ................
+
+ * configs/users.conf.sample, /: Merged revisions 300433 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r300433 | lmadsen | 2011-01-04 15:00:55 -0600
+ (Tue, 04 Jan 2011) | 15 lines Merged revisions 300431 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r300431 | lmadsen | 2011-01-04 15:00:29 -0600 (Tue, 04 Jan 2011)
+ | 7 lines Add some documentation to users.conf.sample. (closes
+ issue #18531) Reported by: lathama Patches:
+ users.conf.sample2.diff uploaded by lathama (license 1028) Tested
+ by: lathama ........ ................
+
+2011-01-04 21:00 +0000 [r300432] Russell Bryant <russell@digium.com>
+
+ * /, contrib/scripts/autosupport.8, contrib/scripts/autosupport:
+ Merged revisions 300430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r300430 | russell | 2011-01-04 15:00:16 -0600
+ (Tue, 04 Jan 2011) | 18 lines Merged revisions 300429 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r300429 | russell | 2011-01-04 14:59:56 -0600
+ (Tue, 04 Jan 2011) | 11 lines Merged revisions 300428 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011)
+ | 4 lines Update the autosupport script from Digium support.
+ (closes AST-395) ........ ................ ................
+
+2011-01-04 19:45 +0000 [r300385] Leif Madsen <lmadsen@digium.com>
+
+ * phoneprov/000000000000.cfg, /: Merged revisions 300384 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r300384 | lmadsen | 2011-01-04 13:45:22 -0600 (Tue, 04 Jan 2011)
+ | 7 lines Update STAT() to use the comma instead of the pipe.
+ (closes issue #18503) Reported by: cjacobsen Patches:
+ old_separator.diff uploaded by cjacobsen (license 1029) Tested
+ by: lathama ........
+
+2011-01-04 18:51 +0000 [r300345] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: Update MFC-R2 code to use new DTMF-R2
+ functionality in OpenR2 (closes issue #18576)
+
+2011-01-04 18:06 +0000 [r300302] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 300301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r300301 | twilson | 2011-01-04 11:54:41 -0600
+ (Tue, 04 Jan 2011) | 29 lines Merged revisions 300298 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r300298 | twilson | 2011-01-04 11:37:26 -0600
+ (Tue, 04 Jan 2011) | 22 lines Merged revisions 300216 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011)
+ | 15 lines Don't authenticate SUBSCRIBE re-transmissions This
+ only skips authentication on retransmissions that are already
+ authenticated. A similar method is already used for INVITES. This
+ is the kind of thing we end up having to do when we don't have a
+ transaction layer... (closes issue #18075) Reported by: mdu113
+ Patches: diff.txt uploaded by twilson (license 396) Tested by:
+ twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/
+ ........ ................ ................
+
+2011-01-04 17:04 +0000 [r300215] Jan Kalab <pitlicek@gmail.com>
+
+ * res/res_calendar_exchange.c, res/res_calendar_icalendar.c, /:
+ Merged revisions 300214 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r300214 | pitel | 2011-01-04 18:01:52 +0100 (Út, 04 led 2011) | 7
+ lines Memory leaking in calendars ne_request_destroy() was
+ missing in icalendar and exchange calendar modules, causing
+ memory leak. (closes issue #18521) Review:
+ https://reviewboard.asterisk.org/r/1068/ ........
+
+2011-01-04 16:38 +0000 [r300168-300212] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, UPGRADE.txt, CHANGES,
+ channels/sig_pri.c: Optional HOLD/RETRIEVE signaling for PTMP TE
+ when the bridge goes on and off hold. Added the moh_signaling
+ option to specify what to do when the channel's bridged peer puts
+ the ISDN channel on and off of hold. Implemented as a FSM to
+ control libpri ISDN signaling when the bridged peer places the
+ channel on and off of hold with the AST_CONTROL_HOLD and
+ AST_CONTROL_UNHOLD control frames. JIRA SWP-2687 JIRA ABE-2691
+ Review: https://reviewboard.asterisk.org/r/1063/
+
+ * /, main/features.c: Merged revisions 300166 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r300166 | rmudgett | 2011-01-03 17:14:55 -0600
+ (Mon, 03 Jan 2011) | 11 lines Merged revisions 300165 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r300165 | rmudgett | 2011-01-03 17:02:13 -0600 (Mon, 03 Jan 2011)
+ | 4 lines Use correct variable for atxfercallbackretries config
+ option. * Misc formatting changes. ........ ................
+
+2011-01-03 14:09 +0000 [r300121] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: initialize playing_silence in struct
+ initialization playing_silence was not initialized with the
+ struct was initialized, it was being set after the fact which
+ caused problems if something that relied on playing_silence being
+ set was called too quickly (closes issue #18430) Reported by:
+ stevebrandli Patches: externalivr.patch uploaded by
+ thedavidfactor (license 903) Tested by: thedavidfactor,
+ stevebrandli
+
+2011-01-03 13:15 +0000 [r300083] Leif Madsen <lmadsen@digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 300082 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r300082 | lmadsen | 2011-01-03 07:14:25 -0600 (Mon, 03 Jan 2011)
+ | 11 lines Increase side of mapping response field. I've
+ increased the size of the response field in a DUNDi mapping
+ because of some documentation I'm writing. Previously it was set
+ to AST_MAX_EXTENSION which is only 80 characters, which is far
+ too small when you're using some dialplan functions to craft a
+ response. The example I'm using is: extensions =>
+ RegisteredDevices,0,SIP,dundi:very_awesome_password/${IF($[${DB_EXISTS(phones/${NUMBER}/device)}]?${DB(phones/${NUMBER}/device)}:None)},nopartial
+ ........
+
+2010-12-31 09:29 +0000 [r300044-300045] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * cdr/cdr_adaptive_odbc.c, CHANGES,
+ configs/cdr_adaptive_odbc.conf.sample: Support negative filters.
+ (closes issue #17979) Reported by: tilghman Patches:
+ 20100911__for_blitzrage.diff.txt uploaded by tilghman (license
+ 14) Tested by: lmadsen
+
+ * main/logger.c, CHANGES: Support an alternate configuration file
+ for the 'logger reload' command. (closes issue #17668) Reported
+ by: tilghman Patches: 20100718__logger_reload_altconf__2.diff.txt
+ uploaded by tilghman (license 14) Review: (by lmadsen, russell
+ within comments on issue tracker)
+
+2010-12-29 22:19 +0000 [r300000] Sean Bright <sean@malleable.com>
+
+ * main/asterisk.c: Remove some trailing whitespace and steal
+ revision 300000.
+
+2010-12-29 22:03 +0000 [r299990] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/file.c, apps/app_voicemail.c: Merged revisions 299989 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r299989 | tilghman | 2010-12-29 16:02:59 -0600 (Wed, 29 Dec 2010)
+ | 4 lines Quote arguments, just in case there's a space in a
+ pathname. (Diagnosed by pabelanger on #asterisk-dev, fixed by
+ me.) ........
+
+2010-12-29 19:29 +0000 [r299866-299949] Paul Belanger <pabelanger@digium.com>
+
+ * /, sounds/Makefile: Merged revisions 299948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r299948 | pabelanger | 2010-12-29 14:28:36 -0500 (Wed, 29 Dec
+ 2010) | 2 lines Only remove /tmp/astdatadir, not
+ /var/lib/asterisk ........
+
+ * Makefile, /, build_tools/make_sample_voicemail, sounds/Makefile:
+ Merged revisions 299907 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r299907 | pabelanger | 2010-12-29 13:22:23 -0500 (Wed, 29 Dec
+ 2010) | 2 lines Properly quote varibles for MAC OS X ........
+
+ * /, apps/app_chanspy.c: Merged revisions 299865 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r299865 | pabelanger | 2010-12-28 13:53:37 -0500
+ (Tue, 28 Dec 2010) | 9 lines Merged revisions 299864 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r299864 | pabelanger | 2010-12-28 13:51:13 -0500 (Tue,
+ 28 Dec 2010) | 2 lines Documentation typo ........
+ ................
+
+2010-12-27 21:23 +0000 [r299754-299824] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, sounds/Makefile: Merged revisions 299820 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r299820 | tilghman | 2010-12-27 15:23:10 -0600 (Mon, 27 Dec 2010)
+ | 2 lines More space-in-pathname issues. ........
+
+ * Makefile, /, Makefile.moddir_rules, sounds/Makefile: Merged
+ revisions 299794 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r299794 | tilghman | 2010-12-27 14:41:04 -0600 (Mon, 27 Dec 2010)
+ | 2 lines Mac OS X spaces-in-pathnames fix. ........
+
+ * /, configure, configure.ac: Merged revisions 299752 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r299752 | tilghman | 2010-12-26 15:15:58 -0600 (Sun, 26
+ Dec 2010) | 2 lines Properly quote path on Darwin. ........
+
+2010-12-25 16:35 +0000 [r299715] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooh245.c,
+ addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c:
+ Change order of sending TCS and MSD packets Change order of
+ sending Terminal Capability Set and MasterSlave Determination
+ packets, MSD send when TCS exchange procedure is done (we send
+ tcs ack to remote and we have remote tcs ack already or we
+ receive tcs ack from remote and we have send our tcs ack to
+ remote already). Some endpoints can work in this sequence only, i
+ suggest they can't work with both (tcs and msd) exchange
+ procedures simultaneously. Also changed StartH245 facility
+ message sending. It send on incoming calls only due to some
+ endpoints can't proccess properly this facility messages on their
+ incoming calls. (closes issue #18433) Reported by: MrHanMan
+ Patches: tcs-msd-h245-3.patch uploaded by may213 (license 454)
+ Tested by: MrHanMan, may213
+
+2010-12-25 10:08 +0000 [r299584-299627] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * channels/chan_local.c, /: Merged revisions 299626 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r299626 | tilghman | 2010-12-25 04:07:15 -0600
+ (Sat, 25 Dec 2010) | 19 lines Merged revisions 299625 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r299625 | tilghman | 2010-12-25 04:05:00 -0600
+ (Sat, 25 Dec 2010) | 12 lines Merged revisions 299624 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010)
+ | 5 lines Move check for extension existence below variable
+ inheritance, due to the possible use of an eswitch. (closes issue
+ #16228) Reported by: jlaguilar ........ ................
+ ................
+
+ * /, addons/res_config_mysql.c: Merged revisions 299583 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r299583 | tilghman | 2010-12-24 11:58:30 -0600 (Fri, 24 Dec 2010)
+ | 7 lines Reset 'first' variable after usage. (closes issue
+ #18525) Reported by: makoto Patches:
+ res-config-mysql-update2.patch uploaded by makoto (license 38)
+ ........
+
+2010-12-23 01:46 +0000 [r299493] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: Enqueue AST_CONTROL_PROGRESS after
+ AST_CONTROL_RINGING when MFC-R2 calls are accepted (closes issue
+ #18438) Reported by: mariner7 Tested by: moy
+
+2010-12-22 20:10 +0000 [r299450] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * res/ael/pval.c, pbx/ael/ael-test/ref.ael-vtest25,
+ pbx/ael/ael-test/ref.ael-vtest17, /,
+ pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 299449 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r299449 | tilghman | 2010-12-22 14:05:02 -0600
+ (Wed, 22 Dec 2010) | 15 lines Merged revisions 299448 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r299448 | tilghman | 2010-12-22 14:03:30 -0600 (Wed, 22 Dec 2010)
+ | 8 lines Resolve warnings by disambiguating the "s" extension as
+ used by chan_dahdi from the "s" extension as used by the AEL
+ macros. (closes issue #18480) Reported by: nivek Patches:
+ 20101215__issue18480__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: nivek ........ ................
+
+2010-12-22 02:12 +0000 [r299406] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 299405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r299405 | rmudgett | 2010-12-21 20:10:39 -0600 (Tue, 21 Dec 2010)
+ | 17 lines Chan_dahdi sends an empty COLP on the bridged channel.
+ Chan_dahdi always inserts a connected party IE when you call from
+ one dahdi channel to another dahdi channel, even if no such
+ information was received on the 2nd channel. This clears the
+ display of many phones. * Removed leftover artifact from before
+ the valid flag was added. * Updated all of the channel's caller
+ id information with the new connected line information instead of
+ just the string parts. (closes issue #18508) Reported by: wimpy
+ Patches: issue18508_trunk.patch uploaded by rmudgett (license
+ 664) Tested by: wimpy, rmudgett ........
+
+2010-12-21 16:02 +0000 [r299355] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 299353 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r299353 | mnicholson | 2010-12-21 09:25:03 -0600
+ (Tue, 21 Dec 2010) | 30 lines Merged revisions 299242 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r299242 | mnicholson | 2010-12-20 15:25:35 -0600
+ (Mon, 20 Dec 2010) | 23 lines Merged revisions
+ 299194,299198,299220 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec
+ 2010) | 6 lines Respond as soon as possible with a 202 Accepted
+ to refer requests. This change also plugs a few memory leaks that
+ can occur when parking sip calls. ABE-2656 ........ r299198 |
+ mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2
+ lines Remove changes to via processing that were not supposed to
+ go into the last commit. ........ r299220 | mnicholson |
+ 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use
+ ast_free() instead of free() ABE-2656 ........ ................
+ ................
+
+2010-12-21 00:45 +0000 [r299313] Paul Belanger <pabelanger@digium.com>
+
+ * configs/cel.conf.sample, /: Merged revisions 299312 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r299312 | pabelanger | 2010-12-20 19:44:08 -0500 (Mon,
+ 20 Dec 2010) | 8 lines Correct typo with USER_DEFINED event.
+ (closes issue #18461) Reported by: joscas Patches:
+ cel.conf.sample.diff uploaded by lathama (license 1028) Tested
+ by: lathama, joscas ........
+
+2010-12-20 21:40 +0000 [r299249] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 299248 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r299248 | mmichelson | 2010-12-20 15:38:30 -0600 (Mon, 20 Dec
+ 2010) | 20 lines Fix a couple of CCSS issues. * Make sure to
+ allocate a cc_params structure when creating autopeers. * Use
+ sip_uri_cmp when retrieving SIP CC agents and monitors in case
+ parameters appear in the URI. (closes issue #18504) Reported by:
+ kkm (closes issue #18338) Reported by: GeorgeKonopacki Patches:
+ 18338.diff uploaded by mmichelson (license 60) Tested by:
+ GeorgeKonopacki ........
+
+2010-12-20 18:18 +0000 [r299142] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, sample.call: Merged revisions 299138 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r299138 | tilghman | 2010-12-20 12:17:28 -0600
+ (Mon, 20 Dec 2010) | 9 lines Merged revisions 299136 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r299136 | tilghman | 2010-12-20 12:16:37 -0600 (Mon, 20
+ Dec 2010) | 2 lines Documentation fix ........ ................
+
+2010-12-20 18:03 +0000 [r299135] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/astobj2.h, main/astobj2.c: New astobj2 flag for
+ issuing a callback without locking the container.
+
+2010-12-20 17:59 +0000 [r299133-299134] Russell Bryant <russell@digium.com>
+
+ * channels/chan_misdn.c: Fix chan_misdn build after sched API
+ changes.
+
+ * addons/chan_ooh323.c, addons/chan_mobile.c: Fix some build errors
+ in addons due to sched API changes.
+
+2010-12-20 17:48 +0000 [r299132] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, cdr/cdr_pgsql.c: Merged revisions 299131 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r299131 | tilghman | 2010-12-20 11:47:10 -0600
+ (Mon, 20 Dec 2010) | 18 lines Merged revisions 299130 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r299130 | tilghman | 2010-12-20 11:41:24 -0600 (Mon, 20 Dec 2010)
+ | 11 lines If a call was not answered, then the billsec was
+ calculated unusually large. Also, due to a copy and paste error,
+ a request for the answer field would have given the start value,
+ instead. (closes issue #18460) Reported by: joscas Patches:
+ 20101215__issue18460.diff.txt uploaded by tilghman (license 14)
+ Tested by: joscas ........ ................
+
+2010-12-20 17:15 +0000 [r299091] Russell Bryant <russell@digium.com>
+
+ * channels/chan_unistim.c, main/udptl.c, res/res_rtp_asterisk.c,
+ include/asterisk.h, main/rtp_engine.c, main/dnsmgr.c,
+ channels/chan_sip.c, main/ccss.c, include/asterisk/channel.h,
+ channels/chan_gtalk.c, tests/test_sched.c, channels/chan_iax2.c,
+ res/res_rtp_multicast.c, main/channel.c, main/cdr.c,
+ channels/chan_jingle.c, channels/chan_skinny.c,
+ channels/sip/include/globals.h, res/res_stun_monitor.c,
+ channels/sip/dialplan_functions.c, channels/chan_h323.c,
+ include/asterisk/sched.h, pbx/pbx_dundi.c,
+ include/asterisk/udptl.h, include/asterisk/rtp_engine.h,
+ main/sched.c, channels/chan_mgcp.c, res/res_calendar.c: Some
+ scheduler API cleanup and improvements. Previously, I had added
+ the ast_sched_thread stuff that was a generic scheduler thread
+ implementation. However, if you used it, it required using
+ different functions for modifying scheduler contents. This patch
+ reworks how this is done and just allows you to optionally start
+ a thread on the original scheduler context structure that has
+ always been there. This makes it trivial to switch to the generic
+ scheduler thread implementation without having to touch any of
+ the other code that adds or removes scheduler entries. In
+ passing, I made some naming tweaks to add ast_ prefixes where
+ they were not there before. Review:
+ https://reviewboard.asterisk.org/r/1007/
+
+2010-12-20 16:19 +0000 [r299089] Leif Madsen <lmadsen@digium.com>
+
+ * /, main/features.c: Merged revisions 299088 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r299088 | lmadsen | 2010-12-20 10:18:26 -0600
+ (Mon, 20 Dec 2010) | 13 lines Merged revisions 299087 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r299087 | lmadsen | 2010-12-20 10:18:03 -0600 (Mon, 20 Dec 2010)
+ | 5 lines Note that Park() timeout is milliseconds. (closes issue
+ #15758) Reported by: mmurdock Tested by: mmurdock, seanbright
+ ........ ................
+
+2010-12-20 09:14 +0000 [r299005] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/sig_pri.h, channels/chan_sip.c, main/aoc.c: Typos:
+ recieved => received
+
+2010-12-18 00:08 +0000 [r298819-298961] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * utils/refcounter.c, include/asterisk/utils.h,
+ build_tools/cflags-devmode.xml, /, main/Makefile,
+ include/asterisk/autoconfig.h.in, configure.ac, utils/hashtest.c,
+ main/utils.c, main/astobj2.c, utils/conf2ael.c,
+ include/asterisk/logger.h, configure,
+ build_tools/menuselect-deps.in, main/logger.c, utils/hashtest2.c,
+ utils/ael_main.c, makeopts.in, utils/check_expr.c: Merged
+ revisions 298960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r298960 | tilghman | 2010-12-17 17:52:04 -0600
+ (Fri, 17 Dec 2010) | 20 lines Merged revisions 298957 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298957 | tilghman | 2010-12-17 17:30:55 -0600
+ (Fri, 17 Dec 2010) | 13 lines Merged revisions 298905 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010)
+ | 6 lines Let Asterisk find better backtrace information with
+ libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will
+ use libbfd to search for better symbol information within both
+ the Asterisk binary, as well as loaded modules, to assist when
+ using inline backtraces to track down problems. Review:
+ https://reviewboard.asterisk.org/r/1055/ ........
+ ................ ................
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 298827 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r298827 | tilghman | 2010-12-17 15:18:18 -0600 (Fri, 17 Dec 2010)
+ | 8 lines -v implies -f, so override with -F. (closes issue
+ #18446) Reported by: lathama Patches: rc.debian.asterisk.diff
+ uploaded by lathama (license 1028) Tested by: lathama ........
+
+ * /, configure, configure.ac: Merged revisions 298818 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r298818 | tilghman | 2010-12-17 15:04:21 -0600
+ (Fri, 17 Dec 2010) | 15 lines Merged revisions 298817 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r298817 | tilghman | 2010-12-17 15:03:06 -0600 (Fri, 17 Dec 2010)
+ | 8 lines Also include PTHREAD_LIBS and PTHREAD_CFLAGS for SQLite
+ 3, as it's needed on some platforms. (closes issue #18493)
+ Reported by: pprindeville Patches: asterisk-1.8-sqlite3.patch
+ uploaded by pprindeville (license 347) Tested by: pprindeville
+ ........ ................
+
+2010-12-17 17:29 +0000 [r298774] Brad Watkins <Marquis42@gmail.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 298773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r298773 | marquis | 2010-12-17 12:26:31 -0500 (Fri, 17 Dec 2010)
+ | 10 lines Fix parsing of mwi => lines in sip.conf Reworking
+ parsing of mwi => lines to resolve a segfault. Also add a set of
+ unit tests for the function that does the parsing. (closes issue
+ #18350) Reported by: gbour Tested by: Marquis, gbour Review:
+ https://reviewboard.asterisk.org/r/1053/ ........
+
+2010-12-16 23:33 +0000 [r298599-298686] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 298685 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r298685 | jpeeler | 2010-12-16 17:31:50 -0600
+ (Thu, 16 Dec 2010) | 16 lines Merged revisions 298684 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298684 | jpeeler | 2010-12-16 17:30:59 -0600
+ (Thu, 16 Dec 2010) | 9 lines Merged revisions 298683 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16
+ Dec 2010) | 2 lines After recording only silence for a voicemail
+ prepending, restore backup files. ........ ................
+ ................
+
+ * /, apps/app_queue.c: Merged revisions 298598 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r298598 | jpeeler | 2010-12-16 14:51:44 -0600
+ (Thu, 16 Dec 2010) | 21 lines Merged revisions 298597 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298597 | jpeeler | 2010-12-16 14:49:33 -0600
+ (Thu, 16 Dec 2010) | 14 lines Merged revisions 298596 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010)
+ | 7 lines Fix improper hangup when doing an attended transfer to
+ queue. Had to indicate ringing in wait_for_answer so the attended
+ transfer code would not try and hang up the local channel it
+ created, which would kill the call. ABE-2624 ........
+ ................ ................
+
+2010-12-16 09:29 +0000 [r298441-298545] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, channels/chan_sip.c: Merged revisions 298539 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r298539 | tilghman | 2010-12-16 03:28:17 -0600 (Thu, 16 Dec 2010)
+ | 8 lines Ensure the ipaddr field in realtime is large enough to
+ handle IPv6 addresses. (closes issue #18464) Reported by: IgorG
+ Patches: realtime_ipv6store.diff uploaded by IgorG (license 20)
+ (plus a few additional lines by tilghman) ........
+
+ * res/res_config_odbc.c, /: Merged revisions 298482 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r298482 | tilghman | 2010-12-16 03:05:28 -0600
+ (Thu, 16 Dec 2010) | 28 lines Merged revisions 298481 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298481 | tilghman | 2010-12-16 03:04:38 -0600
+ (Thu, 16 Dec 2010) | 21 lines Merged revisions 298480 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298480 | tilghman | 2010-12-16 03:03:40 -0600 (Thu, 16 Dec 2010)
+ | 14 lines Only increment the pointer once per loop, otherwise we
+ corrupt the value. (closes issue #18251) Reported by: bcnit
+ Patches: 20101110__issue18251.diff.txt uploaded by tilghman
+ (license 14) Tested by: trev, jthurman, elguero (closes issue
+ #18279) Reported by: zerohalo Patches:
+ 20101109__issue18279.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo ........ ................ ................
+
+ * /, funcs/func_dialgroup.c: Merged revisions 298478 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r298478 | tilghman | 2010-12-16 02:56:13 -0600
+ (Thu, 16 Dec 2010) | 15 lines Merged revisions 298477 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r298477 | tilghman | 2010-12-16 02:54:23 -0600 (Thu, 16 Dec 2010)
+ | 8 lines Eliminate duplicates from container. (closes issue
+ #18091) Reported by: bunny Patches: 20101006__issue18091.diff.txt
+ uploaded by tilghman (license 14) Tested by: bunny ........
+ ................
+
+ * /, cdr/cdr_sqlite.c: Merged revisions 298394 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r298394 | tilghman | 2010-12-15 18:30:04 -0600
+ (Wed, 15 Dec 2010) | 22 lines Merged revisions 298393 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298393 | tilghman | 2010-12-15 18:29:10 -0600
+ (Wed, 15 Dec 2010) | 15 lines Merged revisions 298392 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298392 | tilghman | 2010-12-15 18:28:04 -0600 (Wed, 15 Dec 2010)
+ | 8 lines Unregister before shutting down the connection, to
+ avoid a race. (closes issue #18481) Reported by: pabelanger
+ Patches: 20101215__issue18481.diff.txt uploaded by tilghman
+ (license 14) Tested by: pabelanger ........ ................
+ ................
+
+2010-12-13 22:10 +0000 [r298201-298288] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Post AMI hold events on PRI spans when the
+ remote party HOLD/RETRIEVEs the call. Part of JIRA
+ SWP-2687/ABE-2691.
+
+ * channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions
+ 298195 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r298195 | rmudgett | 2010-12-13 11:11:43 -0600
+ (Mon, 13 Dec 2010) | 33 lines Merged revisions 298194 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298194 | rmudgett | 2010-12-13 11:04:41 -0600
+ (Mon, 13 Dec 2010) | 26 lines Merged revisions 298193 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010)
+ | 19 lines Outgoing PRI/BRI calls cannot do DTMF triggered
+ transfers. Outgoing PRI/BRI calls cannot do DTMF triggered
+ transfers if a PROCEEDING message is not received. The debug
+ output shows that the DTMF begin event is seen, but the DTMF end
+ event is missing. When the DTMF begin happens, the call is muted
+ so we now have one way audio (until a DTMF end event is somehow
+ seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER
+ event is received. * Made absorb the DTMF begin and DTMF end
+ events if we are overlap dialing and have not seen a PROCEEDING
+ message. * Added a debug message when absorbing a DTMF event.
+ JIRA SWP-2690 JIRA ABE-2697 ........ ................
+ ................
+
+2010-12-12 03:58 +0000 [r298137] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/utils.h, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, main/logger.c,
+ main/utils.c, main/asterisk.c: Add support for several platforms
+ to obtain the real thread ID. Already had the pthread ID which is
+ not the same. The most obvious enhancement is in the "core show
+ threads" output. As stated in the utils header, if the platform
+ isn't supported -1 is reported (instead of the process ID
+ previously).
+
+2010-12-11 21:47 +0000 [r298100] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c: Correction to work with
+ gatekeeper which don't send GK ID Don't use GK ID if it's not
+ presented in GK replies Extract GK ID not only in GK confirm but
+ in GK register confirm also (closes issue #18401) Reported by:
+ MrHanMan Patches: no-gkid-2.patch uploaded by may213 (license
+ 454) Tested by: may213, MrHanMan
+
+2010-12-10 16:53 +0000 [r298055] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 298054 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r298054 | mnicholson | 2010-12-10 10:52:11 -0600 (Fri, 10 Dec
+ 2010) | 2 lines Prevent a memcpy overlap in
+ GENERIC_FAX_EXEC_SET_VARS ........
+
+2010-12-10 16:28 +0000 [r298052] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/netsock.c: Merged revisions 298051 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r298051 | tilghman | 2010-12-10 10:26:46 -0600
+ (Fri, 10 Dec 2010) | 18 lines Merged revisions 298050 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r298050 | tilghman | 2010-12-10 10:24:13 -0600 (Fri, 10 Dec 2010)
+ | 11 lines Portability issue on OpenSolaris. Also detect the
+ required structure element, because OpenSolaris defines
+ SIOCGIFHWADDR, but without support for IP sockets. (closes issue
+ #18442) Reported by: ranjtech Patches:
+ 20101209__issue18442.diff.txt uploaded by tilghman (license 14)
+ Tested by: ranjtech ........ ................
+
+2010-12-09 22:19 +0000 [r297972] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297965 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r297965 | twilson | 2010-12-09 16:18:19 -0600
+ (Thu, 09 Dec 2010) | 28 lines Merged revisions 297960 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297960 | twilson | 2010-12-09 16:10:31 -0600
+ (Thu, 09 Dec 2010) | 21 lines Merged revisions 297959 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010)
+ | 14 lines Ignore spurious REGISTER requests If a REGISTER
+ request with a Call-ID matching an existing transaction is
+ received it was possible that the REGISTER request would
+ overwrite the initreq of the private structure. This info is used
+ to generate messages for other responses in the transaction. This
+ patch ignores REGISTER requests that match non-REGISTER
+ transactions. (closes issue #18051) Reported by: eeman Tested by:
+ twilson Review: https://reviewboard.asterisk.org/r/1050/ ........
+ ................ ................
+
+2010-12-09 21:33 +0000 [r297958] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 297957 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r297957 | dvossel | 2010-12-09 15:32:20 -0600 (Thu, 09
+ Dec 2010) | 11 lines Fixes issue with outbound google voice calls
+ not working. Thanks to az1234 and nevermind_quack for their input
+ in helping debug the issue. (closes issue #18412) Reported by:
+ nevermind_quack Patches: fix uploaded by dvossel (license 671)
+ ........
+
+2010-12-09 21:26 +0000 [r297956] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 297952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r297952 | twilson | 2010-12-09 14:48:44 -0600 (Thu, 09 Dec 2010)
+ | 10 lines Don't crash after Set(CDR(userfield)=...) in
+ ast_bridge_call Instead of setting peer->cdr = NULL, set it to
+ not post. (closes issue #18415) Reported by: macbrody Patches:
+ patch-18415 uploaded by jsolares (license 1167) Tested by:
+ jsolares, twilson ........
+
+2010-12-08 18:08 +0000 [r297910] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, configs/extensions.conf.sample: Merged revisions 297909 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r297909 | tilghman | 2010-12-08 12:06:04 -0600
+ (Wed, 08 Dec 2010) | 11 lines Merged revisions 297908 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r297908 | tilghman | 2010-12-08 12:04:38 -0600 (Wed, 08 Dec 2010)
+ | 4 lines Use inheritance to get correct results for
+ SIPFROMDOMAIN. (from an internal Digium discussion) ........
+ ................
+
+2010-12-07 23:00 +0000 [r297826] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 297825 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r297825 | jpeeler | 2010-12-07 16:59:30 -0600
+ (Tue, 07 Dec 2010) | 26 lines Merged revisions 297824 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297824 | jpeeler | 2010-12-07 16:58:54 -0600
+ (Tue, 07 Dec 2010) | 19 lines Merged revisions 297823 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010)
+ | 12 lines Revert code that changed SSRC for DTMF. Some previous
+ behavior was attempted to be restored, but mistakingly I did not
+ realize that the previous behavior was incorrect. This fixes DTMF
+ not being detected since DTMF shouldn't cause the SSRC to change.
+ (related to issue #17404) (closes issue #18189) (closes issue
+ #18352) Reported by: marcbou Tested by: cmbaker82 ........
+ ................ ................
+
+2010-12-07 22:54 +0000 [r297734-297822] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * Makefile, contrib/init.d/org.asterisk.asterisk.plist,
+ utils/muted.c, /, contrib/init.d/org.asterisk.muted.plist
+ (added): Merged revisions 297821 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r297821 | tilghman | 2010-12-07 16:51:05 -0600
+ (Tue, 07 Dec 2010) | 18 lines Merged revisions 297819 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297819 | tilghman | 2010-12-07 16:40:45 -0600
+ (Tue, 07 Dec 2010) | 11 lines Merged revisions 297818 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010)
+ | 4 lines Use non-deprecated APIs for CoreAudio Review:
+ https://reviewboard.asterisk.org/r/1040/ ........
+ ................ ................
+
+ * /, apps/app_followme.c: Merged revisions 297733 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r297733 | tilghman | 2010-12-06 18:29:26 -0600
+ (Mon, 06 Dec 2010) | 22 lines Merged revisions 297713 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297713 | tilghman | 2010-12-06 18:21:50 -0600
+ (Mon, 06 Dec 2010) | 15 lines Merged revisions 297689 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010)
+ | 8 lines Don't create a Local channel if the target extension
+ does not exist. (closes issue #18126) Reported by: junky Patches:
+ followme.diff uploaded by junky (license 177) (partially
+ restructured by me to avoid a possible memory leak) ........
+ ................ ................
+
+2010-12-06 22:10 +0000 [r297608] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297607 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r297607 | jpeeler | 2010-12-06 16:06:37 -0600
+ (Mon, 06 Dec 2010) | 25 lines Merged revisions 297605 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297605 | jpeeler | 2010-12-06 16:03:04 -0600
+ (Mon, 06 Dec 2010) | 18 lines Merged revisions 297603 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010)
+ | 12 lines Improve handling of REGISTER requests with multiple
+ contact headers. The changes here attempt to more strictly follow
+ RFC 3261 section 10.3. Basically the following will now cause a
+ 400 Bad Response to be returned, if: - multiple Contact headers
+ are present with one set to expire all bindings ("*") - wildcard
+ parameter is specified for Contact without Expires header or
+ Expires header is not set to zero. ABE-2442 ABE-2443 ........
+ ................ ................
+
+2010-12-03 17:42 +0000 [r297536] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_console.c: Merged revisions 297535 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r297535 | seanbright | 2010-12-03 12:41:30 -0500
+ (Fri, 03 Dec 2010) | 9 lines Merged revisions 297534 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r297534 | seanbright | 2010-12-03 12:40:52 -0500 (Fri,
+ 03 Dec 2010) | 3 lines The CLI command should not contain
+ <placeholder>s, these are for descriptions. ........
+ ................
+
+2010-12-03 15:32 +0000 [r297496] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions
+ 297157,297486,297495 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r297157 | mnicholson | 2010-12-01 13:47:33 -0600 (Wed, 01 Dec
+ 2010) | 2 lines Changed some NOTICE and WARNING messages to DEBUG
+ messages. ........ r297486 | mnicholson | 2010-12-02 15:30:47
+ -0600 (Thu, 02 Dec 2010) | 6 lines Add support for reserving a
+ fax session before answering the channel. Note: this change
+ breaks ABI compatibility. FAX-217 ........ r297495 | mnicholson |
+ 2010-12-03 09:21:52 -0600 (Fri, 03 Dec 2010) | 4 lines Print a
+ DEBUG message instead of a WARNING message when the selected fax
+ tech does not support reserving sessions. Answer the channel
+ before quering it for t.38 support. This is necessary for the
+ query to work properly over local channels. ........
+
+2010-12-02 20:11 +0000 [r297407] Paul Belanger <pabelanger@digium.com>
+
+ * Makefile, /: Merged revisions 297406 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r297406 | pabelanger | 2010-12-02 15:09:29 -0500
+ (Thu, 02 Dec 2010) | 21 lines Merged revisions 297405 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297405 | pabelanger | 2010-12-02 15:06:43 -0500
+ (Thu, 02 Dec 2010) | 14 lines Merged revisions 297404 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec
+ 2010) | 7 lines Resolve compile error under FreeBSD We now set
+ _ASTCFLAGS+=-march=i686 for i386 processors, still allowing
+ ASTCFLAGS to override the setting. Review:
+ https://reviewboard.asterisk.org/r/1043/ ........
+ ................ ................
+
+2010-12-02 18:28 +0000 [r297356] Terry Wilson <twilson@digium.com>
+
+ * /, main/abstract_jb.c: Merged revisions 297312 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r297312 | twilson | 2010-12-02 12:13:49 -0600
+ (Thu, 02 Dec 2010) | 28 lines Merged revisions 297311 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297311 | twilson | 2010-12-02 12:07:39 -0600
+ (Thu, 02 Dec 2010) | 21 lines Merged revisions 297310 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010)
+ | 12 lines Initialize offset for adaptive jitter buffer When the
+ adaptive jitter buffer is enabled in sip.conf, the first frame
+ placed in the jitter buffer fails with something like:
+ jb_warning_output: Resyncing the jb. last_delay 0, this delay
+ -215886466, threshold 1000, new offset 215886466 This happens
+ because the offset is not initialized before calling jb_put().
+ This patch modifies jb_put_first_adaptive() to set the offset to
+ the frame's timestamp. Review:
+ https://reviewboard.asterisk.org/r/1041/ ........
+ ................ ................
+
+2010-12-02 13:20 +0000 [r297248] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 297245 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r297245 | russell | 2010-12-02 07:20:19 -0600
+ (Thu, 02 Dec 2010) | 20 lines Merged revisions 297229 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297229 | russell | 2010-12-02 07:16:47 -0600
+ (Thu, 02 Dec 2010) | 13 lines Merged revisions 297228 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010)
+ | 6 lines Add "DAHDI" to a couple of app_meetme error messages.
+ This is in response to some questions on IRC. To the user, there
+ was nothing that made it obvious that this error had anything to
+ do with DAHDI not being loaded. ........ ................
+ ................
+
+2010-12-01 17:53 +0000 [r297076] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297075 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r297075 | jpeeler | 2010-12-01 11:53:13 -0600
+ (Wed, 01 Dec 2010) | 37 lines Merged revisions 297073 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297073 | jpeeler | 2010-12-01 11:52:46 -0600
+ (Wed, 01 Dec 2010) | 30 lines Merged revisions 297072 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010)
+ | 23 lines Fix not stopping MOH when transfered local channel
+ queue member is answered. The problem here is only present when
+ local channels are used with the MOH passthru option as well as
+ no optimization (/nm). I will describe the slightly bizarre
+ scenario that was used to test, where phones B and C are queue
+ members: Phone A dials into a queue with two members using local
+ channels and the above options. Phone B answers. Phone A blind
+ transfers phone B into the same queue. Phone A hangs up. Phone C
+ answers, but phone B didn't stop playing MOH. In this scenario,
+ the unhold frame that should have gotten to phone B never arrived
+ due to the masquerade from the blind transfer. This is usually
+ fine since app_queue manages the starting and stopping of MOH.
+ However, with the passthrough option enabled when app_queue
+ attempts to stop MOH it tries to do so on the local channel
+ rather than the real channel. The easiest solution was to just
+ make sure to send an unhold frame during the transfer since it
+ wouldn't make sense to have MOH playing after a transfer anyway.
+ This only modifies SIP transfers, but the other transfers did not
+ seem to be a problem. If DTMF based transfers were a problem it
+ might be okay to add ast_moh_stop to finishup, but I didn't want
+ to have to add that unless required. ABE-2624 ........
+ ................ ................
+
+2010-12-01 17:03 +0000 [r296952-296993] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, include/asterisk/frame.h: Merged revisions 296992 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r296992 | tilghman | 2010-12-01 11:01:56 -0600
+ (Wed, 01 Dec 2010) | 19 lines Merged revisions 296991 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296991 | tilghman | 2010-12-01 11:01:00 -0600
+ (Wed, 01 Dec 2010) | 12 lines Merged revisions 296990 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01 Dec 2010)
+ | 5 lines Clarify documentation on how we store codec preference
+ lists. (closes issue #18397) Reported by: birgita ........
+ ................ ................
+
+ * /, channels/chan_iax2.c: Merged revisions 296951 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r296951 | tilghman | 2010-11-30 19:46:32 -0600
+ (Tue, 30 Nov 2010) | 9 lines Merged revisions 296950 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r296950 | tilghman | 2010-11-30 19:38:19 -0600 (Tue, 30
+ Nov 2010) | 2 lines Missed initializations caused startup errors
+ on Mac OS X (and possibly others, too). ........ ................
+
+2010-12-01 00:28 +0000 [r296871] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 296870 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r296870 | jpeeler | 2010-11-30 18:28:16 -0600
+ (Tue, 30 Nov 2010) | 18 lines Merged revisions 296869 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296869 | jpeeler | 2010-11-30 18:24:58 -0600
+ (Tue, 30 Nov 2010) | 11 lines Merged revisions 296868 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010)
+ | 4 lines Properly restore backup information file when hanging
+ up during message prepending. ABE-2654 ........ ................
+ ................
+
+2010-11-30 22:32 +0000 [r296788-296826] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * include/asterisk/frame.h: Add a comment on why the reserved bit
+ is reserved. Came up when reviewing discussion on the CODEC PREFS
+ IE in IAX2.
+
+ * /, apps/app_meetme.c: Merged revisions 296787 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r296787 | tilghman | 2010-11-30 13:12:48 -0600 (Tue, 30 Nov 2010)
+ | 2 lines DOC: Conference number can be omitted; if omitted, all
+ users in a meetme are listed. ........
+
+2010-11-30 09:49 +0000 [r296752] Stefan Schmidt <sst@sil.at>
+
+ * include/asterisk.h, main/pbx.c, main/asterisk.c: move devices
+ from hints into an ao2_container by splitting up devices from
+ hints into an own ao2_container the callback to get these devices
+ for statechange handling is faster. with this changes the length
+ of a device used in a hint isnt longer restricted to 80
+ characters. Tests showed that calling handle_statechange is 40
+ times faster if no hints are used and 25 times faster if there
+ are any hints. (closes issue #17928) Reported by: mdu113 Tested
+ by: schmidts Review: https://reviewboard.asterisk.org/r/1003/
+
+2010-11-29 23:07 +0000 [r296674] Paul Belanger <pabelanger@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 296673 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r296673 | pabelanger | 2010-11-29 18:05:45 -0500
+ (Mon, 29 Nov 2010) | 19 lines Merged revisions 296671 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296671 | pabelanger | 2010-11-29 17:54:14 -0500
+ (Mon, 29 Nov 2010) | 12 lines Merged revisions 296670 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon, 29 Nov
+ 2010) | 5 lines Make sure nothing else is needed before
+ destroying the scheduler. (closes issue #18398) Reported by:
+ pabelanger ........ ................ ................
+
+2010-11-29 21:31 +0000 [r296630] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 296628 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r296628 | russell | 2010-11-29 15:26:44 -0600 (Mon, 29 Nov 2010)
+ | 6 lines Complete some error handling in transmit_publish() in
+ chan_sip.c. This error handling block caught my eye. It was
+ missing a couple of things, but it should be safe now. Thanks to
+ mmichelson for the quick peer review on IRC. ........
+
+2010-11-29 20:54 +0000 [r296585] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
+ Merged revisions 296582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r296582 | rmudgett | 2010-11-29 14:46:03 -0600
+ (Mon, 29 Nov 2010) | 24 lines Merged revision 296575 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r296575 | rmudgett | 2010-11-29 14:27:37 -0600 (Mon,
+ 29 Nov 2010) | 13 lines Invalid mISDN PTMP redirecting signaling
+ as TE towards NT. The mISDN PTMP redirection signaling (NOTIFY
+ redirecting number and notification code, SETUP redirecting
+ number) is also sent in PTMP/TE mode. It should only apply in
+ PTMP/NT mode. The call setup proceeds but the network (Deutsche
+ Telekom) reacts with ugly ISDN STATUS messages. Also don't send
+ the redirecting number ie when PTP is also sending the
+ DivertingLegInformation2 facility. The redirecting number ie is
+ redundant and the network (Deutsche Telekom) complains about it.
+ Patches: abe_2651_v4.patch uploaded by rmudgett (license 664)
+ JIRA ABE-2651 JIRA SWP-2537 .......... ................
+
+2010-11-29 07:30 +0000 [r296535] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/asterisk.c: Merged revisions 296534 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r296534 | tilghman | 2010-11-29 01:28:44 -0600
+ (Mon, 29 Nov 2010) | 20 lines Merged revisions 296533 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r296533 | tilghman | 2010-11-29 01:27:09 -0600 (Mon, 29 Nov 2010)
+ | 13 lines I love standards. There are so many to choose from.
+ Except when there isn't one. Linux and *BSD disagree on the
+ elements within the ucred structure. Detect which one is in use
+ on the system. (closes issue #18384) Reported by: bjm Patches:
+ cred-diffs uploaded by bjm (license 473)
+ 20101127__issue18384__1.6.2.diff.txt uploaded by tilghman
+ (license 14) 20101127__issue18384__1.8.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman, bjm ........
+ ................
+
+2010-11-27 10:41 +0000 [r296430-296468] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, apps/app_meetme.c: Merged revisions 296467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r296467 | tilghman | 2010-11-27 04:40:22 -0600
+ (Sat, 27 Nov 2010) | 12 lines Merged revisions 296466 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r296466 | tilghman | 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010)
+ | 5 lines 18 characters is too short for most date/times (20 is
+ the usual, but we add more in case of greater precision). (closes
+ issue #18369) Reported by: tnakonz ........ ................
+
+ * include/asterisk.h, /: Merged revisions 296429 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r296429 | tilghman | 2010-11-27 03:58:57 -0600 (Sat, 27 Nov 2010)
+ | 5 lines Also don't build DEBUG_FD_LEAKS when STANDALONE2 is
+ defined. (closes issue #18385) Reported by: cmaj ........
+
+2010-11-26 22:02 +0000 [r296393] Olle Johansson <oej@edvina.net>
+
+ * /, main/say.c: Merged revisions 296391 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r296391 | oej | 2010-11-26 22:37:21 +0100 (Fre,
+ 26 Nov 2010) | 24 lines Merged revisions 296351 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296351 | oej | 2010-11-26 13:23:03 +0100 (Fre,
+ 26 Nov 2010) | 17 lines Merged revisions 296309 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11
+ lines Fix bugs in saying numbers using the Swedish language
+ syntax (closes issue #18355) Reported by: oej Patch by: oej Much
+ help from Peter Lindahl. Testing by the ClearIT team during a
+ coffee break. Review: https://reviewboard.asterisk.org/r/1033/
+ ........ ................ ................
+
+2010-11-26 18:31 +0000 [r296353-296355] Brad Watkins <Marquis42@gmail.com>
+
+ * /, res/res_jabber.c: Merged revisions 296354 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r296354 | marquis | 2010-11-26 13:31:17 -0500 (Fri, 26 Nov 2010)
+ | 12 lines Fix XMPP PubSub-based distributed device state.
+ Initialize pubsubflags to 0 so res_jabber doesn't think there is
+ already an XMPP connection sending device state. Also clean up
+ CLI commands a bit. (closes issue #18272) Reported by: klaus3000
+ Patches: res_jabber_fix_pubsubflags_and_CLI-patch.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000, Marquis Review:
+ https://reviewboard.asterisk.org/r/1030/ ........
+
+ * /, channels/chan_sip.c: Merged revisions 296352 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r296352 | marquis | 2010-11-26 13:19:02 -0500 (Fri, 26 Nov 2010)
+ | 12 lines Fix reloading of peer when a user is requested.
+ Prevent peer reloading from causing multiple MWI subscriptions to
+ be created when using realtime. This had the effect of sending
+ one NOTIFY for every time a sip peer made a call, in one case
+ eventually overwhelming the phone and causing it to reboot.
+ (closes issue #18342) Reported by: nivek Patches:
+ issue0018342p1.patch uploaded by nivek (license 636) Tested by:
+ nivek Review: https://reviewboard.asterisk.org/r/1029/ ........
+
+2010-11-24 23:46 +0000 [r296249] Andrew Parisio <parisioa@gmail.com>
+
+ * apps/app_meetme.c, CHANGES: Meetme use voicemail greet for
+ join/leave announce Added option v(mailbox@[context]) which tells
+ MeetMe where to look for a users greet file. If one does not
+ exist it clears the v option and defers to the functionality of
+ i/I as/if set by the MeetMe() command. Review:
+ https://reviewboard.asterisk.org/r/1009/ (closes issue #18297)
+ Reported by: parisioa Patches: meetme_final_patch_v.diff uploaded
+ by parisioa (license 1153)
+
+2010-11-24 23:30 +0000 [r296235] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 296230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r296230 | russell | 2010-11-24 17:29:44 -0600
+ (Wed, 24 Nov 2010) | 20 lines Merged revisions 296221 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296221 | russell | 2010-11-24 17:28:19 -0600
+ (Wed, 24 Nov 2010) | 13 lines Merged revisions 296213 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010)
+ | 6 lines Make Asterisk less crashy. Since we might not put a new
+ translation path on the channel, go ahead and set it to NULL
+ right after destroying the old one to ensure we don't try to free
+ an invalid translation path later on. ........ ................
+ ................
+
+2010-11-24 22:52 +0000 [r296168] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ /, channels/sig_analog.h: Merged revisions 296167 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r296167 | rmudgett | 2010-11-24 16:49:48 -0600
+ (Wed, 24 Nov 2010) | 57 lines Merged revisions 296166 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296166 | rmudgett | 2010-11-24 16:42:45 -0600
+ (Wed, 24 Nov 2010) | 50 lines Merged revisions 296165 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010)
+ | 43 lines Oneway audio to SIP phone from FXS port after FXS port
+ gets a CallWaiting pip. The FXS connected phone has to have
+ CW/CID support to fail, as it will send back a DTMF 'A' or 'D'
+ when it's ready to receive CallerID. A normal phone with no CID
+ never fails. Also the SIP phone does not hear MOH when the CW
+ call is answered. The DTMF end frame is suppressed when the phone
+ acknowledges the CW signal for CID. The problem is the DTMF begin
+ frame needs to be suppressed as well. The DTMF begin frame is
+ causing SIP to start sending the DTMF RTP frames. Since the DTMF
+ end frame is suppressed, SIP will not stop sending those DTMF RTP
+ packets. * Suppress the DTMF begin and end frames when the
+ channel driver is looking for DTMF digits. * Fixed a couple
+ issues caused by not cleaning up the CID spill if you answer the
+ CW call while it is sending the CID spill. * Fixed not sending
+ CW/CID spill to the phone when the call is natively bridged.
+ (Fixed by not using native bridge if CW/CID is possible.) *
+ Suppress received audio when sending CW/CID spills. The other
+ parties involved do not need to hear the CW/CID spills and may be
+ confused if the CW call is for them. (closes issue #18129)
+ Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch
+ uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+ NOTE: * v1.4 does not have the main problem fixed by suppressing
+ the DTMF start frames. The other three items fixed are relevant.
+ * If you really must restore native bridging between analog
+ ports, you need to disable CW/CID either by configuring
+ chan_dahdi.conf callwaitingcallerid=no or dialing *70 before
+ dialing the number to temporarily disable CW. ........
+ ................ ................
+
+2010-11-24 20:24 +0000 [r296034-296085] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 296084 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r296084 | russell | 2010-11-24 14:23:46 -0600
+ (Wed, 24 Nov 2010) | 26 lines Merged revisions 296083 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296083 | russell | 2010-11-24 14:23:11 -0600
+ (Wed, 24 Nov 2010) | 19 lines Merged revisions 296082 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010)
+ | 12 lines Fix false reporting of an error by set_format(). In
+ the case that the native format was able to be changed to match
+ the new requested format, the code proceeded to attempt to build
+ a translation path, anyway. The result would be NULL, since no
+ translation path is necessary and resulted in this function
+ thinking an error has occurred. This case is now specifically
+ caught and no attempt to build a translation path is attempted.
+ Thanks to our automated tests and bamboo.asterisk.org for
+ catching this problem and making a whole lot of noise when things
+ started failing. :-) ........ ................ ................
+
+ * apps/app_dial.c, main/channel.c, /: Merged revisions 296002 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r296002 | russell | 2010-11-24 11:13:08 -0600
+ (Wed, 24 Nov 2010) | 52 lines Merged revisions 296001 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296001 | russell | 2010-11-24 11:03:16 -0600
+ (Wed, 24 Nov 2010) | 45 lines Merged revisions 296000 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010)
+ | 38 lines Handle failures building translation paths more
+ effectively. The problem scenario occurred on a heavily loaded
+ system that was using the codec_dahdi module and exceeded the
+ hardware transcoding capacity. The failure mode at that point was
+ not good. The report came in to us as an Asterisk lock-up. The
+ "core show locks" shows a ton of threads locked up (but no
+ obvious deadlock). Upon deeper investigation, when the system is
+ in this state, the CPU was maxed out. The CPU was being consumed
+ by the Asterisk logger spewing messages on every audio frame for
+ calls set up after transcoder capacity was reached. The purpose
+ of this patch is to make Asterisk handle failures to create a
+ translation path in a more graceful manner. If we can't
+ translate, then the call just needs to be dropped, as it's not
+ going to work. These are the changes: 1) In set_format() of
+ channel.c (which is called by set_read_format() and
+ set_write_format()), it was ignoring if
+ ast_translator_build_path() failed and returned NULL. It now pays
+ attention to that case and returns a result reflecting failure.
+ With this change in place, the bridging code will immediately
+ detect a failure and end the bridge instead of proceeding to try
+ to bridge frames that can't be translated and making channel
+ drivers freak out by sending them frames in a format they weren't
+ expecting. 2) In ast_indicate_data() of channel.c, failure of
+ ast_playtones_start() was ignored. It is now reflected in the
+ return value of the function. This didn't turn out to have any
+ affect on the bug, but seemed like a good change to leave in. 3)
+ In app_dial(), when only sending a call to a single endpoint, it
+ will attempt to do some bridging of its own of early audio. It
+ uses make_compatible() when it's going to do this. However, it
+ ignored failure from make compatible. So, even with the fix from
+ #1, if there was early audio going through app_dial, there would
+ still be a period of invalid frames passing through. After
+ detecting failure here, Dial() exits. ABE-2658 ........
+ ................ ................
+
+2010-11-23 10:34 +0000 [r295950] Olle Johansson <oej@edvina.net>
+
+ * /, main/say.c: Merged revisions 295949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r295949 | oej | 2010-11-23 11:30:05 +0100 (Tis,
+ 23 Nov 2010) | 21 lines Merged revisions 295907 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r295907 | oej | 2010-11-23 10:36:38 +0100 (Tis,
+ 23 Nov 2010) | 14 lines Merged revisions 295906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8
+ lines Fix support of saynumber(1,n) in the Swedish language
+ (closes issue #18353) Reported by: oej Review:
+ https://reviewboard.asterisk.org/r/1031/ ........
+ ................ ................
+
+2010-11-22 20:05 +0000 [r295870] Sean Bright <sean@malleable.com>
+
+ * configs/chan_dahdi.conf.sample, /: Merged revisions 295869 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r295869 | seanbright | 2010-11-22 15:03:49 -0500
+ (Mon, 22 Nov 2010) | 9 lines Merged revisions 295868 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r295868 | seanbright | 2010-11-22 15:02:37 -0500 (Mon,
+ 22 Nov 2010) | 2 lines Change some documentation to suggest
+ dahdi_monitor instead of ztmonitor. ........ ................
+
+2010-11-22 19:42 +0000 [r295867] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, main/pbx.c, /, apps/app_macro.c,
+ include/asterisk/channel.h, include/asterisk/frame.h: Merged
+ revisions 295866 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r295866 | rmudgett | 2010-11-22 13:36:10 -0600
+ (Mon, 22 Nov 2010) | 60 lines Merged revisions 295843 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600
+ (Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010)
+ | 46 lines The channel redirect function (CLI or AMI) hangs up
+ the call instead of redirecting the call. To recreate the
+ problem: 1) Party A calls Party B 2) Invoke CLI "channel
+ redirect" command to redirect channel call leg associated with A.
+ 3) All associated channels are hung up. Note that if the CLI
+ command were done on the channel call leg associated with B it
+ works. This regression was a result of the fix for issue #16946
+ (https://reviewboard.asterisk.org/r/740/). The regression affects
+ all features that use an async goto to execute the dialplan
+ because of an external event: Channel redirect, AMI redirect, SIP
+ REFER, and FAX detection. The struct ast_channel._softhangup code
+ is a mess. The variable is used for several purposes that do not
+ necessarily result in the call being hung up. I have added
+ doxygen comments to describe how the various _softhangup bits are
+ used. I have corrected all the places where the variable was
+ tested in a non-bit oriented manner. The primary fix is the new
+ AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so
+ the soft hangup requests that do not normally result in a hangup
+ do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171)
+ Reported by: SantaFox (closes issue #18185) Reported by:
+ kwemheuer (closes issue #18211) Reported by: zahir_koradia
+ (closes issue #18230) Reported by: vmarrone (closes issue #18299)
+ Reported by: mbrevda (closes issue #18322) Reported by: nerbos
+ Review: https://reviewboard.asterisk.org/r/1013/ ........
+ ................ ................
+
+2010-11-22 18:43 +0000 [r295789] Erin Spiceland <erin@thespicelands.com>
+
+ * res/res_agi.c: Revert to the previous behavior of AGI command
+ WAIT FOR DIGIT, since the behavior of the command with this patch
+ is almost exactly like that of GET DATA.
+
+2010-11-20 03:13 +0000 [r295748] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Merged revisions 295747 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r295747 | rmudgett | 2010-11-19 21:11:15 -0600 (Fri, 19 Nov 2010)
+ | 13 lines One way audio before answering call waiting call on
+ analog port. * Analog call waiting Caller ID spills could get
+ stuck resulting in one way audio until the waiting call is
+ answered. This only happens on the second (and later) call
+ waiting call if the active call is not the first call. * The
+ CLI/AMI "dahdi show channel" command could report the wrong
+ channel information. Must keep the struct analog_pvt.owner and
+ struct dahdi_pvt.owner pointer in sync. ........
+
+2010-11-20 00:52 +0000 [r295712] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/event.h, /, main/event.c: Merged revisions
+ 295711 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r295711 | russell | 2010-11-19 18:50:00 -0600
+ (Fri, 19 Nov 2010) | 36 lines Merged revisions 295710 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010)
+ | 29 lines Fix cache of device state changes for multiple
+ servers. This patch addresses a regression where device states
+ across multiple servers were not being processing completely
+ correctly. The code works to determine the overall state by
+ looking at the last known state of a device on each server.
+ However, there was a regression due to some invasive rewrites of
+ how the cache works that led to the cache only storing the last
+ device state change for a device, regardless of which server it
+ was on. The code is set up to cache device state change events by
+ ensuring that each event in the cache has a unique device name +
+ entity ID (server ID). The code that was responsible for
+ comparing raw information elements (which EID is) always returned
+ a match due to a memcmp() with a length of 0. There isn't much
+ code to fix the actual bug. This patch also introduces a new CLI
+ command that was very useful for debugging this problem. The
+ command allows you to dump the contents of the event cache.
+ (closes issue #18284) Reported by: klaus3000 Patches:
+ issue18284.rev1.txt uploaded by russell (license 2) Tested by:
+ russell, klaus3000 (closes issue #18280) Reported by: klaus3000
+ Review: https://reviewboard.asterisk.org/r/1012/ ........
+ ................
+
+2010-11-19 22:15 +0000 [r295674] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 295673 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r295673 | twilson | 2010-11-19 14:06:10 -0800
+ (Fri, 19 Nov 2010) | 22 lines Merged revisions 295672 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r295672 | twilson | 2010-11-19 13:55:48 -0800
+ (Fri, 19 Nov 2010) | 15 lines Merged revisions 295628 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010)
+ | 8 lines Discard responses with more than one Via This is not a
+ perfect solution as headers that are joined via commas are not
+ detected. This is a parsing issue that to fix "correctly" would
+ necessitate a new SIP parser. Review:
+ https://reviewboard.asterisk.org/r/1019/ ........
+ ................ ................
+
+2010-11-19 21:42 +0000 [r295671] Brett Bryant <bbryant@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 295670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r295670 | bbryant | 2010-11-19 16:40:21 -0500 (Fri, 19 Nov 2010)
+ | 8 lines Patch for deadlock from ordering issue between
+ channel/queue locks in app_queue (set_queue_variables). (closes
+ issue #18031) Reported by: rain Review:
+ https://reviewboard.asterisk.org/r/1018/ ........
+
+2010-11-19 19:32 +0000 [r295554] Erin Spiceland <erin@thespicelands.com>
+
+ * res/res_agi.c: Add extra functionality to AGI command WAIT FOR
+ DIGIT. Add the ability to play a sound file, listen for more than
+ just one digit, specify escape characters. Backwards compatible
+ (to work with only timeout specified). (closes issue #15531)
+ Reported by: diLLec Patches:
+ asterisk-res_agi-203638-patched.patch uploaded by diLLec (license
+ 839) Tested by: diLLec, espiceland
+
+2010-11-19 16:49 +0000 [r295517] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Merged revisions 295516 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r295516 | rmudgett | 2010-11-19 10:47:11 -0600 (Fri, 19 Nov 2010)
+ | 13 lines Bring sig_analog extraction more into alignment with
+ orig-trunk/v1.6.2 chan_dahdi. * Restore SMDI support. * Fixed
+ initial value of struct analog_pvt.use_callerid. It may get
+ forced on depending upon other config options. * Call
+ analog_dnd() instead of manual inlined code. * Removed unused
+ struct analog_pvt.usedistinctiveringdetection. * Removed the
+ struct analog_pvt.unknown_alarm flag. It was really the struct
+ analog_pvt.inalarm flag. * Use ast_debug() instead of
+ ast_log(LOG_DEBUG). * Rename several function's index variable to
+ idx. * Some formatting tweaks. ........
+
+2010-11-18 20:31 +0000 [r295478] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip_notify.conf.sample, /: Merged revisions 295477 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r295477 | lmadsen | 2010-11-18 14:30:35 -0600 (Thu, 18 Nov 2010)
+ | 6 lines 'sip notify clear-mwi' needs terminating CRLF. (closes
+ issue #18275) Reported by: klaus3000 Patches:
+ fix_body_CRLF_patch.txt uploaded by klaus3000 (license 65)
+ ........
+
+2010-11-18 18:08 +0000 [r295364-295442] Paul Belanger <pabelanger@digium.com>
+
+ * /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions
+ 295441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r295441 | pabelanger | 2010-11-18 13:02:12 -0500
+ (Thu, 18 Nov 2010) | 11 lines Merged revisions 295440 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r295440 | pabelanger | 2010-11-18 12:51:34 -0500 (Thu, 18 Nov
+ 2010) | 4 lines Fix compiler warnings when using openssl-dev
+ 1.0.0+ Review: https://reviewboard.asterisk.org/r/1016/ ........
+ ................
+
+ * /, contrib/scripts/install_prereq: Merged revisions 295404 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r295404 | pabelanger | 2010-11-18 00:12:05 -0500 (Thu, 18 Nov
+ 2010) | 2 lines Add RedHat specific dependencies ........
+
+ * /, configs/res_curl.conf.sample: Merged revisions 295361 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r295361 | pabelanger | 2010-11-17 09:09:38 -0500 (Wed, 17 Nov
+ 2010) | 2 lines Uncomment settings under [global], to surpress
+ warning when loading Asterisk. ........
+
+2010-11-16 23:04 +0000 [r295283] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Merged revisions 295282 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r295282 | rmudgett | 2010-11-16 17:02:36 -0600
+ (Tue, 16 Nov 2010) | 16 lines Merged revisions 295281 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r295281 | rmudgett | 2010-11-16 16:57:07 -0600
+ (Tue, 16 Nov 2010) | 9 lines Merged revisions 295280 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16
+ Nov 2010) | 1 line Dead code elimination in
+ channel.c:ast_channel_bridge() variable who. ........
+ ................ ................
+
+2010-11-16 22:41 +0000 [r295125-295279] Russell Bryant <russell@digium.com>
+
+ * /, build_tools/prep_tarball: Merged revisions 295278 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r295278 | russell | 2010-11-16 16:41:11 -0600 (Tue, 16
+ Nov 2010) | 2 lines Check for pdftotext and give a useful error
+ if not found. ........
+
+ * /, build_tools/prep_tarball: Merged revisions 295201 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r295201 | russell | 2010-11-16 15:46:18 -0600 (Tue, 16
+ Nov 2010) | 2 lines Remove intentional typo I had added when
+ testing the check. oops. ........
+
+ * /, build_tools/prep_tarball: Merged revisions 295164 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r295164 | russell | 2010-11-16 14:50:03 -0600 (Tue, 16
+ Nov 2010) | 2 lines Check for wikiexport.py in PATH and give a
+ useful error message if not found. ........
+
+ * main/app.c: Remove a trailing space. (testing something with
+ bamboo ...)
+
+2010-11-15 19:11 +0000 [r294990-295079] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * tests/test_expr.c (added), /: Merged revisions 295078 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r295078 | tilghman | 2010-11-15 12:30:13 -0600
+ (Mon, 15 Nov 2010) | 16 lines Merged revisions 295062 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r295062 | tilghman | 2010-11-15 12:24:02 -0600
+ (Mon, 15 Nov 2010) | 9 lines Merged revisions 295026 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15
+ Nov 2010) | 2 lines Create test verifying results of expression
+ parser ........ ................ ................
+
+ * funcs/func_curl.c, /: Merged revisions 294989 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r294989 | tilghman | 2010-11-15 01:44:38 -0600
+ (Mon, 15 Nov 2010) | 15 lines Merged revisions 294988 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294988 | tilghman | 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010)
+ | 8 lines It is possible to crash Asterisk by feeding the curl
+ engine invalid data. (closes issue #18161) Reported by: wdoekes
+ Patches: 20101029__issue18161.diff.txt uploaded by tilghman
+ (license 14) Tested by: tilghman ........ ................
+
+2010-11-12 21:15 +0000 [r294907-294912] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 294911 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r294911 | jpeeler | 2010-11-12 15:14:43 -0600
+ (Fri, 12 Nov 2010) | 11 lines Merged revisions 294910 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294910 | jpeeler | 2010-11-12 15:14:23 -0600 (Fri, 12 Nov 2010)
+ | 4 lines Return correct error code if lock path fails. The
+ recent changes to open_mailbox actually caused it to be fixed,
+ but let's be consistent. Reported by alecdavis in asterisk-dev.
+ ........ ................
+
+ * /, apps/app_voicemail.c: Merged revisions 294905 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r294905 | jpeeler | 2010-11-12 14:52:06 -0600
+ (Fri, 12 Nov 2010) | 30 lines Merged revisions 294904 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r294904 | jpeeler | 2010-11-12 14:51:15 -0600
+ (Fri, 12 Nov 2010) | 23 lines Merged revisions 294903 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010)
+ | 16 lines Fix regression causing abort in voicemail after
+ opening a mailbox with no mesgs. In order to be more safe, some
+ error handling code was changed to respect more error conditions
+ including the potential memory allocation failure for deleted and
+ heard message tracking introduced in 293004. However,
+ last_message_index returns -1 for zero messages (perhaps as
+ expected) and was triggering the stricter error checking. Because
+ last_message_index is only called directly in one place, just
+ return 0 from open_mailbox (for file based storage) when no
+ messages are detected unless a real error has occurred. (closes
+ issue #18240) Reported by: leobrown Patches:
+ bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
+ Tested by: pabelanger ........ ................ ................
+
+2010-11-12 02:46 +0000 [r294824] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, /, channels/sig_pri.c: Merged revisions
+ 294823 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r294823 | rmudgett | 2010-11-11 20:45:22 -0600
+ (Thu, 11 Nov 2010) | 25 lines Merged revisions 294822 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r294822 | rmudgett | 2010-11-11 20:44:12 -0600
+ (Thu, 11 Nov 2010) | 18 lines Merged revisions 294821 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010)
+ | 11 lines Asterisk is getting a "No D-channels available!"
+ warning message every 4 seconds. Asterisk is just whining too
+ much with this message: "No D-channels available! Using Primary
+ channel XXX as D-channel anyway!". Filtered the message so it
+ only comes out once if there is no D channel available without an
+ intervening D channel available period. (closes issue #17270)
+ Reported by: jmls ........ ................ ................
+
+2010-11-11 22:18 +0000 [r294741-294749] Russell Bryant <russell@digium.com>
+
+ * /, doc/CCSS_architecture.pdf (removed): Merged revisions 294745
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r294745 | russell | 2010-11-11 16:17:57 -0600 (Thu, 11 Nov 2010)
+ | 6 lines Remove CCSS architecture PDF. It has been moved to:
+ https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture
+ ........
+
+ * doc/CODING-GUIDELINES (removed), doc/ss7.txt (removed), /,
+ doc/backtrace.txt (removed), doc/India-CID.txt (removed),
+ doc/digium-mib.txt (removed), doc/followme.txt (removed),
+ doc/building_queues.txt (removed), doc/timing.txt (removed),
+ doc/advice_of_charge.txt (removed), doc/unistim.txt (removed),
+ doc/video_console.txt (removed), doc/macroexclusive.txt
+ (removed), doc/google-soc2009-ideas.txt (removed), doc/README.txt
+ (added), doc/callfiles.txt (removed), build_tools/prep_tarball,
+ doc/codec-64bit.txt (removed), doc/externalivr.txt (removed),
+ doc/video.txt (removed), doc/jingle.txt (removed),
+ doc/modules.txt (removed), doc/manager_1_1.txt (removed),
+ doc/PEERING (removed), doc/snmp.txt (removed), doc/siptls.txt
+ (removed), doc/HOWTO_collect_debug_information.txt (removed),
+ doc/ldap.txt (removed), doc/sip-retransmit.txt (removed),
+ doc/distributed_devstate.txt (removed),
+ doc/voicemail_odbc_postgresql.txt (removed), doc/tex (removed),
+ doc/queue.txt (removed), doc/jabber.txt (removed),
+ doc/chan_sip-perf-testing.txt (removed), doc/asterisk-mib.txt
+ (removed), Makefile, doc/database_transactions.txt (removed),
+ doc/smdi.txt (removed), doc/janitor-projects.txt (removed),
+ doc/hoard.txt (removed), doc/res_config_sqlite.txt (removed),
+ doc/osp.txt (removed), doc/speechrec.txt (removed), doc/sms.txt
+ (removed), doc/distributed_devstate-XMPP.txt (removed),
+ doc/valgrind.txt (removed), doc/realtimetext.txt (removed),
+ doc/cli.txt (removed), doc/rtp-packetization.txt (removed),
+ doc/datastores.txt (removed): Merged revisions 294740 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r294740 | russell | 2010-11-11 16:13:38 -0600 (Thu, 11 Nov 2010)
+ | 11 lines Remove most of the contents of the doc dir in favor of
+ the wiki content. This merge does the following things: * Removes
+ most of the contents from the doc/ directory in favor of the wiki
+ - http://wiki.asterisk.org/ * Updates the
+ build_tools/prep_tarball script to know how to export the
+ contents of the wiki in both PDF and plain text formats so that
+ the documentation is still included in Asterisk release tarballs.
+ ........
+
+2010-11-11 22:01 +0000 [r294735] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 294734 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r294734 | jpeeler | 2010-11-11 15:58:25 -0600
+ (Thu, 11 Nov 2010) | 32 lines Merged revisions 294733 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r294733 | jpeeler | 2010-11-11 15:57:22 -0600
+ (Thu, 11 Nov 2010) | 25 lines Merged revisions 294688 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010)
+ | 18 lines Fix problem with qualify option packets for realtime
+ peers never stopping. The option packets not only never stopped,
+ but if a realtime peer was not in the peer list multiple options
+ dialogs could accumulate over time. This scenario has the
+ potential to progress to the point of saturating a link just from
+ options packets. The fix was to ensure that the poke scheduler
+ checks to see if a peer is in the peer list before continuing to
+ poke. The reason a peer must be in the peer list to be able to
+ properly manage an options dialog is because otherwise the call
+ pointer is lost when the peer is regenerated from the database,
+ which is how existing qualify dialogs are detected. (closes issue
+ #16382) (closes issue #17779) Reported by: lftsy Patches:
+ bug16382-3.patch uploaded by jpeeler (license 325) Tested by:
+ zerohalo ........ ................ ................
+
+2010-11-10 23:27 +0000 [r294570-294606] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * pbx/pbx_spool.c, /: Merged revisions 294605 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r294605 | tilghman | 2010-11-10 17:26:39 -0600 (Wed, 10 Nov 2010)
+ | 2 lines Fixing the Mac OS X build (bamboo warning) ........
+
+ * pbx/pbx_spool.c, /: Merged revisions 294569 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r294569 | tilghman | 2010-11-10 17:13:37 -0600 (Wed, 10 Nov 2010)
+ | 8 lines Properly queue files with inotify(7). (closes issue
+ #18089) Reported by: abelbeck Patches:
+ 20101021__issue18089.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........
+
+2010-11-10 14:15 +0000 [r294502-294536] Russell Bryant <russell@digium.com>
+
+ * /, res/ais/clm.c, res/ais/evt.c, UPGRADE-1.8.txt: Merged
+ revisions 294535 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r294535 | russell | 2010-11-10 08:14:51 -0600 (Wed, 10 Nov 2010)
+ | 5 lines Tweak a couple of CLI commands back to their original
+ form. The "module" in this case is two parts, so there are two
+ words before the verb of the CLI command. ........
+
+ * /, main/devicestate.c: Merged revisions 294501 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r294501 | russell | 2010-11-10 06:46:27 -0600
+ (Wed, 10 Nov 2010) | 14 lines Merged revisions 294500 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294500 | russell | 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010)
+ | 7 lines Improve a debug message to be more readable and
+ consistent. (closes issue #18282) Reported by: klaus3000 Patches:
+ ast_devstate2str-patch.txt uploaded by klaus3000 (license 65)
+ ........ ................
+
+2010-11-09 22:52 +0000 [r294467] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Merged revisions 294466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r294466 | rmudgett | 2010-11-09 16:46:45 -0600 (Tue, 09 Nov 2010)
+ | 1 line Allow ast_do_masquerade() failure to be reported again.
+ ........
+
+2010-11-09 20:35 +0000 [r294431] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 294430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r294430 | tilghman | 2010-11-09 14:33:05 -0600
+ (Tue, 09 Nov 2010) | 15 lines Merged revisions 294429 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010)
+ | 8 lines Detect GMime properly on systems where gmime flags and
+ libs are configured with pkg-config. (closes issue #16155)
+ Reported by: jcollie Patches: 20100917__issue16155.diff.txt
+ uploaded by tilghman (license 14) Tested by: tilghman ........
+ ................
+
+2010-11-09 17:00 +0000 [r294351] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, channels/chan_misdn.c, channels/sig_analog.c, /,
+ include/asterisk/channel.h, channels/sig_pri.c: Merged revisions
+ 294349 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r294349 | rmudgett | 2010-11-09 10:55:32 -0600 (Tue, 09 Nov 2010)
+ | 17 lines Analog lines do not transfer CONNECTED LINE or execute
+ the interception macros. Add connected line update for sig_analog
+ transfers and simplify the corresponding sig_pri and chan_misdn
+ transfer code. Note that if you create a three-way call in
+ sig_analog before transferring the call, the distinction of the
+ caller/callee interception macros make little sense. The
+ interception macro writer needs to be prepared for either
+ caller/callee macro to be executed. The current implementation
+ swaps which caller/callee interception macro is executed after a
+ three-way call is created. Review:
+ https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA
+ SWP-2372 ........
+
+2010-11-08 22:33 +0000 [r294279-294314] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_timing_timerfd.c: Merged revisions 294313 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r294313 | jpeeler | 2010-11-08 16:32:13 -0600
+ (Mon, 08 Nov 2010) | 9 lines Merged revisions 294312 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r294312 | jpeeler | 2010-11-08 16:30:49 -0600 (Mon, 08
+ Nov 2010) | 1 line add missing unlock not present in 294277
+ ........ ................
+
+ * main/channel.c, /, res/res_timing_timerfd.c,
+ include/asterisk/timing.h, main/timing.c: Merged revisions 294278
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r294278 | jpeeler | 2010-11-08 15:59:45 -0600
+ (Mon, 08 Nov 2010) | 23 lines Merged revisions 294277 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010)
+ | 16 lines Fix playback failure when using IAX with the timerfd
+ module. To fix this issue the alert pipe will now be used when
+ the timerfd module is in use. There appeared to be a race that
+ was not solved by adding locking in the timerfd module, but
+ needed to be there anyway. The race was between the timer being
+ put in non-continuous mode in ast_read on the channel thread and
+ the IAX frame scheduler queuing a frame which would enable
+ continuous mode before the non-continuous mode event was read.
+ This race for now is simply avoided. (closes issue #18110)
+ Reported by: tpanton Tested by: tpanton I put tested by tpanton
+ because it was tested on his hardware. Thanks for the remote
+ access to debug this issue! ........ ................
+
+2010-11-08 21:04 +0000 [r294244] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 294243 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r294243 | mnicholson | 2010-11-08 14:56:30 -0600
+ (Mon, 08 Nov 2010) | 15 lines Merged revisions 294242 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov
+ 2010) | 8 lines Go off hold when we get an empty reinvite telling
+ us to. (closes issue 0014448) Reported by: frawd (closes issue
+ #17878) Reported by: frawd ........ ................
+
+2010-11-08 19:59 +0000 [r294208] Terry Wilson <twilson@digium.com>
+
+ * /, configs/calendar.conf.sample, res/res_calendar.c: Merged
+ revisions 294207 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r294207 | twilson | 2010-11-08 13:56:10 -0600 (Mon, 08 Nov 2010)
+ | 2 lines Set a default waittime, and make sure to convert it to
+ milliseconds ........
+
+2010-11-08 17:19 +0000 [r294127] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 294125 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08
+ Nov 2010) | 33 lines valgrind reported references to freed memory
+ during a mISDN hangup collision. Bad things have been happening
+ in chan_misdn because the chan_misdn channel private struct
+ chan_list is not protected from reentrancy. Hangup collisions
+ have be causing read and write accesses to freed memory.
+ Converted chan_misdn struct chan_list to an ao2 object for its
+ reference counting feature. ********** Removed an impediment to
+ converting chan_list to an ao2 object. The use of the other_ch
+ member in chan_list is shaky at best. It is set if the incoming
+ and outgoing call legs are mISDN. The use of the other_ch member
+ goes against the Asterisk architecture and can even cause
+ problems. 1) It is used to disable echo cancellation. This could
+ be bad if the call is forked and the winning call leg is not
+ mISDN or the winning call leg is not the last mISDN channel
+ called by the fork. The other_ch would become a dangling pointer.
+ 2) It is used when the far end is alerting to hear the far end's
+ inband audio instead of Asterisk's generated ringback tone. This
+ is bad if the call is forked. You would only hear the last forked
+ mISDN channel and it may not be ringing yet. The other_ch would
+ become a dangling pointer if the call is later transferred.
+ ********** JIRA SWP-2423 JIRA ABE-2614 ........
+
+2010-11-05 22:17 +0000 [r294086] Brett Bryant <bbryant@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 294084 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r294084 | bbryant | 2010-11-05 18:03:11 -0400 (Fri, 05 Nov 2010)
+ | 9 lines Fixed deadlock avoidance issues while locking channel
+ when adding the Max-Forwards header to a request. (closes issue
+ #17949) (closes issue #18200) Reported by: bwg Review:
+ https://reviewboard.asterisk.org/r/997/ ........
+
+2010-11-05 21:56 +0000 [r294083] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Perform proper handling of forked outbound
+ INVITE requests. RFC3261 section 12 about dialog creation says an
+ INVITE transaction results in an established dialog once it
+ receives the 200 OK response. It is possible to receive multiple
+ differing 200 OK responses for a single outbound INVITE Request,
+ and this should result in establishing multiple dialogs. This
+ patch allows for all differing 200 OK responses to an INVITE
+ request to establish a separate dialog, but only the first dialog
+ is kept. All other resulting dialogs from the initial request are
+ immediately ACKed and then immediately terminated with a BYE
+ request. Review: https://reviewboard.asterisk.org/r/946/
+
+2010-11-05 16:07 +0000 [r294048-294050] Terry Wilson <twilson@digium.com>
+
+ * contrib/scripts/ast_tls_cert, /: Merged revisions 294049 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r294049 | twilson | 2010-11-05 09:05:50 -0700 (Fri, 05 Nov 2010)
+ | 2 lines Corret spelling and example ........
+
+ * contrib/scripts/ast_tls_cert, /: Merged revisions 294047 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r294047 | twilson | 2010-11-05 08:36:20 -0700 (Fri, 05 Nov 2010)
+ | 2 lines Tell people to use the correct common name for clients
+ as well ........
+
+2010-11-05 15:26 +0000 [r294046] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 293924 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r293924 | dvossel | 2010-11-04 16:39:51 -0500 (Thu, 04 Nov 2010)
+ | 4 lines Fixes ringback tone on sip semi-attended transfer.
+ ABE-2168 ........
+
+2010-11-05 00:08 +0000 [r293971] Shaun Ruffell <sruffell@digium.com>
+
+ * /, codecs/codec_dahdi.c: Merged revisions 293970 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r293970 | sruffell | 2010-11-04 19:07:11 -0500
+ (Thu, 04 Nov 2010) | 32 lines Merged revisions 293969 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293969 | sruffell | 2010-11-04 19:06:02 -0500
+ (Thu, 04 Nov 2010) | 25 lines Merged revisions 293968 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010)
+ | 17 lines codecs/codec_dahdi: Prevent "choppy" audio when
+ receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically
+ commit 9034) added the capability for the wctc4xxp to return more
+ than a single packet of data in response to a read. However, when
+ decoding packets, codec_dahdi was still assuming that the default
+ number of samples was in each read. In other words, each packet
+ your provider sent you, regardless of size, would result in 20 ms
+ of decoded data (30 ms if decoding G723). If your provider was
+ sending 60 ms packets then codec_dahdi would end up stripping 40
+ ms of data from each transcoded frame resulting in "choppy"
+ audio. This would only affect systems where G729 packets are
+ arriving in sizes greater than 20ms or G723 packets arriving in
+ sizes greater than 30ms. DAHDI-744. ........ ................
+ ................
+
+2010-11-04 13:29 +0000 [r293888] Paul Belanger <pabelanger@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 293887 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r293887 | pabelanger | 2010-11-04 09:27:54 -0400 (Thu, 04 Nov
+ 2010) | 8 lines Do not output port in IPaddress for AMI sippeers.
+ (closes issue #18248) Reported by: orn Patches:
+ ami_sippeers.patch uploaded by pabelanger (license 224) Tested
+ by: orn ........
+
+2010-11-03 18:43 +0000 [r293809] Terry Wilson <twilson@digium.com>
+
+ * main/rtp_engine.c, /, channels/chan_sip.c,
+ include/asterisk/rtp_engine.h: Merged revisions 293803 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r293803 | twilson | 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010)
+ | 25 lines Avoid valgrind warnings for
+ ast_rtp_instance_get_xxx_address The documentation for
+ ast_rtp_instance_get_(local/remote)_address stated that they
+ returned 0 for success and -1 on failure. Instead, they returned
+ 0 if the address structure passed in was already equivalent to
+ the address instance local/remote address or 1 otherwise. 90% of
+ the calls to these functions completely ignored the return
+ address and passed in an uninitialized struct, which would make
+ valgrind complain even though the operation was technically safe.
+ This patch fixes the documentation and converts the
+ get_xxx_address functions to void since all they really do is
+ copy the address and cannot fail. Additionally two new functions
+ (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created
+ for the 3 times where the return value was actually checked. The
+ get_and_cmp_local_address function is currently unused, but
+ exists for the sake of symmetry. The only functional change as a
+ result of this change is that we will not do an
+ ast_sockaddr_cmp() on (mostly uninitialized) addresses before
+ doing the ast_sockaddr_copy() in the get_*_address functions. So,
+ even though it is an API change, it shouldn't have a noticeable
+ change in behavior. Review:
+ https://reviewboard.asterisk.org/r/995/ ........
+
+2010-11-03 18:38 +0000 [r293808] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 293807 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r293807 | rmudgett | 2010-11-03 13:35:19 -0500
+ (Wed, 03 Nov 2010) | 34 lines Merged revisions 293806 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293806 | rmudgett | 2010-11-03 13:31:57 -0500
+ (Wed, 03 Nov 2010) | 27 lines Merged revisions 293805 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010)
+ | 20 lines Party A in an analog 3-way call would continue to hear
+ ringback after party C answers. All parties are analog FXS ports.
+ 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to
+ bring C into 3-way call before C answers. (A and B hear ringback)
+ 4) C answers 5) A continues to hear ringback during the 3-way
+ call. (All parties can hear each other.) * Fixed use of wrong
+ variable in dahdi_bridge() that stopped ringback on the wrong
+ subchannel. * Made several debug messages have more information.
+ A similar issue happens if B and C are SIP channels. B continues
+ to hear ringback. For some reason this only affects v1.8 and
+ trunk. * Don't start ringback on the real and 3-way subchannels
+ when creating the 3-way conference. Removing this code is benign
+ on v1.6.2 and earlier. ........ ................ ................
+
+2010-11-02 23:10 +0000 [r293725] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 293724 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r293724 | jpeeler | 2010-11-02 18:09:06 -0500
+ (Tue, 02 Nov 2010) | 22 lines Merged revisions 293723 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293723 | jpeeler | 2010-11-02 18:07:13 -0500
+ (Tue, 02 Nov 2010) | 15 lines Merged revisions 293722 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010)
+ | 8 lines Add enabled/disabled information for rtautoclear sip
+ show settings output. When setting to zero/"no", the numeric
+ default was shown making it not obvious the disabled setting was
+ respected. (closes issue #18123) Reported by: zerohalo ........
+ ................ ................
+
+2010-11-02 21:31 +0000 [r293649] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 293648 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r293648 | rmudgett | 2010-11-02 16:29:25 -0500
+ (Tue, 02 Nov 2010) | 20 lines Merged revisions 293647 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293647 | rmudgett | 2010-11-02 16:26:30 -0500
+ (Tue, 02 Nov 2010) | 13 lines Merged revisions 293639 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010)
+ | 6 lines Make warning message have more useful information in
+ it. Change "Unable to get index, and nullok is not asserted" to
+ "Unable to get index for '<channel-name>' on channel <number>
+ (<function>(), line <number>)". ........ ................
+ ................
+
+2010-11-02 20:47 +0000 [r293578-293612] Paul Belanger <pabelanger@digium.com>
+
+ * main/manager.c, /: Merged revisions 293611 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r293611 | pabelanger | 2010-11-02 16:45:09 -0400 (Tue, 02 Nov
+ 2010) | 2 lines If manager and tls are disabled, do not display
+ TCP/TLS Bindaddress. ........
+
+ * configs/gtalk.conf.sample, UPGRADE.txt, channels/chan_gtalk.c,
+ CHANGES: New CLI command 'gtalk show settings'. Review:
+ https://reviewboard.asterisk.org/r/984/
+
+2010-11-02 14:43 +0000 [r293577] Mark Michelson <mmichelson@digium.com>
+
+ * CHANGES: Add to the CHANGES file that the HTTP server supports
+ IPv6 addressing.
+
+2010-11-01 17:32 +0000 [r293531] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Merged revisions 293530 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r293530 | rmudgett | 2010-11-01 12:29:30 -0500 (Mon, 01 Nov 2010)
+ | 10 lines Analog 3-way call would not connect all parties if one
+ was using sig_pri. Also the "dahdi show channel" would not show
+ the correct 3-way call status. * Synchronized the inthreeway flag
+ between chan_dahdi and sig_analog. * Fixed a my_set_linear_mode()
+ sign error and made take an analog sub channel enum. ........
+
+2010-11-01 16:11 +0000 [r293497] Paul Belanger <pabelanger@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 293496 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r293496 | pabelanger | 2010-11-01 12:09:05 -0400 (Mon,
+ 01 Nov 2010) | 13 lines Use ast_sockaddr_from_sin function not
+ memcpy This resolves some IAX2 registration issue report on the
+ asterisk-users mailing list. (closes issue #18202) Reported by:
+ pabelanger Patches: update_registry.patch.v2 uploaded by
+ pabelanger (license 224) Tested by: pabelanger, Nic Colledge
+ (mailing list) Review: https://reviewboard.asterisk.org/r/993
+ ........
+
+2010-10-30 01:55 +0000 [r293342-293419] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 293418 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r293418 | rmudgett | 2010-10-29 20:53:29 -0500
+ (Fri, 29 Oct 2010) | 16 lines Merged revisions 293417 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293417 | rmudgett | 2010-10-29 20:49:15 -0500
+ (Fri, 29 Oct 2010) | 9 lines Merged revisions 293416 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29
+ Oct 2010) | 1 line Remove some more code that serves no purpose.
+ ........ ................ ................
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 293341 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r293341 | rmudgett | 2010-10-29 19:46:41 -0500
+ (Fri, 29 Oct 2010) | 16 lines Merged revisions 293340 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293340 | rmudgett | 2010-10-29 19:40:10 -0500
+ (Fri, 29 Oct 2010) | 9 lines Merged revisions 293339 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29
+ Oct 2010) | 1 line Remove some code that serves no purpose.
+ ........ ................ ................
+
+2010-10-29 21:50 +0000 [r293306] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 293305 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r293305 | jpeeler | 2010-10-29 16:48:38 -0500 (Fri, 29 Oct 2010)
+ | 9 lines Modify sip_setoption to not complain about unknown
+ options. This now behaves just like the other setoption
+ callbacks. For the curious the offending option for the reporter
+ was AST_OPTION_CHANNEL_WRITE which was getting passed due to a
+ fix for chan_local in 286189. (closes issue #17985) Reported by:
+ globalnetinc ........
+
+2010-10-29 20:46 +0000 [r293273] Mark Michelson <mmichelson@digium.com>
+
+ * main/http.c, UPGRADE.txt, configs/http.conf.sample: Enable IPv6
+ for the built-in HTTP server. Review:
+ https://reviewboard.asterisk.org/r/986
+
+2010-10-28 20:01 +0000 [r293198] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/ast_expr2.h, res/ael/ael.tab.c, main/ast_expr2.y,
+ res/ael/ael_lex.c, res/ael/ael.tab.h, main/ast_expr2.c: Merged
+ revisions 293197 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r293197 | tilghman | 2010-10-28 15:00:06 -0500
+ (Thu, 28 Oct 2010) | 33 lines Merged revisions 293195-293196 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293195 | tilghman | 2010-10-28 14:52:52 -0500
+ (Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+ | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+ Reported (though the reporter did not understand he was reporting
+ a bug) on the asterisk-users list:
+ http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+ ........ ................ r293196 | tilghman | 2010-10-28
+ 14:54:34 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions
+ 293194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+ | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+ Reported (though the reporter did not understand he was reporting
+ a bug) on the asterisk-users list:
+ http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+ ........ ................ ................
+
+2010-10-28 16:11 +0000 [r293160] Jeff Peeler <jpeeler@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 293159 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r293159 | jpeeler | 2010-10-28 11:11:08 -0500
+ (Thu, 28 Oct 2010) | 18 lines Merged revisions 293158 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28 Oct 2010)
+ | 11 lines Fix infinite loop in FILTER(). Specifically when
+ you're using characters above \x7f or invalid character escapes
+ (e.g. \xgg). (closes issue #18060) Reported by: wdoekes Patches:
+ issue18060_func_strings_filter_infinite_loop.patch uploaded by
+ wdoekes (license 717) Tested by: wdoekes ........
+ ................
+
+2010-10-26 18:54 +0000 [r293120] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 293119 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r293119 | jpeeler | 2010-10-26 13:49:08 -0500
+ (Tue, 26 Oct 2010) | 43 lines Merged revisions 293118 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293118 | jpeeler | 2010-10-26 13:33:24 -0500
+ (Tue, 26 Oct 2010) | 36 lines Merged revisions 293004 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010)
+ | 29 lines Fix inprocess_container in voicemail to correctly
+ restrict max messages. The comparison function logic was off, so
+ the number of sessions for a given mailbox were not being
+ incremented properly. This problem caused the maximum number of
+ messages per folder to not be respected when simultaneously
+ leaving multiple voicemails just below the threshold. These
+ problems should be fixed by the above, but just in case: Fixed
+ resequence_mailbox to rely on the actual number of detected
+ number of files in a directory rather than just assuming only 10
+ messages more than the maximum had been left. Also if more
+ messages than the maximum are deleted they are actually removed
+ now. The second purpose of this commit should have been separated
+ out probably, but is related to the above. Again, if the number
+ of messages in a given voicemail folder exceeds the maximum set
+ limit make sure to allocate enough space for the deleted and
+ heard index tracking array. A few random fixes: There was a
+ forgotten decrement of the inprocess count in imap_store_file.
+ When using IMAP storage, do not look in the directory where file
+ based storage messages may still reside and influence the message
+ count. Ensure to use only the first format in sendmail. ABE-2516
+ ........ ................ ................
+
+2010-10-26 16:33 +0000 [r293047-293082] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 293081 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r293081 | rmudgett | 2010-10-26 11:32:59 -0500 (Tue, 26 Oct 2010)
+ | 1 line No need to define the struct if there are no users.
+ ........
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sig_pri.c: Merged revisions 293046 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r293046 | rmudgett | 2010-10-26 10:53:58 -0500 (Tue, 26 Oct 2010)
+ | 4 lines Allow the DAHDI driver to compile, even with a
+ sufficiently older version of libpri. Fixes our Bamboo builds.
+ ........
+
+2010-10-25 21:16 +0000 [r292915-292970] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, channels/sig_pri.c: Merged revisions 292969 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292969 | tilghman | 2010-10-25 16:15:19 -0500 (Mon, 25 Oct 2010)
+ | 2 lines Several more defines that need to be altered for
+ compiling against an older version of libpri ........
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sig_pri.c: Merged revisions 292906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292906 | tilghman | 2010-10-25 14:28:35 -0500 (Mon, 25 Oct 2010)
+ | 4 lines Allow the DAHDI driver to compile, even with a
+ sufficiently older version of libpri. Fixes our Bamboo builds.
+ ........
+
+2010-10-25 19:11 +0000 [r292869] David Vossel <dvossel@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 292868 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r292868 | dvossel | 2010-10-25 14:07:50 -0500
+ (Mon, 25 Oct 2010) | 39 lines Merged revisions 292867 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r292867 | dvossel | 2010-10-25 14:06:21 -0500
+ (Mon, 25 Oct 2010) | 32 lines Merged revisions 292866 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010)
+ | 27 lines This patch turns chan_local pvts into astobj2 objects.
+ chan_local does some dangerous things involving deadlock
+ avoidance. tech_pvt functions like hangup and queue_frame are
+ provided with a locked channel upon entry. Those functions are
+ completely safe as long as you don't attempt to give up that
+ channel lock, but that is impossible to guarantee due to the
+ required deadlock avoidance necessary to lock both the tech_pvt
+ and both channels involved. In the past, we have tried to account
+ for this by doing things like setting a "glare" flag that
+ indicates what function should destroy the pvt. This was used in
+ local_hangup and local_queue_frame to decided who should destroy
+ the pvt if they collided in separate threads. I have removed the
+ need to do this by converting all chan_local tech_pvts to
+ astobj2. This means we can ref a pvt before deadlock avoidance
+ and not have to worry about that pvt possibly getting destroyed
+ under us. It also cleans up where we destroy the tech_pvt. The
+ only unlink from the tech_pvt container occurs in local_hangup
+ now, which is where it should occur. Since there still may be
+ thread collisions on some functions like local_hangup after
+ deadlock avoidance, I have added some checks to detect those
+ collisions and exit appropriately. I think this patch is going to
+ solve quite a bit of weirdness we have had with local channels in
+ the past. ........ ................ ................
+
+2010-10-22 22:40 +0000 [r292808-292826] Terry Wilson <twilson@digium.com>
+
+ * contrib/scripts/ast_tls_cert, /: Merged revisions 292825 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292825 | twilson | 2010-10-22 15:35:29 -0700 (Fri, 22 Oct 2010)
+ | 4 lines Don't create directories without at least o+x Also,
+ making files that you are going to modify read-only is dumb.
+ ........
+
+ * contrib/scripts/ast_tls_cert, /: Merged revisions 292794 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292794 | twilson | 2010-10-22 15:18:36 -0700 (Fri, 22 Oct 2010)
+ | 2 lines Make files readable only by the owner ........
+
+2010-10-22 21:29 +0000 [r292788] Leif Madsen <lmadsen@digium.com>
+
+ * /, channels/chan_sip.c, configs/res_ldap.conf.sample,
+ contrib/scripts/asterisk.ldif: Merged revisions 292787 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r292787 | lmadsen | 2010-10-22 16:28:43 -0500
+ (Fri, 22 Oct 2010) | 21 lines Merged revisions 292786 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010)
+ | 13 lines Update the LDIF file for LDAP. The LDIF file
+ asterisk.ldif was quite a bit out of date from the
+ asterisk.ldap-schema file, so I've now updated that to be in
+ sync. The asterisk.ldif file being out of sync was a problem on
+ my systems where I was doing an ldapadd to import the schema into
+ the LDAP database, and the existing file would cause problems and
+ ERROR messages when registering. Additional documention has been
+ added based on feedback in the issue I'm closing. (closes issue
+ #13861) Reported by: scramatte Patches: ldap-update.txt uploaded
+ by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec,
+ rgenthner ........ ................
+
+2010-10-22 17:16 +0000 [r292743] Terry Wilson <twilson@digium.com>
+
+ * contrib/scripts/ast_tls_cert (added), /: Merged revisions 292740
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292740 | twilson | 2010-10-22 09:49:34 -0700 (Fri, 22 Oct 2010)
+ | 45 lines Add TLS cert helper script This script is useful for
+ quickly generating self-signed CA, server, and client
+ certificates for use with Asterisk. It is still recommended to
+ obtain certificates from a recognized Certificate Authority and
+ to develop an understanding how SSL certificates work. Real
+ security is hard work. OPTIONS: -h Show this message -m Type of
+ cert "client" or "server". Defaults to server. -f Config filename
+ (openssl config file format) -c CA cert filename (creates new CA
+ cert/key as ca.crt/ca.key if not passed) -k CA key filename -C
+ Common name (cert field) For a server cert, this should be the
+ same address that clients attempt to connect to. Usually this
+ will be the Fully Qualified Domain Name, but might be the IP of
+ the server. For a CA or client cert, it is merely informational.
+ Make sure your certs have unique common names. -O Org name (cert
+ field) An informational string (company name) -o Output filename
+ base (defaults to asterisk) -d Output directory (defaults to the
+ current directory) Example: To create a CA and a server
+ (pbx.mycompany.com) cert with output in /tmp: ast_tls_cert -C
+ pbx.mycompany.com -O "My Company" -d /tmp This will create a CA
+ cert and key as well as asterisk.pem and the the two files that
+ it is made from: asterisk.crt and asterisk.key. Copy asterisk.pem
+ and ca.crt somewhere (like /etc/asterisk) and set
+ tlscertfile=/etc/asterisk.pem and tlscafile=/etc/ca.crt. Since
+ this is a self-signed key, many devices will require you to
+ import the ca.crt file as a trusted cert. To create a client cert
+ using the CA cert created by the example above: ast_tls_cert -m
+ client -c /tmp/ca.crt -k /tmp/ca.key -C "Joe User" -O \ "My
+ Company" -d /tmp -o joe_user This will create client.crt/key/pem
+ in /tmp. Use this if your device supports a client certificate.
+ Make sure that you have the ca.crt file set up as a tlscafile in
+ the necessary Asterisk configs. Make backups of all .key files in
+ case you need them later. ........
+
+2010-10-22 17:10 +0000 [r292742] Mark Michelson <mmichelson@digium.com>
+
+ * /, tests/test_event.c: Merged revisions 292741 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292741 | mmichelson | 2010-10-22 12:09:52 -0500 (Fri, 22 Oct
+ 2010) | 12 lines Prevent multiple runs of event_sub_test from
+ producing false failure results. The array of test subscriptions
+ was declared "static," meaning that the data.count field would
+ retain its value between runs of the test. After the first test
+ run, this would result in false reports of test failures. I chose
+ to just remove the "static" keyword from the structure since it's
+ not a huge deal to construct this structure during each run of
+ the test. Another alternative would have been to zero out the
+ data.count fields of each test subscription instead. ........
+
+2010-10-22 15:47 +0000 [r292705] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, channels/chan_misdn.c, /, channels/sig_pri.c:
+ Merged revisions 292704 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292704 | rmudgett | 2010-10-22 10:47:08 -0500 (Fri, 22 Oct 2010)
+ | 19 lines Connected line is not updated when chan_dahdi/sig_pri
+ or chan_misdn transfers a call. When a call is transfered by ECT
+ or implicitly by disconnect in sig_pri or implicitly by
+ disconnect in chan_misdn, the connected line information is not
+ exchanged. The connected line interception macros also need to be
+ executed if defined. The CALLER interception macro is executed
+ for the held call. The CALLEE interception macro is executed for
+ the active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches:
+ abe_2589_c3bier.patch uploaded by rmudgett (license 664)
+ abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review:
+ https://reviewboard.asterisk.org/r/958/ ........
+
+2010-10-21 22:11 +0000 [r292668] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, channels/misdn/ie.c: Merged revisions 292667 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292667 | tilghman | 2010-10-21 17:09:25 -0500 (Thu, 21 Oct 2010)
+ | 2 lines Compile correctly on Linux (asterisk/localtime.h
+ depends upon asterisk/autoconfig.h loading first). ........
+
+2010-10-21 18:23 +0000 [r292630] Paul Belanger <pabelanger@digium.com>
+
+ * /, contrib/init.d/rc.suse.asterisk: Merged revisions 292628 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292628 | pabelanger | 2010-10-21 14:13:18 -0400 (Thu, 21 Oct
+ 2010) | 5 lines Fix typo in SUSE init script. Reported by: Dave
+ Cotton on asterisk-users list. ........
+
+2010-10-21 16:46 +0000 [r292597] David Vossel <dvossel@digium.com>
+
+ * main/manager.c, /: Merged revisions 292595 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292595 | dvossel | 2010-10-21 11:14:33 -0500 (Thu, 21 Oct 2010)
+ | 14 lines Fixes recursive lock problem in manager.c It is
+ possible for a AMI session to freeze because of invalid use of
+ recursive locks during the EVENT processing. This patch removes
+ the unnecessary locks. (closes issue #18167) Reported by: sustav
+ Patches: manager_locking_v1.diff uploaded by dvossel (license
+ 671) Tested by: sustav ........
+
+2010-10-21 13:17 +0000 [r292559] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/res_ldap.conf.sample: Merged revisions 292557 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r292557 | lmadsen | 2010-10-21 08:12:19 -0500
+ (Thu, 21 Oct 2010) | 14 lines Merged revisions 292556 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r292556 | lmadsen | 2010-10-21 08:11:52 -0500 (Thu, 21 Oct 2010)
+ | 6 lines Change res_ldap.sample.conf to match the schema.
+ (closes issue #17376) Reported by: jcovert Patches:
+ res_ldap.conf.sample.patch uploaded by jcovert (license 551)
+ ........ ................
+
+2010-10-21 11:38 +0000 [r292524] Russell Bryant <russell@digium.com>
+
+ * /, res/res_config_ldap.c: Merged revisions 292523 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r292523 | russell | 2010-10-21 06:36:47 -0500 (Thu, 21
+ Oct 2010) | 4 lines Add var=value to log message on update
+ failure, and add newline. ... just for you, Leif. ........
+
+2010-10-21 01:03 +0000 [r292490] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 292489 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292489 | rmudgett | 2010-10-20 20:02:50 -0500 (Wed, 20 Oct 2010)
+ | 7 lines Send CONNECT_ACKNOWLEDGE for CIS calls too. The
+ originator of the Q.SIG call completion signaling link was not
+ changed to the active state when the CONNECT message came in. The
+ T309 processing would immediately kill the signaling link because
+ it was not in the active state. ........
+
+2010-10-21 00:23 +0000 [r292414-292443] Paul Belanger <pabelanger@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 292436 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r292436 | pabelanger | 2010-10-20 20:21:59 -0400 (Wed,
+ 20 Oct 2010) | 8 lines Application not properly unregister in
+ voicemail (closes issue #18128) Reported by: junky Patches:
+ vm_unregister.diff uploaded by junky (license 177) Tested by:
+ pabelanger, lmadsen ........
+
+ * apps/app_dial.c, /: Merged revisions 292413 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r292413 | pabelanger | 2010-10-20 20:07:17 -0400
+ (Wed, 20 Oct 2010) | 24 lines Merged revisions 292412 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r292412 | pabelanger | 2010-10-20 20:05:45 -0400
+ (Wed, 20 Oct 2010) | 17 lines Merged revisions 292411 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct
+ 2010) | 10 lines Record priv-recordintro as sln, not gsm This
+ removes the gsm->sln step when transcoding priv-recordintro.
+ (closes issue #18176) Reported by: pabelanger Patches:
+ chan_sip.diff uploaded by pabelanger (license 224) ........
+ ................ ................
+
+2010-10-20 00:41 +0000 [r292377] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, res/res_musiconhold.c: Merged revisions 292376 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r292376 | tilghman | 2010-10-19 19:40:29 -0500 (Tue, 19
+ Oct 2010) | 5 lines Oops. This module uses the generic timer and
+ no longer uses DAHDI. This causes a problem with the Solaris and
+ other system builds that have gcc 4.1 (where optional_api is
+ non-optional). ........
+
+2010-10-19 22:19 +0000 [r292345] Paul Belanger <pabelanger@digium.com>
+
+ * /, contrib/scripts/install_prereq: Merged revisions 292343 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292343 | pabelanger | 2010-10-19 18:14:23 -0400 (Tue, 19 Oct
+ 2010) | 2 lines Add resample and imap_tk dependencies. ........
+
+2010-10-19 19:35 +0000 [r292310] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c, res/res_srtp.c: Merged revisions 292309
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292309 | twilson | 2010-10-19 12:27:32 -0700 (Tue, 19 Oct 2010)
+ | 10 lines Add sip show peer info about crypto and remove dated
+ comment This patch adds information about the encryption setting
+ to 'sip show peers' and removes an out-of-date comment from
+ res_srtp.c and instead directs users to the proper documentation.
+ (closes issue #18140) Reported by: chodorenko ........
+
+2010-10-18 22:14 +0000 [r292231] Leif Madsen <lmadsen@digium.com>
+
+ * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 292225
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r292225 | lmadsen | 2010-10-18 16:51:23 -0500
+ (Mon, 18 Oct 2010) | 24 lines Merged revisions 292224 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r292224 | lmadsen | 2010-10-18 16:50:47 -0500
+ (Mon, 18 Oct 2010) | 17 lines Merged revisions 292222 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010)
+ | 9 lines Add support for the new English (Australian Accent)
+ sound files. (closes issue #17426) Reported by: camsown Patches:
+ core-sounds-en_AU.txt uploaded by camsown (license 1050)
+ add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested
+ by: camsown, lmadsen, jtodd, qwell ........ ................
+ ................
+
+2010-10-18 21:56 +0000 [r292228] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 292227 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r292227 | jpeeler | 2010-10-18 16:55:46 -0500
+ (Mon, 18 Oct 2010) | 25 lines Merged revisions 292226 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r292226 | jpeeler | 2010-10-18 16:54:38 -0500
+ (Mon, 18 Oct 2010) | 18 lines Merged revisions 292223 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010)
+ | 11 lines Fix improper operator key acceptance and clean up temp
+ recording files. This is a fix for when pressing the operator key
+ after recording an unavailable, busy, name, or temporary message
+ in mailbox options. The operator key should not be accepted here,
+ but should be allowed during the message recording. If the
+ operator key is pressed during ensure the file is saved or
+ deleted as apporopriate. Also, ensure removal of temporary
+ recorded files after an early hang up or when message acceptance
+ confirmation times out. ABE-2518 ........ ................
+ ................
+
+2010-10-18 19:52 +0000 [r292189] Russell Bryant <russell@digium.com>
+
+ * main/netsock2.c, /: Merged revisions 292188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292188 | russell | 2010-10-18 14:50:04 -0500 (Mon, 18 Oct 2010)
+ | 9 lines Resolve some compiler errors in ast_sockaddr_is_any().
+ These errors came up once this function was used from within
+ netsock2.c. The errors were like the following: netsock2.c:393:
+ error: dereferencing pointer ‘({anonymous})’ does break
+ strict-aliasing rules The usage of a union here avoids this
+ problem. ........
+
+2010-10-18 19:16 +0000 [r292156] David Vossel <dvossel@digium.com>
+
+ * main/netsock2.c, /: Merged revisions 292155 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292155 | dvossel | 2010-10-18 14:16:00 -0500 (Mon, 18 Oct 2010)
+ | 2 lines Fixes build error for systems not supporting
+ IPV6_TCLASS. ........
+
+2010-10-18 17:18 +0000 [r292124] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, addons/chan_mobile.c: Merged revisions 292122 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r292122 | mnicholson | 2010-10-18 12:15:24 -0500 (Mon,
+ 18 Oct 2010) | 5 lines Fix the cmgr parser. (closes issue
+ 0018152) Reported by: menschentier ........
+
+2010-10-18 16:03 +0000 [r292086] David Vossel <dvossel@digium.com>
+
+ * main/netsock2.c, /: Merged revisions 292085 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292085 | dvossel | 2010-10-18 11:02:17 -0500 (Mon, 18 Oct 2010)
+ | 7 lines Fixes qos settings for sockets bound to any IPv6 or
+ IPv4 address. (closes issue #18099) Reported by: jamesnet
+ Patches: issues_18099_v3.diff uploaded by dvossel (license 671
+ ........
+
+2010-10-18 15:33 +0000 [r292084] Jeff Peeler <jpeeler@digium.com>
+
+ * pbx/pbx_spool.c, /: Merged revisions 292083 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292083 | jpeeler | 2010-10-18 10:32:40 -0500 (Mon, 18 Oct 2010)
+ | 4 lines Disable use of inotify for call file handling as it is
+ not working properly. (related to #18089) ........
+
+2010-10-16 11:51 +0000 [r292052] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, configs/musiconhold.conf.sample, res/res_musiconhold.c: Merged
+ revisions 292050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r292050 | tzafrir | 2010-10-16 12:47:00 +0200
+ (ש', 16 אוק 2010) | 22 lines Merged revisions 292049 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16
+ אוק 2010) | 15 lines Base directory for MOH should be ASTDATADIR
+ If the directive 'directory' is relative, make it relative to the
+ datadir, rather than to the varlibdir. In the sample
+ configuration it is relative ('moh'). This has no effect unless
+ you have actively set the datadir explicitly (at build time or at
+ run time). (closes issue #16906) Patches: moh_datadir uploaded by
+ tzafrir (license 46) Review:
+ https://reviewboard.asterisk.org/r/974/ ........ ................
+
+2010-10-15 21:49 +0000 [r292017] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_srtp.c: Merged revisions 292016 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r292016 | twilson | 2010-10-15 16:40:56 -0500 (Fri, 15 Oct 2010)
+ | 5 lines Ref/unref res_srtp when we create/destroy a session
+ This avoids unhappy crashing when we try to 'core stop
+ gracefully' and res_srtp tries to unload before chan_sip does.
+ Thanks, Russell! ........
+
+2010-10-15 20:12 +0000 [r291943] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 291942 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r291942 | dvossel | 2010-10-15 15:12:04 -0500 (Fri, 15 Oct 2010)
+ | 8 lines Fixes peer's host port information being lost on sip
+ reload. (closes issue #18135) Reported by: lmadsen Patches:
+ crazy_ports_v2.diff uploaded by dvossel (license 671) Tested by:
+ lmadsen ........
+
+2010-10-15 19:53 +0000 [r291941] Paul Belanger <pabelanger@digium.com>
+
+ * configs/gtalk.conf.sample, /: Merged revisions 291940 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r291940 | pabelanger | 2010-10-15 15:50:22 -0400
+ (Fri, 15 Oct 2010) | 16 lines Merged revisions 291939 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291939 | pabelanger | 2010-10-15 15:35:20 -0400
+ (Fri, 15 Oct 2010) | 9 lines Merged revisions 291938 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri,
+ 15 Oct 2010) | 2 lines Clean up formatting. ........
+ ................ ................
+
+2010-10-15 16:54 +0000 [r291906] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_jabber.c: Merged revisions 291905 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r291905 | twilson | 2010-10-15 09:39:58 -0700
+ (Fri, 15 Oct 2010) | 14 lines Merged revisions 291904 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010)
+ | 7 lines Don't crash or deadlock on module unload We can't hold
+ the lock while pthread_join is called since aji_log_hook will
+ attempt to lock from the other therad. We reorder the
+ pthread_join and ast_aji_disconnect so that we don't do an
+ SSL_read() while SSL_shutdown is running, causing a crash.
+ ........ ................
+
+2010-10-14 22:10 +0000 [r291828-291830] David Vossel <dvossel@digium.com>
+
+ * main/netsock2.c, /: Merged revisions 291829 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r291829 | dvossel | 2010-10-14 17:09:32 -0500 (Thu, 14 Oct 2010)
+ | 8 lines Set TCLASS field of IPv6 header when sip qos options
+ are set. (closes issue #18099) Reported by: jamesnet Patches:
+ issues_18099_v2.diff uploaded by dvossel (license 671) Tested by:
+ dvossel, jamesnet ........
+
+ * /, channels/chan_gtalk.c: Merged revisions 291827 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r291827 | dvossel | 2010-10-14 16:27:42 -0500 (Thu, 14
+ Oct 2010) | 18 lines Safer xml parsing, treat all clients the
+ same, and better local candidate selection. The gtalk channel
+ driver was doing several unsafe operations in regards to how it
+ parsed incoming XML messages. I have cleaned that code up so it
+ should be much safer now. We now treat all clients types the
+ same. We have no reason to distinguish between GMAIL and GOOGLE
+ VOICE clients anymore because they all work the same way. I also
+ modified how the local ip is found. If no bindaddress is provided
+ in the config file, we attempt to determine the local ip we would
+ use to connect to google.com. If that fails, then we fall back to
+ the ast_find_ourip() function as a last resort. Using the new
+ method makes it much less likely that we would ever advertise a
+ local RTP candidate as a loopback address. ........
+
+2010-10-14 18:46 +0000 [r291792] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/stdtime/localtime.c: Merged revisions 291791 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r291791 | jpeeler | 2010-10-14 13:45:02 -0500 (Thu, 14
+ Oct 2010) | 10 lines Add missing ifdefs for test framework and
+ new locale code. (closes issue #18137) Reported by: ovi Patches:
+ 18137_test_framework_ifdef.patch uploaded by wdoekes (license
+ 717) 18137_localelist_warning.patch uploaded by wdoekes (license
+ 717) Tested by: ovi ........
+
+2010-10-14 15:21 +0000 [r291760] Paul Belanger <pabelanger@digium.com>
+
+ * channels/chan_jingle.c, include/asterisk/acl.h, /,
+ channels/chan_sip.c, channels/chan_h323.c, main/acl.c,
+ channels/chan_gtalk.c: Merged revisions 291758 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r291758 | pabelanger | 2010-10-14 11:15:12 -0400 (Thu, 14 Oct
+ 2010) | 11 lines Add the ability for ast_find_ourip to return
+ IPv4, IPv6 or both. While testing chan_gtalk I noticed jabber was
+ using my IPv6 address and not IPv4. When using bindaddr=0.0.0.0
+ it is possible for ast_find_ourip() to return both IPv6 and IPv4
+ results. Adding a family parameter gives you the ablility to
+ choose. Since jabber/gtalk/h323 do not support IPv6, we should
+ only return IPv4 results. Review:
+ https://reviewboard.asterisk.org/r/973/ ........
+
+2010-10-14 12:10 +0000 [r291726] Russell Bryant <russell@digium.com>
+
+ * /, doc/tex/secure-calls.tex: Merged revisions 291725 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r291725 | russell | 2010-10-14 07:08:43 -0500 (Thu, 14
+ Oct 2010) | 2 lines Fix a typo - s/seucre/secure/ ........
+
+2010-10-13 23:52 +0000 [r291658] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Merged revisions 291656 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r291656 | rmudgett | 2010-10-13 18:45:11 -0500
+ (Wed, 13 Oct 2010) | 34 lines Merged revisions 291655 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291655 | rmudgett | 2010-10-13 18:36:50 -0500
+ (Wed, 13 Oct 2010) | 27 lines Merged revisions 291643 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010)
+ | 20 lines Deadlock between dahdi_exception() and
+ dahdi_indicate(). There is a deadlock between dahdi_exception()
+ and dahdi_indicate() for analog ports. The call-waiting and
+ three-way-calling feature can experience deadlock if these
+ features are trying to do something and an event from the bridged
+ channel happens at the same time. Deadlock avoidance code added
+ to obtain necessary channel locks before attemting an operation
+ with call-waiting and three-way-calling. (closes issue #16847)
+ Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch
+ uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+ Review: https://reviewboard.asterisk.org/r/971/ ........
+ ................ ................
+
+2010-10-13 23:47 +0000 [r291657] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Merged revisions 291581 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r291581 | twilson | 2010-10-13 16:01:56 -0700
+ (Wed, 13 Oct 2010) | 35 lines Merged revisions 291580 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291580 | twilson | 2010-10-13 15:58:43 -0700
+ (Wed, 13 Oct 2010) | 28 lines Merged revisions 291577 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010)
+ | 21 lines Don't ignore frames that have been queued when
+ softhangup'd When an outgoing call is answered and hung up by the
+ far end *very* quickly, we may not read any frames and therefor
+ end up with a call that displays the wrong
+ disposition/DIALSTATUS. The reason is because ast_queue_hangup()
+ immediately sets the _softhangup flag on the channel and then
+ queues the HANGUP control frame, but __ast_read refuses to read
+ any frames if ast_check_hangup() indicates that a hangup request
+ has been made (which it will if _softhangup is set). So, we end
+ up losing control frames. This change makes __ast_read continue
+ to read frames even if a soft hangup has been requested. It
+ queues a hangup frame to make sure that __ast_read() will still
+ eventually return NULL. Much thanks to David Vossel for all of
+ the reviews, discussion, and help! (closes issue #16946) Reported
+ by: davidw Review: https://reviewboard.asterisk.org/r/740/
+ ........ ................ ................
+
+2010-10-13 22:47 +0000 [r291579] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 291578 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r291578 | dvossel | 2010-10-13 17:46:34 -0500 (Wed, 13
+ Oct 2010) | 4 lines More fixup for chan_gtalk. This patch makes
+ the xml parsing safer. ........
+
+2010-10-13 22:34 +0000 [r291576] Terry Wilson <twilson@digium.com>
+
+ * Makefile, /, static-http/mantest.html (added): Merged revisions
+ 291575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r291575 | twilson | 2010-10-13 15:24:44 -0700 (Wed, 13 Oct 2010)
+ | 8 lines Add a simple AMI client web page This patch uses the
+ XML docs to parse all of the available AMI commands and allows
+ you to enter the command name and be presented with a form with
+ the available fields. You can then rapidly tab through the fields
+ and submit the command and view the response. It is much
+ faster/easier than having to use telnet for testing purposes.
+ ........
+
+2010-10-13 20:24 +0000 [r291470-291542] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 291541 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r291541 | rmudgett | 2010-10-13 15:21:02 -0500 (Wed, 13
+ Oct 2010) | 26 lines The chan_dahdi faxdetect option only works
+ for the first FAX call. The chan_dahdi faxdetect option only
+ works for the first call. After that the option no longer works.
+ The struct dahdi_pvt.callprogress member is the encoded user
+ config setting for the callprogress and faxdetect config options.
+ Changing this value alters the configuration for all following
+ calls until the chan_dahdi.conf file is reloaded. * Fixed the
+ chan_dahdi ast_channel_setoption callback to not change the users
+ faxdetect config setting except for the current call. * Fixed the
+ chan_dahdi ast_channel_queryoption callback to read the active
+ DSP setting of the faxdetect option. * Made actually disable the
+ active faxdetect DSP setting for the current call on the analog
+ port. my_handle_dtmfup() is used for normal analog ports.
+ dahdi_handle_dtmfup() is the legacy code and is no longer used
+ unless in a radio mode. (closes issue #18116) Reported by:
+ seandarcy Patches: issue18116_v1.8.patch uploaded by rmudgett
+ (license 664) Review: https://reviewboard.asterisk.org/r/972/
+ ........
+
+ * channels/chan_misdn.c, /: Merged revisions 291507 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r291507 | rmudgett | 2010-10-13 14:01:48 -0500
+ (Wed, 13 Oct 2010) | 18 lines Merged revision 291504 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed,
+ 13 Oct 2010) | 11 lines Hold off ast_hangup() from destroying the
+ ast_channel. Must get the ast_channel lock before proceeding with
+ release_chan() and release_chan_early() to hold off ast_hangup()
+ from destroying the ast_channel. Missed this change for -r291468.
+ JIRA ABE-2598 JIRA SWP-2317 .......... ................
+
+ * channels/chan_misdn.c, /: Merged revisions 291469 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r291469 | rmudgett | 2010-10-13 13:10:21 -0500
+ (Wed, 13 Oct 2010) | 23 lines Merge revision 291468 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed,
+ 13 Oct 2010) | 16 lines Memory overwrites when releasing mISDN
+ call. Phone <--> Asterisk <-- ALERTING --> DISCONNECT <-- RELEASE
+ --> RELEASE_COMPLETE * Add lock protection around channel list
+ for find/add/delete operations. * Protect misdn_hangup() from
+ release_chan() and vise versa using the release_lock. JIRA
+ ABE-2598 JIRA SWP-2317 .......... ................
+
+2010-10-13 15:51 +0000 [r291395] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 291394 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r291394 | russell | 2010-10-13 10:46:39 -0500
+ (Wed, 13 Oct 2010) | 20 lines Merged revisions 291393 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291393 | russell | 2010-10-13 10:29:21 -0500
+ (Wed, 13 Oct 2010) | 13 lines Merged revisions 291392 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010)
+ | 6 lines Lock pvt so pvt->owner can't disappear when queueing up
+ a frame. This fixes a crash due to a hangup race condition.
+ ABE-2601 ........ ................ ................
+
+2010-10-13 08:58 +0000 [r291361] Stefan Schmidt <sst@sil.at>
+
+ * apps/app_macro.c: Report what extension called a failed macro Add
+ the extension and context of the calling channel to the log
+ output if a macro could not be found. (closes issue #18112)
+ Reported by: prado Patches: app_macro-info.diff uploaded by prado
+ (license 510) Tested by: schmidts
+
+2010-10-12 17:21 +0000 [r291287] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/phoneprov.conf.sample: Merged revisions 291284 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r291284 | lmadsen | 2010-10-12 12:20:43 -0500
+ (Tue, 12 Oct 2010) | 15 lines Merged revisions 291280 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010)
+ | 7 lines Add undocumented variables to phoneprov.conf.sample
+ (closes issue #18107) Reported by: lathama Patches:
+ phoneprov.conf.sample.diff uploaded by lathama (license 1028)
+ ........ ................
+
+2010-10-12 17:07 +0000 [r291266] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/acl.c: Merged revisions 291265 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r291265 | tilghman | 2010-10-12 12:06:23 -0500
+ (Tue, 12 Oct 2010) | 16 lines Merged revisions 291264 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291264 | tilghman | 2010-10-12 12:05:31 -0500
+ (Tue, 12 Oct 2010) | 9 lines Merged revisions 291263 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12
+ Oct 2010) | 2 lines Oops, incorrect range (although unallocated
+ at ARIN) ........ ................ ................
+
+2010-10-12 16:08 +0000 [r291231] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/manager.conf.sample: Merged revisions 291230 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r291230 | lmadsen | 2010-10-12 11:08:04 -0500
+ (Tue, 12 Oct 2010) | 10 lines Merged revisions 291229 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291229 | lmadsen | 2010-10-12 11:07:28 -0500 (Tue, 12 Oct 2010)
+ | 2 lines Add documention that mentions options are defined but
+ not used. (Issue #18101) ........ ................
+
+2010-10-12 16:00 +0000 [r291193-291228] David Vossel <dvossel@digium.com>
+
+ * main/manager.c, /: Merged revisions 291227 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r291227 | dvossel | 2010-10-12 10:58:56 -0500 (Tue, 12 Oct 2010)
+ | 16 lines Fixes manager.c crash. This issue was caused by
+ improper use of the mansession lock and manession_session lock.
+ These two structures are confusing to begin with so I'm not
+ surprised this occurred. I fixed this by consistently making sure
+ we use each of these locks only to protect the data in the
+ corresponding structure. We had mismatched usage of these locks
+ which resulted in no mutual exclusivity occurring at all. (closes
+ issue #17994) Reported by: vrban Patches:
+ mansession_locking_fix.diff uploaded by dvossel (license 671)
+ Tested by: vrban ........
+
+ * /, CHANGES: Merged revisions 291194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r291194 | dvossel | 2010-10-11 16:44:04 -0500 (Mon, 11 Oct 2010)
+ | 2 lines Update CHANGES to reflect new gtalk.conf options.
+ ........
+
+ * configs/gtalk.conf.sample, /, res/res_stun_monitor.c,
+ channels/chan_gtalk.c, include/asterisk/stun.h: Merged revisions
+ 291192 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r291192 | dvossel | 2010-10-11 16:38:39 -0500 (Mon, 11 Oct 2010)
+ | 19 lines Gtalk enhancements and general code cleanup. This
+ patch includes several chan_gtalk enhancements. Two new
+ gtalk.conf options have been added, externip and stunadd. Setting
+ externip allows us to manually specify what the external IP
+ address is outside of a NAT environment. Setting the stunaddr
+ option to a valid stun server allows for that external ip to be
+ retrieved via a STUN server automatically. This external IP is
+ then advertised during call setup as a possible candidate. I have
+ also attempted to clean up chan_gtalk's code so it meets our
+ coding guidelines. During this cleanup I noticed several things
+ that need to be done in the code and made a TODO section at the
+ top of the file. ........
+
+2010-10-11 19:07 +0000 [r291076-291115] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Add todo comment about handle_incoming()
+ calling assumption.
+
+ * /, channels/chan_sip.c: Merged revisions 291112-291113 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r291112 | rmudgett | 2010-10-11 13:48:15 -0500
+ (Mon, 11 Oct 2010) | 20 lines Merged revisions 291110-291111 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500
+ (Mon, 11 Oct 2010) | 9 lines Merged revisions 291109 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11
+ Oct 2010) | 1 line Add missing unlock to an exception condition
+ in reload_config(). ........ ................ r291111 | rmudgett
+ | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit
+ from handle_request_do() consistent. ................
+ ................ r291113 | rmudgett | 2010-10-11 13:51:13 -0500
+ (Mon, 11 Oct 2010) | 1 line Move declaration closer to where now
+ used. ................
+
+ * /, main/cli.c: Merged revisions 291075 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r291075 | rmudgett | 2010-10-11 11:42:54 -0500
+ (Mon, 11 Oct 2010) | 22 lines Merged revisions 291073 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010)
+ | 15 lines Fixed infinite loop in verbose/debug message output.
+ Setting the module/filename specific message level and then
+ changing it resulted in the linked list being looped on itself.
+ Traversing this linked list is an infinite loop if what you are
+ looking for is not in the list. Also plugged some CLI parsing
+ holes in the associated CLI command: * Removing a nonexistent
+ module from the list actually added it with a level of zero. *
+ Setting the non-module specific level to zero is now equivalent
+ to setting it to "off" as documented. ........ ................
+
+2010-10-11 03:20 +0000 [r291039] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Merged
+ revisions 291038 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r291038 | tilghman | 2010-10-09 18:25:37 -0500 (Sat, 09 Oct 2010)
+ | 2 lines Add missing option to set calls to be logged in
+ GMT/UTC. ........
+
+2010-10-09 14:04 +0000 [r291006] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c: Added fast start and h.245 tunneling
+ options per user and peer. Added options for faststart/h.245
+ tunneling per user/peer, properly handle these and global
+ options, correction of handling fs/tunneling fields in signalling
+ responses (closes issue #17972) Reported by: salecha Patches:
+ fs-tunnel-per-point-3.patch uploaded by may213 (license 454)
+ Tested by: may213, salecha
+
+2010-10-08 20:45 +0000 [r290974] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 290973 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r290973 | dvossel | 2010-10-08 15:44:59 -0500 (Fri, 08
+ Oct 2010) | 12 lines Make outbound Google Voice calls. This patch
+ allows for outbound Google Voice calls to be dialed from Asterisk
+ using chan_gtalk. Below is an example dialstring. exten ->
+ blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,) In
+ this example, 'asterisk' is the jabber.conf profile configured to
+ connect to your gmail account. In order to receive Google Voice
+ calls make sure to enable 'allowguest=yes' in gtalk.conf.
+ ........
+
+2010-10-08 16:27 +0000 [r290939] Erin Spiceland <erin@thespicelands.com>
+
+ * addons/app_mysql.c, configs/res_config_mysql.conf.sample, /,
+ addons/res_config_mysql.c: Add option to res_config_mysql and
+ app_mysql to specify a character set that MySQL should use.
+ (closes issue 17948) Reported by qmax.
+
+2010-10-08 03:00 +0000 [r290865] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 290864 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r290864 | jpeeler | 2010-10-07 21:56:24 -0500
+ (Thu, 07 Oct 2010) | 23 lines Merged revisions 290863 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290863 | jpeeler | 2010-10-07 21:45:44 -0500
+ (Thu, 07 Oct 2010) | 16 lines Merged revisions 290862 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010)
+ | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed
+ at control console. A recent change was made to avoid a race
+ condition on shutdown which only called the end functions from
+ the console thread. However, when pressing Ctrl-C the quit
+ handler is called from the signal handler thread. (closes issue
+ #17698) Reported by: jmls ........ ................
+ ................
+
+2010-10-07 22:39 +0000 [r290830-290831] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 290829 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r290829 | dvossel | 2010-10-07 17:38:05 -0500 (Thu, 07
+ Oct 2010) | 6 lines Add Philippe Sultan to chan_gtalk author
+ list. Philippe has made some notable contributions to the gtalk
+ channel driver. His name deserves to be listed amoung the authors
+ of that file. Thanks Philippe! ........
+
+ * /, channels/chan_gtalk.c: Merged revisions 290828 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r290828 | dvossel | 2010-10-07 16:44:58 -0500 (Thu, 07
+ Oct 2010) | 5 lines Outbound gtalk calls now work correctly.
+ There was a problem with how the candidates were being built on
+ an outbound call. This patch fixes that. ........
+
+2010-10-07 20:59 +0000 [r290753] Jason Parker <jparker@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ autoconf/ast_ext_lib.m4: Merged revisions 290752 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r290752 | qwell | 2010-10-07 15:58:47 -0500
+ (Thu, 07 Oct 2010) | 23 lines Merged revisions 290751 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290751 | qwell | 2010-10-07 15:57:14 -0500
+ (Thu, 07 Oct 2010) | 16 lines Merged revisions 290750 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) |
+ 9 lines Allow PRI to build properly when using --with-pri. Use
+ the directories found for the parent when using lib dependencies.
+ (closes issue #17314) Reported by: tzafrir Patches:
+ 17314-withdeps.diff uploaded by qwell (license 4) ........
+ ................ ................
+
+2010-10-07 11:12 +0000 [r290714] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, /: Merged revisions 290713 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r290713 | russell | 2010-10-07 13:00:52 +0200
+ (Thu, 07 Oct 2010) | 11 lines Merged revisions 290712 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290712 | russell | 2010-10-07 12:53:56 +0200 (Thu, 07 Oct 2010)
+ | 4 lines Don't crash when Set() is called without a value.
+ Review: https://reviewboard.asterisk.org/r/949/ ........
+ ................
+
+2010-10-06 21:23 +0000 [r290649-290677] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 290674 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r290674 | dvossel | 2010-10-06 16:22:51 -0500 (Wed, 06
+ Oct 2010) | 4 lines Fixes commented out code to use #if 0
+ instead. Thanks to rmudgett for catching this! ........
+
+ * /, channels/chan_gtalk.c: Merged revisions 290648 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r290648 | dvossel | 2010-10-06 16:08:19 -0500 (Wed, 06
+ Oct 2010) | 12 lines Fixes gtalk outbound DTMF to work properly.
+ Outbound DTMF with gtalk needs to be done within the RTP stream.
+ I discovered this after investigating a packet capture from the
+ gmail client. Instead of performing jingle signaling DTMF, the
+ gtalk servers expect all DTMF to arrive on the RTP stream using
+ RFC2833 way of doing things. Chan_gtalk also had an issue with
+ negotiating RTP payload type 106 for the telephony-event and then
+ sending DTMF as payload 101. This has been resolved by always
+ negotiating 101 as the payload type like we do everywhere else.
+ With this patch, incoming google voice calls forwarded to
+ Asterisk via gtalk work. ........
+
+2010-10-06 18:56 +0000 [r290615] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 290614 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r290614 | rmudgett | 2010-10-06 13:50:37 -0500
+ (Wed, 06 Oct 2010) | 12 lines Merged revision 290613 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed,
+ 06 Oct 2010) | 5 lines Eliminate a redundant test for
+ AST_CONTROL_REDIRECTING. Eliminate redundant test for
+ AST_CONTROL_REDIRECTING that prevents running the redirecting
+ interception macro if it is defined. .......... ................
+
+2010-10-06 13:50 +0000 [r290577] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/file.c: Merged revisions 290576 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r290576 | tilghman | 2010-10-06 08:49:19 -0500
+ (Wed, 06 Oct 2010) | 15 lines Merged revisions 290575 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010)
+ | 8 lines Allow streaming audio from a pipe. (closes issue
+ #18001) Reported by: jamicque Patches:
+ 20100926__issue18001.diff.txt uploaded by tilghman (license 14)
+ Tested by: jamicque ........ ................
+
+2010-10-06 04:47 +0000 [r290543] Terry Wilson <twilson@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Merged revisions 290542 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r290542 | twilson | 2010-10-05 21:35:51 -0700 (Tue, 05
+ Oct 2010) | 6 lines Don't try to send RTP when remote_address is
+ null It is possible for ast_rtp_stop() to be called which will
+ clear the remote address and cause the sendto to fail and spam
+ warnings. Don't send in this case. ........
+
+2010-10-05 22:23 +0000 [r290480-290509] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 290506 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r290506 | dvossel | 2010-10-05 17:23:00 -0500 (Tue, 05
+ Oct 2010) | 2 lines Fixes uninitialized memory problem in 'iax2
+ set debug peer' option. ........
+
+ * /, include/asterisk/jabber.h, include/asterisk/jingle.h,
+ channels/chan_gtalk.c, res/res_jabber.c: Merged revisions 290479
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r290479 | dvossel | 2010-10-05 17:00:43 -0500 (Tue, 05 Oct 2010)
+ | 6 lines Fixes chan_gtalk to work with gmail client This patch
+ was written by Philippe Sultan (phsultan). Thanks for keeping
+ this up to date! ........
+
+2010-10-05 20:24 +0000 [r290414] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, res/res_jabber.c: Merged revisions 290408 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r290408 | tilghman | 2010-10-05 15:23:33 -0500
+ (Tue, 05 Oct 2010) | 22 lines Merged revisions 290396 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290396 | tilghman | 2010-10-05 15:21:02 -0500
+ (Tue, 05 Oct 2010) | 15 lines Merged revisions 290392 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010)
+ | 8 lines Fix a crash by ensuring that we don't alter memory
+ after it's freed. (closes issue #17387) Reported by: jmls
+ Patches: 20100726__issue17387.diff.txt uploaded by tilghman
+ (license 14) Tested by: jmls ........ ................
+ ................
+
+2010-10-05 20:10 +0000 [r290377-290379] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 290378 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r290378 | dvossel | 2010-10-05 15:09:06 -0500 (Tue, 05
+ Oct 2010) | 11 lines Resolves dnsmgr memory corruption in
+ chan_iax2. (closes issue #17902) Reported by: afried Patches:
+ issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
+ afried, russell, dvossel Review:
+ https://reviewboard.asterisk.org/r/965/ ........
+
+ * /, apps/app_directed_pickup.c: Merged revisions 290376 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r290376 | dvossel | 2010-10-05 14:56:29 -0500
+ (Tue, 05 Oct 2010) | 16 lines Merged revisions 290375 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010)
+ | 10 lines Fixes PickupChan() not working with full channel name.
+ (closes issue #18011) Reported by: schern Patches:
+ app_directed_pickup.c.2.patch uploaded by schern (license 995)
+ app_directed_pickup.c.trunk.patch uploaded by schern (license
+ 995) Tested by: schern, dvossel ........ ................
+
+2010-10-05 14:17 +0000 [r290067-290291] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, configure, configure.ac: Merged revisions 290289 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r290289 | tilghman | 2010-10-05 09:15:46 -0500 (Tue, 05
+ Oct 2010) | 2 lines Restore run directory for OS X, as well as
+ standardizing some other paths to Mac OS X. ........
+
+ * res/ael/pval.c, main/pbx.c, pbx/ael/ael-test/ref.ael-vtest17, /,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+ pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
+ pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5,
+ pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 290255 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r290255 | tilghman | 2010-10-04 18:23:11 -0500
+ (Mon, 04 Oct 2010) | 18 lines Merged revisions 290254 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010)
+ | 11 lines Change new pattern matcher to regard dashes the same
+ as the old pattern matcher -- as visual candy to be ignored. Also
+ change the AEL parser to not generate dashes within extensions,
+ as those dashes would be ignored. Update the AEL tests to match
+ this behavior. (closes issue #17366) Reported by: murf Patches:
+ 20100727__issue17366.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........ ................
+
+ * /, configure, configure.ac: Merged revisions 290209 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r290209 | tilghman | 2010-10-04 15:23:13 -0500
+ (Mon, 04 Oct 2010) | 16 lines Merged revisions 290201 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290201 | tilghman | 2010-10-04 15:22:03 -0500
+ (Mon, 04 Oct 2010) | 9 lines Merged revisions 290177 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04
+ Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........
+ ................ ................
+
+ * /, configure, configure.ac: Merged revisions 290102 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r290102 | tilghman | 2010-10-03 16:08:45 -0500
+ (Sun, 03 Oct 2010) | 16 lines Merged revisions 290101 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290101 | tilghman | 2010-10-03 16:06:58 -0500
+ (Sun, 03 Oct 2010) | 9 lines Merged revisions 290100 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03
+ Oct 2010) | 2 lines Automatically re-run configure test for
+ menuselect, when the relevant makeopts settings change. ........
+ ................ ................
+
+ * pbx/pbx_spool.c, /: Merged revisions 290066 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r290066 | tilghman | 2010-10-03 15:02:29 -0500 (Sun, 03 Oct 2010)
+ | 8 lines Get notification only when file is closed, not when
+ created. (closes issue #17924) Reported by: mkeuter Patches:
+ asterisk-1.8-bugid17924.patch uploaded by abelbeck (license 946)
+ Tested by: abelbeck ........
+
+2010-10-02 18:06 +0000 [r290027] Kevin P. Fleming <kpfleming@digium.com>
+
+ * contrib/scripts/get_mp3_source.sh, /: Merged revisions 290026 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r290026 | kpfleming | 2010-10-02 12:57:13 -0500 (Sat, 02 Oct
+ 2010) | 6 lines Allow users to pass additional arguments to the
+ Subversion command that obtains the MP-3 source code. (reported
+ on IRC by jmls) ........
+
+2010-10-02 08:58 +0000 [r289952] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c, /: Merged revisions 289951 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r289951 | oej | 2010-10-02 10:56:08 +0200 (Lör,
+ 02 Okt 2010) | 16 lines Merged revisions 289950 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289950 | oej | 2010-10-02 10:52:03 +0200 (Lör,
+ 02 Okt 2010) | 9 lines Merged revisions 289949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2
+ lines Add documentation for undocumented option to AMI action
+ originate ........ ................ ................
+
+2010-10-02 04:54 +0000 [r289876] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, apps/app_voicemail.c: Merged revisions 289875 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r289875 | tilghman | 2010-10-01 23:46:43 -0500
+ (Fri, 01 Oct 2010) | 22 lines Merged revisions 289874 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289874 | tilghman | 2010-10-01 23:45:49 -0500
+ (Fri, 01 Oct 2010) | 15 lines Merged revisions 289873 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010)
+ | 8 lines When forwarding a message, a prepend means that the
+ filesystem will always have a better copy. (closes issue #17803)
+ Reported by: dpetersen Patches: 20100923__issue17803.diff.txt
+ uploaded by tilghman (license 14) Tested by: dpetersen ........
+ ................ ................
+
+2010-10-02 02:46 +0000 [r289841] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, /,
+ channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged
+ revisions 289840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r289840 | jpeeler | 2010-10-01 21:43:45 -0500
+ (Fri, 01 Oct 2010) | 29 lines Merged revisions 289798 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500
+ (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010)
+ | 15 lines Change RFC2833 DTMF event duration on end to report
+ actual elapsed time. The scenario here is with a non P2P early
+ media session. The reported time length of DTMF presses are
+ coming up short when sending to the remote side. Currently the
+ event duration is a running total that is incremented when
+ sending continuation packets. These continuation packets are only
+ triggered upon incoming media from the remote side, which means
+ that the running total probably is not going to end up matching
+ the actual length of time Asterisk received DTMF. This patch
+ changes the end event duration to be lengthened if it is detected
+ that the end event is going to come up short. Review:
+ https://reviewboard.asterisk.org/r/957/ ABE-2476 ........
+ ................ ................
+
+2010-10-01 17:22 +0000 [r289732] Paul Belanger <pabelanger@digium.com>
+
+ * /, configs/jabber.conf.sample, res/res_jabber.c: Merged revisions
+ 289718 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r289718 | pabelanger | 2010-10-01 13:19:49 -0400
+ (Fri, 01 Oct 2010) | 20 lines Merged revisions 289704 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289704 | pabelanger | 2010-10-01 13:09:03 -0400
+ (Fri, 01 Oct 2010) | 13 lines Merged revisions 289703 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct
+ 2010) | 6 lines Disable debugging by default and reformat .config
+ file. Review: https://reviewboard.asterisk.org/r/929/ ........
+ ................ ................
+
+2010-10-01 16:23 +0000 [r289702] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 289701 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r289701 | jpeeler | 2010-10-01 11:22:19 -0500
+ (Fri, 01 Oct 2010) | 28 lines Merged revisions 289700 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500
+ (Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010)
+ | 14 lines Ensure user portion of SIP URI matches dialplan when
+ using encoded characters. This commit takes a simliar approach to
+ 288112 and checks the dialplan to determine the proper action for
+ an incoming contact header as to whether or not it should be
+ decoded or not. sip_new was blindly always decoding the
+ extension, which also caused the outgoing contact header to be
+ incorrect as well as failing to match the encoded extension in
+ the dialplan. (closes issue #17892) Reported by: wdoekes Patches:
+ bug17892-1.patch uploaded by jpeeler (license 325) Tested by:
+ wdoekes ........ ................ ................
+
+2010-10-01 10:04 +0000 [r289623] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: don't iterate through all dialogs to find
+ and delete old subscribes On every incoming subscribe there is a
+ iteration through all dialogs to find old subscribes and delete
+ them. This is slow and not RFC conform. This was only needed in
+ 1.2 cause a subscribe was not deleted when a dialog was
+ destroyed, after 1.4 a subscribe get removed when its dialog is
+ destroyed. Review: https://reviewboard.asterisk.org/r/901/
+
+2010-09-30 20:40 +0000 [r289588] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, tests/test_time.c, funcs/func_env.c, tests/test_utils.c,
+ res/res_agi.c, include/asterisk/localtime.h,
+ main/stdtime/localtime.c: Merged revisions 289543,289581 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r289543 | tilghman | 2010-09-30 12:50:52 -0500 (Thu, 30 Sep 2010)
+ | 2 lines More Solaris compatibility fixes ........ r289581 |
+ tilghman | 2010-09-30 15:23:10 -0500 (Thu, 30 Sep 2010) | 2 lines
+ Solaris fixes. ........
+
+2010-09-30 19:54 +0000 [r289555] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 289554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r289554 | mnicholson | 2010-09-30 14:53:10 -0500
+ (Thu, 30 Sep 2010) | 11 lines Merged revisions 289553 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep
+ 2010) | 4 lines Properly handle channel allocation failures duing
+ invites with replaces. ABE-2588 ........ ................
+
+2010-09-30 19:35 +0000 [r289552] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 289549 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r289549 | rmudgett | 2010-09-30 14:28:36 -0500
+ (Thu, 30 Sep 2010) | 17 lines Merged revision 289547 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu,
+ 30 Sep 2010) | 10 lines In chan_misdn, the
+ DivertingLegInformation2 DivertingNr is garbage when the number
+ is restricted. The same thing happens with
+ DivertingLegInformation1 DivertedTo number. The
+ misdn_PresentedNumberUnscreened_extract() extracted the
+ Unscreened PartyNumber field unconditionally. It now checks the
+ presented number unscreened type to see if the PartyNumber was
+ even present. JIRA ABE-2595 .......... ................
+
+2010-09-30 15:40 +0000 [r289427] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_sms.c: Merged revisions 289426 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r289426 | russell | 2010-09-30 10:39:45 -0500
+ (Thu, 30 Sep 2010) | 22 lines Merged revisions 289425 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289425 | russell | 2010-09-30 10:37:29 -0500
+ (Thu, 30 Sep 2010) | 15 lines Merged revisions 289424 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010)
+ | 8 lines Fix a crash in app_sms. Since the data being passed to
+ the generator callback is on the stack of the SMS() application,
+ we must ensure that the generator is stopped before the
+ application exits. ABE-2587 ........ ................
+ ................
+
+2010-09-29 21:19 +0000 [r289354] Jason Parker <jparker@digium.com>
+
+ * main/channel.c, /, main/features.c: Merged revisions 289340 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r289340 | qwell | 2010-09-29 16:12:43 -0500
+ (Wed, 29 Sep 2010) | 22 lines Merged revisions 289339 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289339 | qwell | 2010-09-29 16:03:47 -0500
+ (Wed, 29 Sep 2010) | 15 lines Merged revisions 289338 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) |
+ 8 lines Allow a manager originate to succeed on forwarded
+ devices. The timeout to wait for an answer was being set to 0
+ when a device forwarded to another extension. We don't always
+ need the timeout set like this, so make it an optional parameter,
+ and don't use it in this case. ABE-2544 ........ ................
+ ................
+
+2010-09-29 20:29 +0000 [r289337] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/res_ldap.conf.sample: Merged revisions 289336 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r289336 | lmadsen | 2010-09-29 15:27:25 -0500
+ (Wed, 29 Sep 2010) | 9 lines Merged revisions 289334 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r289334 | lmadsen | 2010-09-29 15:24:47 -0500 (Wed, 29
+ Sep 2010) | 1 line Update sample documentation to note md5secret
+ requirements. ........ ................
+
+2010-09-29 20:24 +0000 [r289335] Russell Bryant <russell@digium.com>
+
+ * /, res/res_config_ldap.c: Merged revisions 289333 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r289333 | russell | 2010-09-29 15:20:23 -0500
+ (Wed, 29 Sep 2010) | 11 lines Merged revisions 289332 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r289332 | russell | 2010-09-29 15:15:57 -0500 (Wed, 29 Sep 2010)
+ | 4 lines Don't completely ignore md5secret from LDAP if the
+ value does not begin with {md5}. This fixes a problem that
+ lmadsen ran in to where md5secret was not working for him.
+ ........ ................
+
+2010-09-29 17:54 +0000 [r289269-289301] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, configs/res_fax.conf.sample: Merged revisions 289300 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r289300 | mnicholson | 2010-09-29 12:53:54 -0500 (Wed, 29 Sep
+ 2010) | 2 lines Add 'ecm' to the sample fax config file ........
+
+ * main/channel.c, /: Merged revisions 289268 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r289268 | mnicholson | 2010-09-29 12:08:20 -0500 (Wed, 29 Sep
+ 2010) | 5 lines Update the CDR record when
+ ast_channel_set_caller_event() is called (related to issue
+ #17569) Reported by: tbelder ........
+
+2010-09-29 16:17 +0000 [r289254] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Merged revisions 289253 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r289253 | rmudgett | 2010-09-29 11:16:47 -0500 (Wed, 29 Sep 2010)
+ | 1 line Make development error message indicate which channel.
+ ........
+
+2010-09-29 15:07 +0000 [r289180] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 289179 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r289179 | mnicholson | 2010-09-29 10:04:56 -0500
+ (Wed, 29 Sep 2010) | 22 lines Merged revisions 289178 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289178 | mnicholson | 2010-09-29 10:04:11 -0500
+ (Wed, 29 Sep 2010) | 15 lines Merged revisions 289177 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep
+ 2010) | 8 lines Set the caller id on CDRs when it is set on the
+ parent channel. (closes issue #17569) Reported by: tbelder
+ Patches: 17569.diff uploaded by tbelder (license 618) Tested by:
+ tbelder ........ ................ ................
+
+2010-09-28 18:24 +0000 [r289131] Brett Bryant <bbryant@digium.com>
+
+ * main/channel.c, /: Merged revisions 289099 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r289099 | bbryant | 2010-09-28 14:18:02 -0400
+ (Tue, 28 Sep 2010) | 28 lines Merged revisions 289095 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289095 | bbryant | 2010-09-28 14:14:19 -0400
+ (Tue, 28 Sep 2010) | 21 lines Merged revisions 289094 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010)
+ | 14 lines Fixes an issue with the Newchannel AMI event during
+ the Masquerading process. Fixes an issue with the Newchannel AMI
+ event during the Masquerading process, where no Newchannel AMI
+ event was generated for the psuedo channel used during the
+ masquerading process. (closes issue #17987) Reported by:
+ RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish
+ (license 1122) Tested by: RadicAlish Review:
+ https://reviewboard.asterisk.org/r/937/ ........ ................
+ ................
+
+2010-09-28 18:20 +0000 [r289112] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * Makefile, /, tests/test_time.c, configure,
+ include/asterisk/autoconfig.h.in, include/asterisk/compat.h,
+ main/strcompat.c, tests/test_utils.c, configure.ac, makeopts.in,
+ apps/app_voicemail.c: Merged revisions 289104 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r289104 | tilghman | 2010-09-28 13:18:43 -0500 (Tue, 28 Sep 2010)
+ | 4 lines Solaris compatibility fixes Review:
+ https://reviewboard.asterisk.org/r/942/ ........
+
+2010-09-28 01:10 +0000 [r289056-289058] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 289057 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r289057 | rmudgett | 2010-09-27 20:04:37 -0500 (Mon, 27 Sep 2010)
+ | 5 lines Avoid deadlock processing incoming AOC-E messages.
+ Deadlock avoidance for the owner channel was not done when
+ processing incoming AOC-E messages. ........
+
+ * /, channels/chan_sip.c: Merged revisions 289054-289055 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r289054 | rmudgett | 2010-09-27 19:32:18 -0500 (Mon, 27 Sep 2010)
+ | 1 line Break up long ast_manager_event_multichan() event lines.
+ ........ r289055 | rmudgett | 2010-09-27 19:35:25 -0500 (Mon, 27
+ Sep 2010) | 1 line Revert stuff not ready for commit in -r289054.
+ ........
+
+2010-09-27 22:03 +0000 [r289023] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: For an INVITE transaction, treat all 2XX
+ responses the same as a 200. ABE-2305
+
+2010-09-27 19:45 +0000 [r288992-288993] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Formatting fixes
+
+ * cdr/cdr_pgsql.c: Formating changes
+
+2010-09-27 18:39 +0000 [r288962] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, channels/chan_sip.c: Merged revisions 288961 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r288961 | tilghman | 2010-09-27 13:37:41 -0500 (Mon, 27 Sep 2010)
+ | 5 lines Still build SIP, even if res_crypto cannot be built
+ (use, not depend). (closes issue #18062) Reported by: a user on
+ the mailing list ........
+
+2010-09-27 13:04 +0000 [r288926-288928] Russell Bryant <russell@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 288927 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r288927 | russell | 2010-09-27 08:03:43 -0500 (Mon, 27 Sep 2010)
+ | 2 lines Fix some documentation typos and spelling errors.
+ ........
+
+ * /, res/res_agi.c: Merged revisions 288925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r288925 | russell | 2010-09-27 07:42:10 -0500 (Mon, 27 Sep 2010)
+ | 2 lines Fix a documentation spelling error. ........
+
+2010-09-25 07:58 +0000 [r288893] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.c: small correction for verbose
+ print h.323 packets
+
+2010-09-24 17:59 +0000 [r288822-288853] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288852 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r288852 | dvossel | 2010-09-24 12:58:57 -0500 (Fri, 24 Sep 2010)
+ | 5 lines Append Retry-After header on 500 error response to
+ Re-INVITE according to RFC3261 section 14.2. ABE-2301 ........
+
+ * /, channels/chan_sip.c: Merged revisions 288821 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r288821 | dvossel | 2010-09-24 12:05:12 -0500 (Fri, 24 Sep 2010)
+ | 4 lines Inspect Require header on BYE transaction according to
+ RFC3261 section 8.2.2.3. ABE-2293 ........
+
+2010-09-24 16:11 +0000 [r288749] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 288748 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r288748 | twilson | 2010-09-24 09:02:27 -0700
+ (Fri, 24 Sep 2010) | 19 lines Merged revisions 288747 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288747 | twilson | 2010-09-24 08:37:39 -0700
+ (Fri, 24 Sep 2010) | 12 lines Merged revisions 288746 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010)
+ | 5 lines Don't fail a masquerade if it is already being hung up
+ This avoids noise on some Local channel situations where we don't
+ use /n. Thanks to Alec Davis for the suggestion. ........
+ ................ ................
+
+2010-09-24 13:55 +0000 [r288607-288714] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, funcs/func_strings.c: Merged revisions 288713 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r288713 | tilghman | 2010-09-24 08:54:17 -0500
+ (Fri, 24 Sep 2010) | 12 lines Merged revisions 288712 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r288712 | tilghman | 2010-09-24 08:53:30 -0500 (Fri, 24 Sep 2010)
+ | 5 lines Solaris won't printf a NULL. (closes issue #18041)
+ Reported by: asgaroth ........ ................
+
+ * /, main/asterisk.exports.in: Merged revisions 288640 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r288640 | tilghman | 2010-09-23 22:42:37 -0500 (Thu, 23
+ Sep 2010) | 2 lines Export timersub for platforms which do not
+ have it ........
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, configure.ac,
+ include/asterisk/channel.h, cdr/cdr_pgsql.c: Merged revisions
+ 288638 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r288638 | tilghman | 2010-09-23 22:39:29 -0500
+ (Thu, 23 Sep 2010) | 16 lines Merged revisions 288637 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288637 | tilghman | 2010-09-23 22:36:01 -0500
+ (Thu, 23 Sep 2010) | 9 lines Merged revisions 288636 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23
+ Sep 2010) | 2 lines Solaris compatibility fixes ........
+ ................ ................
+
+ * /, CHANGES: Merged revisions 288606 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r288606 | tilghman | 2010-09-23 13:44:44 -0500 (Thu, 23 Sep 2010)
+ | 2 lines Add note about the checkhangup option of ${CHANNEL()}
+ ........
+
+2010-09-23 18:08 +0000 [r288519-288573] Terry Wilson <twilson@digium.com>
+
+ * main/manager.c, /: Merged revisions 288572 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r288572 | twilson | 2010-09-23 13:05:16 -0500 (Thu, 23 Sep 2010)
+ | 2 lines Make AMI honor enabled=no ........
+
+ * channels/chan_local.c, /: Merged revisions 288507 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r288507 | twilson | 2010-09-22 16:18:27 -0700
+ (Wed, 22 Sep 2010) | 22 lines Merged revisions 288500 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288500 | twilson | 2010-09-22 16:10:09 -0700
+ (Wed, 22 Sep 2010) | 15 lines Merged revisions 288499 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010)
+ | 8 lines Don't let a Local channel get bridged to itself If a
+ local channel gets bridged to itself, it becomes orphaned with no
+ devices left to actually tell it to hang up. This patch modifies
+ local_fixup() to detect this case and deny it. Review:
+ https://reviewboard.asterisk.org/r/934 ........ ................
+ ................
+
+2010-09-22 17:50 +0000 [r288346-288419] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288418 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r288418 | dvossel | 2010-09-22 12:49:56 -0500
+ (Wed, 22 Sep 2010) | 18 lines Merged revisions 288417 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288417 | dvossel | 2010-09-22 12:49:05 -0500
+ (Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010)
+ | 5 lines RFC3261 section 12.2 explicitly says out of order
+ requests are responded with a 500 Server Internal Error response.
+ ABE-2458 ........ ................ ................
+
+ * /, channels/chan_sip.c: Merged revisions 288345 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r288345 | dvossel | 2010-09-22 11:59:14 -0500
+ (Wed, 22 Sep 2010) | 16 lines Merged revisions 288344 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288344 | dvossel | 2010-09-22 11:53:28 -0500
+ (Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22
+ Sep 2010) | 2 lines During check_pendings, if the dialog is
+ terminated with a CANCEL, change the invitestate to INV_CANCEL
+ like in sip_hangup. ........ ................ ................
+
+2010-09-22 16:46 +0000 [r288342] Russell Bryant <russell@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 288341 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r288341 | russell | 2010-09-22 11:45:18 -0500
+ (Wed, 22 Sep 2010) | 25 lines Merged revisions 288340 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288340 | russell | 2010-09-22 11:44:13 -0500
+ (Wed, 22 Sep 2010) | 18 lines Merged revisions 288339 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010)
+ | 11 lines Fix a 100% CPU consumption problem when setting
+ console=yes in asterisk.conf. The handling of -c and console=yes
+ should be the same, but they were not. When you specify -c, it
+ sets both a flag for console module and for asterisk not to
+ fork() off into the background. The handling of console=yes only
+ set console mode, so you would end up with a background process()
+ trying to run the Asterisk console and freaking out since it
+ didn't have anything to read input from. Thanks to beagles for
+ reporting and helping debug the problem! ........
+ ................ ................
+
+2010-09-22 15:18 +0000 [r288278] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Merged
+ revisions 288268 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r288268 | tilghman | 2010-09-22 10:14:02 -0500
+ (Wed, 22 Sep 2010) | 30 lines Merged revisions 288267 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288267 | tilghman | 2010-09-22 10:11:09 -0500
+ (Wed, 22 Sep 2010) | 23 lines Merged revisions 288265-288266 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010)
+ | 9 lines Allow the encoding to be set, in case local charset
+ does not agree with database. (closes issue #16940) Reported by:
+ jamicque Patches: 20100827__issue16940.diff.txt uploaded by
+ tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt
+ uploaded by tilghman (license 14) Tested by: jamicque ........
+ r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010)
+ | 5 lines Document addition of encoding parameter. (issue #16940)
+ Reported by: jamicque ........ ................ ................
+
+2010-09-22 00:08 +0000 [r288195] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 288194 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r288194 | rmudgett | 2010-09-21 19:06:21 -0500
+ (Tue, 21 Sep 2010) | 40 lines Merged revisions 288193 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288193 | rmudgett | 2010-09-21 19:03:37 -0500
+ (Tue, 21 Sep 2010) | 33 lines Merged revisions 288192 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010)
+ | 26 lines In chan_iax2.c:schedule_delivery() calls
+ ast_bridged_channel() on an unlocked channel. Near the beginning
+ of schedule_delivery(), ast_bridged_channel() is called on
+ iaxs[fr->callno]->owner. However, the channel is not locked,
+ which can result in ast_bridged_channel() crashing should
+ owner->tech change to a technology that doesn't implement
+ bridged_channel. I also fixed the other calls to
+ ast_bridged_channel() in chan_iax2.c since the owner lock was not
+ held there either. Converted the existing channel deadlock
+ avoidance to use iax2_lock_owner(). Using the new function
+ simplified some awkward code. In the process of fixing the
+ locking on ast_bridged_channel(), I also found a memory leak in
+ socket_process() for v1.6.2 and v1.8. The local struct variable
+ ies.vars is not freed on early/abnormal function exits. (closes
+ issue #17919) Reported by: rain Patches: issue17919_v1.4.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch
+ uploaded by rmudgett (license 664) Review:
+ https://reviewboard.asterisk.org/r/926/ ........ ................
+ ................
+
+2010-09-21 22:58 +0000 [r288160] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, channels/chan_sip.c: Merged revisions 288159 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r288159 | tilghman | 2010-09-21 17:57:22 -0500
+ (Tue, 21 Sep 2010) | 29 lines Merged revisions 288113 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288113 | tilghman | 2010-09-21 16:59:46 -0500
+ (Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010)
+ | 15 lines Try both the encoded and unencoded subscription URI
+ for a match in hints. When a phone sends an encoded URI for a
+ subscription, the URI is not matched with the actual hint that is
+ in decoded format. For example, if we have an extension with a
+ hint that is named: "#5601" or "*5601", the subscription will
+ work fine if the phone subscribes with an already decoded URI,
+ but when it's decoded like "%255601" or "%2A5601", Asterisk is
+ unable to match it with the correct hint. (closes issue #17785)
+ Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt
+ uploaded by tilghman (license 14) Tested by: ramonpeek ........
+ ................ ................
+
+2010-09-21 22:28 +0000 [r288158] Paul Belanger <pabelanger@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 288157 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r288157 | pabelanger | 2010-09-21 18:26:15 -0400
+ (Tue, 21 Sep 2010) | 15 lines Merged revisions 288147 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue, 21 Sep
+ 2010) | 9 lines Setup timer before set_config(). (closes issue
+ #18019) Reported by: Netview Patches: issue_0018019.patch
+ uploaded by pabelanger (license 224) Tested by: Netview ........
+ ................
+
+2010-09-21 21:04 +0000 [r288081-288083] Richard Mudgett <rmudgett@digium.com>
+
+ * /, doc/tex/partymanip.tex: Merged revisions 288082 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r288082 | rmudgett | 2010-09-21 16:03:28 -0500 (Tue, 21
+ Sep 2010) | 1 line Add note in party manipulation chapter on
+ interception macros. ........
+
+ * apps/app_dial.c, main/channel.c, /, apps/app_queue.c: Merged
+ revisions 288079-288080 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r288079 | rmudgett | 2010-09-21 15:29:51 -0500 (Tue, 21 Sep 2010)
+ | 2 lines Protect channel access in CONNECTED_LINE and
+ REDIRECTING interception macro launch code. ........ r288080 |
+ rmudgett | 2010-09-21 15:29:59 -0500 (Tue, 21 Sep 2010) | 8 lines
+ Simplify locking code for REDIRECTING interception macro when
+ forwarding a call. Simplified the locking code by using a local
+ copy of the redirecting party information in
+ app_dial.c:do_forward() and app_queue.c:wait_for_answer() for
+ launching the REDIRECTING interception macro when a call is
+ forwarded. Reduced the lock time of the 'o->chan' and 'in'
+ channels. ........
+
+2010-09-21 20:27 +0000 [r288063] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: Instead of iterate through all dialogs, add
+ two separte container for needdestroy and rtptimeout adding two
+ dialog container, one for dialogs which need destroy, another for
+ rtptimeout checks. both container will be checked on every loop
+ of do_monitor instead of iterate through all dialogs. (closes
+ issue #17912) Reported by: schmidts Tested by: schmidts Review:
+ https://reviewboard.asterisk.org/r/917/
+
+2010-09-21 19:50 +0000 [r288008] Brett Bryant <bbryant@digium.com>
+
+ * main/channel.c, /: Merged revisions 288007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r288007 | bbryant | 2010-09-21 15:48:53 -0400
+ (Tue, 21 Sep 2010) | 21 lines Merged revisions 288006 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288006 | bbryant | 2010-09-21 15:46:20 -0400
+ (Tue, 21 Sep 2010) | 14 lines Merged revisions 288005 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010)
+ | 8 lines Add a check to fix a rare segmentation fault you'd get
+ if ast_frdup couldn't allocate memory on the first frame being
+ queued in ast_queue_frame. (closes issue #17882) Reported by:
+ seanbright Tested by: seanbright ........ ................
+ ................
+
+2010-09-21 19:09 +0000 [r287936] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/asterisk.c: Merged revisions 287935 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r287935 | tilghman | 2010-09-21 14:08:36 -0500
+ (Tue, 21 Sep 2010) | 16 lines Merged revisions 287934 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287934 | tilghman | 2010-09-21 14:07:53 -0500
+ (Tue, 21 Sep 2010) | 9 lines Merged revisions 287933 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21
+ Sep 2010) | 2 lines Less than zero is an error, not any non-zero
+ value. ........ ................ ................
+
+2010-09-21 19:04 +0000 [r287932] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Merged revisions 287931 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287931 | twilson | 2010-09-21 14:02:40 -0500 (Tue, 21 Sep 2010)
+ | 2 lines Revert change in favor of a more targeted fix ........
+
+2010-09-21 18:33 +0000 [r287930] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 287929 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287929 | dvossel | 2010-09-21 13:32:12 -0500 (Tue, 21 Sep 2010)
+ | 4 lines Send a "415 Unsupported Media Type" after failure to
+ process sdp due to unknown Content-Encoding header. ABE-2258
+ ........
+
+2010-09-21 15:54 +0000 [r287898] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 287897 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287897 | rmudgett | 2010-09-21 10:53:19 -0500 (Tue, 21 Sep 2010)
+ | 1 line Cut-n-paste error in builtin_blindtransfer(). ........
+
+2010-09-21 15:45 +0000 [r287896] Russell Bryant <russell@digium.com>
+
+ * res/res_rtp_asterisk.c, main/dnsmgr.c, /, channels/chan_sip.c,
+ main/acl.c: Merged revisions 287895 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287895 | russell | 2010-09-21 10:43:33 -0500 (Tue, 21 Sep 2010)
+ | 10 lines Don't use ast_strdupa() from within the arguments to a
+ function. (closes issue #17902) Reported by: afried Patches:
+ issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
+ russell Review: https://reviewboard.asterisk.org/r/927/ ........
+
+2010-09-21 15:27 +0000 [r287894] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, channels/chan_sip.c: Merged revisions 287893 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287893 | tilghman | 2010-09-21 10:24:47 -0500 (Tue, 21 Sep 2010)
+ | 9 lines Anonymous callerid needs a "sip:" uri prefix. (closes
+ issue #17981) Reported by: avalentin Patches:
+ sip-anonymous-aastra.patch uploaded by avalentin (license 1107)
+ (plus an additional fix by me) Tested by: avalentin ........
+
+2010-09-21 13:45 +0000 [r287864] Russell Bryant <russell@digium.com>
+
+ * /, main/logger.c: Merged revisions 287863 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287863 | russell | 2010-09-21 08:41:41 -0500 (Tue, 21 Sep 2010)
+ | 2 lines Fix a regression in verbose logger processing. ........
+
+2010-09-21 04:39 +0000 [r287764-287834] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Merged revisions 287833 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287833 | twilson | 2010-09-20 23:37:44 -0500 (Mon, 20 Sep 2010)
+ | 3 lines Don't generate connected line buffer twice for
+ comparison ........
+
+ * main/channel.c, /: Merged revisions 287757 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287757 | twilson | 2010-09-20 18:51:38 -0500 (Mon, 20 Sep 2010)
+ | 7 lines Avoid infinite loop with certain local channel
+ connected line updates Compare connected line data before sending
+ a connected line indication to avoid possible loops. Review:
+ https://reviewboard.asterisk.org/r/932/ ........
+
+2010-09-21 00:04 +0000 [r287763] Brett Bryant <bbryant@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 287760 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r287760 | bbryant | 2010-09-20 20:00:23 -0400
+ (Mon, 20 Sep 2010) | 30 lines Merged revisions 287759 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287759 | bbryant | 2010-09-20 19:58:26 -0400
+ (Mon, 20 Sep 2010) | 23 lines Merged revisions 287758 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010)
+ | 16 lines Fix misvalidation of meetme pins in conjunction with
+ the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a
+ user and admin pin setup for your conference, using the user pin
+ would gain you admin priviledges. Also, when no user pin was set,
+ an admin pin was, the 'a' MeetMe flag wasn't used, and the user
+ tried to enter a conference then they were still prompted for a
+ pin and forced to hit #. (closes issue #17908) Reported by: kuj
+ Patches: pins_2.patch uploaded by kuj (license 1111) Tested by:
+ kuj Review: [full review board URL with trailing slash] ........
+ ................ ................
+
+2010-09-21 00:01 +0000 [r287761-287762] Terry Wilson <twilson@digium.com>
+
+ * /: Add alecdavis' commit to merged props
+
+ * /: Add merge properties back.
+
+2010-09-20 23:42 +0000 [r287756] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/channel.c, /: Merged revisions 287685 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep
+ 2010) | 18 lines ast_channel_masquerade: Avoid recursive
+ masquerades. Check all 4 combinations of (original/clonechan) *
+ (masq/masqr). Initially original->masq and clonechan->masqr were
+ only checked. It's possible with multiple masq's planned - and
+ not yet executed, that the 'original' chan could already have
+ another masq'd into it - thus original->masqr would be set, that
+ masqr would lost. Likewise for the clonechan->masq. (closes issue
+ #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches:
+ based on bug16057.diff4.txt uploaded by alecdavis (license 585)
+ Tested by: ramonpeek, davidw, alecdavis ........
+
+2010-09-20 23:18 +0000 [r287693] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 287683 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r287683 | rmudgett | 2010-09-20 18:14:42 -0500 (Mon, 20
+ Sep 2010) | 9 lines The inalarm flag was not set in sig_analog
+ struct if the port is initially in alarm. Fixed initial inalarm
+ value for sig_analog ports. Along with -r261007, this gets the
+ inalarm flag in sync with chan_dahdi for sig_analog ports.
+ (closes issue #16983) ........
+
+2010-09-20 22:24 +0000 [r287671] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/channel.c, /: Merged revisions 287661 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287661 | alecdavis | 2010-09-21 10:21:50 +1200 (Tue, 21 Sep
+ 2010) | 14 lines ast_do_masquerade. Keep channels ao2_container
+ locked while unlink and linking channels. Previously, Masquerade
+ would unlock 'original' and 'clonechan' and allow another masq
+ thread to run. End result would be corrupted memory, and the
+ frequent report 'Bad Magic Number'. (closes issue #17801,#17710)
+ Reported by: notthematrix Patches: Based on bug17801.diff1.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/928 ........
+
+2010-09-20 22:16 +0000 [r287646-287648] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, main/framehook.c (added), /,
+ funcs/func_frame_trace.c (added), include/asterisk/channel.h,
+ CHANGES, include/asterisk/framehook.h (added): Merged revisions
+ 287647 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010)
+ | 21 lines Addition of the FrameHook API (AKA AwesomeHooks) So
+ far all our tools for viewing and manipulating media streams
+ within Asterisk have been entirely focused on audio. That made
+ sense then, but is not scalable now. The FrameHook API lets us
+ tap into and manipulate _ANY_ type of media or signaling passed
+ on a channel present today or in the future. This tool is a step
+ in the direction of expanding Asterisk's boundaries and will help
+ generate some rather interesting applications in the future. In
+ addition to the FrameHook API, a simple dialplan function
+ exercising the api has been included as well. This function is
+ called FRAME_TRACE(). FRAME_TRACE() allows for the internal
+ ast_frames read and written to a channel to be output. Filters
+ can be placed on this function to debug only certain types of
+ frames. This function could be thought of as an internal way of
+ doing ast_frame packet captures. Review:
+ https://reviewboard.asterisk.org/r/925/ ........
+
+ * /, channels/chan_sip.c: Merged revisions 287645 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287645 | dvossel | 2010-09-20 16:34:15 -0500 (Mon, 20 Sep 2010)
+ | 9 lines Fixes issue with registrations not working properly
+ with pedantic=yes. (closes issue #18017) Reported by: schmidts
+ Patches: issues_18017_v1.diff uploaded by dvossel (license 671)
+ Tested by: schmidts ........
+
+2010-09-20 21:30 +0000 [r287644] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_skinny.c: Merged revisions 287643 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r287643 | qwell | 2010-09-20 16:29:46 -0500
+ (Mon, 20 Sep 2010) | 15 lines Merged revisions 287642 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r287642 | qwell | 2010-09-20 16:28:32 -0500 (Mon, 20 Sep 2010) |
+ 8 lines Don't crash when parking a non-bridged call. (closes
+ issue #17680) Reported by: jmhunter Patches:
+ chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by:
+ jmhunter, DEA ........ ................
+
+2010-09-20 21:25 +0000 [r287640] Brett Bryant <bbryant@digium.com>
+
+ * /, main/logger.c: Merged revisions 287639 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287639 | bbryant | 2010-09-20 17:19:12 -0400 (Mon, 20 Sep 2010)
+ | 8 lines Fixes an error with the logger that caused verbose
+ messages to be spammed to the screen if syslog was configured in
+ logger.conf (closes issue #17974) Reported by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/915/ ........
+
+2010-09-20 15:57 +0000 [r287560] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 287559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r287559 | mnicholson | 2010-09-20 10:57:14 -0500
+ (Mon, 20 Sep 2010) | 21 lines Merged revisions 287558 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287558 | mnicholson | 2010-09-20 10:56:21 -0500
+ (Mon, 20 Sep 2010) | 14 lines Use ast_str when processing hint
+ state changes Merged revisions 287555 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
+ 2010) | 5 lines Use ast_dynamic_str when processing hint state
+ changes (related to issue #17928) Reported by: mdu113 ........
+ ................ ................
+
+2010-09-19 16:12 +0000 [r287472] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c, /: Merged revisions 287471 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r287471 | oej | 2010-09-19 18:09:28 +0200 (Sön,
+ 19 Sep 2010) | 21 lines Merged revisions 287470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287470 | oej | 2010-09-19 18:06:10 +0200 (Sön,
+ 19 Sep 2010) | 14 lines Merged revisions 287469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7
+ lines Make sure we always free variables properly in manager
+ originate. (closes issue #17891) reported, solved and tested by
+ oej Review: https://reviewboard.asterisk.org/r/869/ ........
+ ................ ................
+
+2010-09-17 21:10 +0000 [r287389] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, apps/app_queue.c: Merged revisions 287388 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r287388 | tilghman | 2010-09-17 16:08:54 -0500
+ (Fri, 17 Sep 2010) | 21 lines Merged revisions 287387 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287387 | tilghman | 2010-09-17 16:08:00 -0500
+ (Fri, 17 Sep 2010) | 14 lines Merged revisions 287386 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010)
+ | 7 lines Blank columns should get set on reload, not ignored.
+ (closes issue #16893) Reported by: haakon Patches:
+ 20100818__issue16893.diff.txt uploaded by tilghman (license 14)
+ ........ ................ ................
+
+2010-09-17 13:38 +0000 [r287310] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 287309 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r287309 | mnicholson | 2010-09-17 08:37:10 -0500
+ (Fri, 17 Sep 2010) | 19 lines Merged revisions 287308 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287308 | mnicholson | 2010-09-17 08:36:07 -0500
+ (Fri, 17 Sep 2010) | 12 lines Merged revisions 287307 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep
+ 2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while
+ processing in ast_hint_state_changed(). (related to issue #17928)
+ Reported by: mdu113 ........ ................ ................
+
+2010-09-17 08:46 +0000 [r287272] Jan Kalab <pitlicek@gmail.com>
+
+ * res/res_calendar_exchange.c, res/res_calendar_icalendar.c, /,
+ res/res_calendar_caldav.c, res/res_calendar_ews.c: Merged
+ revisions 287269-287271 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287269 | pitel | 2010-09-17 10:37:49 +0200 (Pá, 17 zář 2010) | 8
+ lines Support for HTTP redirects in calendar's URL libneon does
+ not support HTTP redirects (3xx responses) by default. You must
+ tell it to follow them. Also, another little unsigned int fix.
+ (closes issue #17776) Review:
+ https://reviewboard.asterisk.org/r/921/ ........ r287270 | pitel
+ | 2010-09-17 10:42:37 +0200 (Pá, 17 zář 2010) | 6 lines Asterisk
+ crashing because of double free when EWS request fails The free
+ is done later in code. I think ast_free() should have built in
+ checks for double free. (closes issue #17782) ........ r287271 |
+ pitel | 2010-09-17 10:44:28 +0200 (Pá, 17 zář 2010) | 6 lines
+ Events are visible after they were removed from EWS calendar
+ Because we must merge calendar even when it's empty. (closes
+ issue #17786) ........
+
+2010-09-16 22:05 +0000 [r287196] Jason Parker <jparker@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 287195 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287195 | qwell | 2010-09-16 17:04:38 -0500 (Thu, 16 Sep 2010) |
+ 7 lines Don't fail when running the Debian init script directly
+ (as one would normally do). readlink apparently returns 1 when
+ the arg isn't a symlink, which caused the script to exit. (closes
+ issue #17910) Reported by: wurstsalat ........
+
+2010-09-16 22:00 +0000 [r287194] Russell Bryant <russell@digium.com>
+
+ * /, configs/queues.conf.sample, apps/app_queue.c, UPGRADE-1.8.txt:
+ Merged revisions 287193 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287193 | russell | 2010-09-16 16:57:51 -0500 (Thu, 16 Sep 2010)
+ | 4 lines Set the default for "autofill" and "shared_lastcall" to
+ "yes" in queues.conf. Review:
+ https://reviewboard.asterisk.org/r/922/ ........
+
+2010-09-16 20:08 +0000 [r287117-287121] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 287120 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r287120 | mnicholson | 2010-09-16 15:07:38 -0500
+ (Thu, 16 Sep 2010) | 22 lines Merged revisions 287119 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287119 | mnicholson | 2010-09-16 15:06:16 -0500
+ (Thu, 16 Sep 2010) | 15 lines Merged revisions 287118 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep
+ 2010) | 8 lines Don't limit hint processing in
+ ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
+ (closes issue #17928) Reported by: mdu113 Patches:
+ 20100831__issue17928.diff.txt uploaded by tilghman (license 14)
+ Tested by: mdu113 ........ ................ ................
+
+ * main/cdr.c, /: Merged revisions 287116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r287116 | mnicholson | 2010-09-16 14:54:48 -0500
+ (Thu, 16 Sep 2010) | 22 lines Merged revisions 287115 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287115 | mnicholson | 2010-09-16 14:53:41 -0500
+ (Thu, 16 Sep 2010) | 15 lines Merged revisions 287114 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep
+ 2010) | 8 lines Don't stop printing cdr variables if we encounter
+ one with a blank name or value. (closes issue #17900) Reported
+ by: under Patches: core-show-channel-cdr-fix1.diff uploaded by
+ mnicholson (license 96) Tested by: mnicholson ........
+ ................ ................
+
+2010-09-16 16:49 +0000 [r287086-287087] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: We do not handle AST_CAUSE_INTERWORKING
+ which we set on a lot of incoming SIP messages. Adding error
+ based on RFC 3398 recommendations.
+
+ * main/indications.c: Add doxygen docs for indications.c
+
+2010-09-15 22:28 +0000 [r287057] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_srtp.c: Merged revisions 287056 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287056 | twilson | 2010-09-15 17:17:17 -0500 (Wed, 15 Sep 2010)
+ | 10 lines Don't hang up a call on an SRTP unprotect failure Also
+ make it more obvious when there is an issue en/decrypting.
+ (closes issue #17563) Reported by: Alexcr Patches:
+ res_srtp.c.patch uploaded by sfritsch (license 1089) Tested by:
+ twilson ........
+
+2010-09-15 21:00 +0000 [r287021] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 287020 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r287020 | jpeeler | 2010-09-15 15:58:39 -0500 (Wed, 15 Sep 2010)
+ | 1 line fix uninintialized variable ........
+
+2010-09-15 20:56 +0000 [r287018] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
+ Merged revisions 287017 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500
+ (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed,
+ 15 Sep 2010) | 58 lines The handling of call transfer signaling
+ for mISDN PTMP is not fully implemented. The handling of call
+ transfer signaling for mISDN PTMP is not fully implemented. The
+ signaling of number updates with ISDN/DSS1 ECT supplementary
+ services (ETS 300 369-1) comes along with a notification
+ indicator IE and redirection number IE for PTMP. The
+ implementation in the current Asterisk mISDN channel
+ unfortunately can handle these information elements only in a
+ NOTIFY message. These information elements are also signaled in a
+ FACILTY message with a RequestSubaddress facility, when the
+ subscriber is already in the active state (see 9.2.4 and 9.2.5 of
+ ETS 300 369-1). ********** abe_2526_ast.patch * Added support to
+ handle the notification indicator IE and redirection number IE
+ with the RequestSubaddress facility. * Made
+ misdn_update_connected_line() send a NOTIFY message if Asterisk
+ originated the call and it is not connected yet. * Made
+ misdn_update_connected_line() send a FACILITY message if the call
+ is already connected. This patch requires the presence of the
+ associated mISDN patches to compile. I had to enhance mISDN to
+ allow the notification indicator IE and the redirection number IE
+ to be used with a FACILITY message. Earlier versions of the
+ Digium enhanced mISDN are no longer going to work. **********
+ abe_2526_misdn.patch * Made an incoming FACILITY message allow
+ the presence of the notification indicator IE and the redirection
+ number IE. ********** abe_2526_misdnuser_v3.patch * Added support
+ to send and receive a FACILITY message with the notification
+ indicator IE and the redirection number IE. * Added the ability
+ to send a NOTIFY message in PTMP/NT mode to all responding
+ subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches:
+ abe_2526_ast.patch uploaded by rmudgett (license 664)
+ abe_2526_misdn.patch uploaded by rmudgett (license 664)
+ abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664)
+ Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526
+ .......... ................
+
+2010-09-15 20:36 +0000 [r286939-287016] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 287015 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r287015 | jpeeler | 2010-09-15 15:32:52 -0500
+ (Wed, 15 Sep 2010) | 21 lines Merged revisions 286998 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286998 | jpeeler | 2010-09-15 15:28:02 -0500
+ (Wed, 15 Sep 2010) | 14 lines Merged revisions 286941 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010)
+ | 7 lines Ensure mailbox is not filled to capacity before doing
+ message forwarding. Specifically, before prompting to record a
+ prepended message the capacity is checked first. If the mailbox
+ is full the extension will be reprompted. ABE-2517 ........
+ ................ ................
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/chan_sip.c, main/features.c, CHANGES,
+ channels/chan_iax2.c, channels/sip/include/sip.h,
+ configs/features.conf.sample, channels/chan_mgcp.c,
+ include/asterisk/features.h: Merged revisions 286931 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15
+ Sep 2010) | 16 lines Add parking extension for non-default
+ parking lots. This is a new feature that allows for parking to
+ custom parking lots to be accessed directly, rather than with
+ channel variables or by changing the default parking lot. The
+ extension is set with the parkext option just as the default
+ parking lot is done. Also, the manager action has been updated to
+ optionally allow a specified parking lot. (closes issue #14882)
+ Reported by: vmikhnevych Patches: patch_14882.txt uploaded by
+ mnick (license 874) modified by me Review:
+ https://reviewboard.asterisk.org/r/884/ ........
+
+2010-09-15 18:30 +0000 [r286906] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_analog.c, /: Merged revisions 286904-286905 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r286904 | rmudgett | 2010-09-15 13:28:05 -0500 (Wed, 15 Sep 2010)
+ | 12 lines Unable to originate calls using E&M over T1. When
+ originating a call from Unit Under Test to Reference Unit using
+ E&M RBS signaling mode, I get the following warning message:
+ "Ring/Off-hook in strange state 3 on channel 1". Fixed the
+ sig_analog outgoing flag. It was never set when sig_analog was
+ extracted from chan_dahdi. JIRA SWP-2191 JIRA AST-408 ........
+ r286905 | rmudgett | 2010-09-15 13:29:21 -0500 (Wed, 15 Sep 2010)
+ | 1 line Simplify some code in sig_analog. ........
+
+2010-09-15 13:10 +0000 [r286869] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 286868 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r286868 | mnicholson | 2010-09-15 08:05:52 -0500 (Wed, 15 Sep
+ 2010) | 16 lines Set tohost to the domain specified in the
+ configuration file instead of the IP address of the host we are
+ calling. This fixes a regression introduced in r274783. (closes
+ issue #17960) Reported by: adriavidal Patches:
+ sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested
+ by: mich, mnicholson, adriavidal (closes issue #17676) Reported
+ by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: mnicholson ........
+
+2010-09-14 22:02 +0000 [r286835] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 286834 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r286834 | dvossel | 2010-09-14 16:57:35 -0500 (Tue, 14 Sep 2010)
+ | 2 lines Sets subscribed type for outgoing MWI subscriptions so
+ correct Event header is used. ........
+
+2010-09-14 19:29 +0000 [r286683-286759] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 286758 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r286758 | mnicholson | 2010-09-14 14:28:38 -0500
+ (Tue, 14 Sep 2010) | 27 lines Merged revisions 286757 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286757 | mnicholson | 2010-09-14 14:27:28 -0500
+ (Tue, 14 Sep 2010) | 20 lines Merged revisions 286756 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep
+ 2010) | 13 lines Don't clear the username from a realtime
+ database when a registration expires. Non-realtime chan_sip does
+ not clear the username from memory when a registration expiries
+ so realtime probably shouldn't either. (closes issue #17551)
+ Reported by: ricardolandim Patches:
+ reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license
+ 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson
+ (license 96) reg-expiry-username-1.8-fix1.diff uploaded by
+ mnicholson (license 96) reg-expiry-username-trunk-fix1.diff
+ uploaded by mnicholson (license 96) Tested by: ricardolandim,
+ mnicholson ........ ................ ................
+
+ * main/channel.c, /: Merged revisions 286682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r286682 | mnicholson | 2010-09-14 13:04:21 -0500
+ (Tue, 14 Sep 2010) | 21 lines Merged revisions 286681 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286681 | mnicholson | 2010-09-14 13:02:24 -0500
+ (Tue, 14 Sep 2010) | 14 lines Merged revisions 286679 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep
+ 2010) | 7 lines Only drop duplicate answer frames if the channel
+ is bridged. Back in r3710 ast_read() was modified to drop answer
+ frames on channels that were in the UP state. This modification
+ prevented bridges that were up before the answer from being
+ broken and reestablished by an ANSWER control frame. That change
+ also prevents pickup of channels called from the ast_dial
+ framework from working properly. The ast_dial framework expects
+ to see an ANSWER frame after dialing and the pickup code queues
+ one but ast_read() drops it. This new change only drops ANSWER
+ frames when the channel is bridged, allowing the answer queued by
+ the pickup code to properly pass through ast_read() on to the
+ ast_dial framework. ABE-2473 (related to issue #2342) ........
+ ................ ................
+
+2010-09-14 15:31 +0000 [r286648] Richard Mudgett <rmudgett@digium.com>
+
+ * /, doc/tex/channelvariables.tex, doc/tex/partymanip.tex: Merged
+ revisions 286647 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r286647 | rmudgett | 2010-09-14 10:30:49 -0500 (Tue, 14 Sep 2010)
+ | 1 line Corrected documented CONNECTED_LINE and REDIRECTING
+ party manipulation macro names. ........
+
+2010-09-14 06:58 +0000 [r286618] Jan Kalab <pitlicek@gmail.com>
+
+ * /, res/res_calendar_ews.c: Merged revisions 286617 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r286617 | pitel | 2010-09-14 08:55:44 +0200 (Út, 14 zář
+ 2010) | 7 lines Merging events for Exchange web service doesn't
+ work as expected, resulting in only one event in calendar The
+ solution is to use "global" counter of events, since we do new
+ requests for every event and calendar sync after every request.
+ So now we do sync only after last request. (closes issue #17877)
+ Review: https://reviewboard.asterisk.org/r/916/ ........
+
+2010-09-14 05:08 +0000 [r286529-286589] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, contrib/realtime/mysql/voicemail_messages.sql (added),
+ contrib/realtime/mysql/voicemail_data.sql (added): Merged
+ revisions 286588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r286588 | tilghman | 2010-09-14 00:07:16 -0500
+ (Tue, 14 Sep 2010) | 9 lines Merged revisions 286587 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14
+ Sep 2010) | 2 lines Add documentation on missing backend tables
+ for Voicemail ........ ................
+
+ * /, main/features.c: Merged revisions 286558 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r286558 | tilghman | 2010-09-13 18:50:34 -0500
+ (Mon, 13 Sep 2010) | 9 lines Merged revisions 286557 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r286557 | tilghman | 2010-09-13 18:48:51 -0500 (Mon, 13
+ Sep 2010) | 2 lines C precedence got me ........ ................
+
+ * /, main/features.c: Merged revisions 286528 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r286528 | tilghman | 2010-09-13 18:12:21 -0500
+ (Mon, 13 Sep 2010) | 9 lines Merged revisions 286527 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r286527 | tilghman | 2010-09-13 18:03:26 -0500 (Mon, 13
+ Sep 2010) | 2 lines Refactor conversion to ast_poll() to fix
+ callparking regression. ........ ................
+
+2010-09-13 22:13 +0000 [r286498] Russell Bryant <russell@digium.com>
+
+ * /, main/db.c: Merged revisions 286112 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r286112 | russell | 2010-09-10 15:31:58 -0500 (Fri, 10 Sep 2010)
+ | 9 lines Rate limit calls to fsync() to 1 per second after astdb
+ updates. Astdb was determined to be one of the most significant
+ bottlenecks in SIP registration processing. This patch improved
+ the speed of an astdb load test by 50000% (yes, Fifty-Thousand
+ Percent). On this particular load test setup, this doubled the
+ number of SIP registrations the server could handle. Review:
+ https://reviewboard.asterisk.org/r/825/ ........
+
+2010-09-13 19:40 +0000 [r286458] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 286457 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r286457 | qwell | 2010-09-13 14:40:05 -0500
+ (Mon, 13 Sep 2010) | 12 lines Merged revisions 286456 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) |
+ 5 lines Remove "Internal IP" from sip show settings, as it's not
+ at all useful to display. (closes issue #17840) Reported by: oej
+ ........ ................
+
+2010-09-13 15:53 +0000 [r286427] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/chan_dahdi.conf.sample, /: Merged revisions 286426 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r286426 | rmudgett | 2010-09-13 10:52:14 -0500 (Mon, 13 Sep 2010)
+ | 1 line Update chan_dahdi.conf.sample to reflect new libpri T309
+ default value. ........
+
+2010-09-11 17:35 +0000 [r286271-286342] Olle Johansson <oej@edvina.net>
+
+ * main/say.c, main/app.c: Whitespace cleanup
+
+ * main/features.c: Whitespace cleanup and reformatting with { and }
+
+ * /, main/file.c: Merged revisions 286270 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r286270 | oej | 2010-09-11 19:09:22 +0200 (Lör,
+ 11 Sep 2010) | 18 lines Merged revisions 286268 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör,
+ 11 Sep 2010) | 11 lines Merged revisions 286267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4
+ lines Handle error response when we can't make file compatible
+ Review: https://reviewboard.asterisk.org/r/911/ ........
+ ................ ................
+
+ * channels/chan_sip.c: Formatting changes.
+
+2010-09-10 22:15 +0000 [r286190] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /, funcs/func_channel.c,
+ include/asterisk/channel.h, include/asterisk/pbx.h,
+ include/asterisk/frame.h: Merged revisions 286189 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r286189 | twilson | 2010-09-10 17:04:53 -0500
+ (Fri, 10 Sep 2010) | 30 lines Merged revisions 286115 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286115 | twilson | 2010-09-10 15:35:25 -0500
+ (Fri, 10 Sep 2010) | 23 lines Merged revisions 286059 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010)
+ | 16 lines Inherit CHANNEL() writes to both sides of a Local
+ channel Having Local (/n) channels as queue members and setting
+ the language in the extension with Set(CHANNEL(language)=fr) sets
+ the language on the Local/...,2 channel. Hold time report
+ playbacks happen on the Local/...,1 channel and therefor do not
+ play in the specified language. This patch modifies
+ func_channel_write to call the setoption callback and pass the
+ CHANNEL() write info to the callback. chan_local uses this
+ information to look up the other side of the channel and apply
+ the same changes to it. (closes issue #17673) Reported by:
+ Guggemand Review: https://reviewboard.asterisk.org/r/903/
+ ........ ................ ................
+
+2010-09-10 21:13 +0000 [r286121] Paul Belanger <pabelanger@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 286120 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r286120 | pabelanger | 2010-09-10 17:11:08 -0400
+ (Fri, 10 Sep 2010) | 18 lines Merged revisions 286117 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286117 | pabelanger | 2010-09-10 16:55:06 -0400
+ (Fri, 10 Sep 2010) | 11 lines Merged revisions 286114 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep
+ 2010) | 4 lines Load iax.conf before registering any
+ functions/applications/actions. Review:
+ https://reviewboard.asterisk.org/r/914/ ........ ................
+ ................
+
+2010-09-10 21:03 +0000 [r286119] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 286118 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r286118 | rmudgett | 2010-09-10 15:55:37 -0500
+ (Fri, 10 Sep 2010) | 25 lines Merged revisions 286116 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286116 | rmudgett | 2010-09-10 15:42:44 -0500
+ (Fri, 10 Sep 2010) | 18 lines Merged revisions 286113 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010)
+ | 11 lines An outgoing call may not get hung up if a pre-connect
+ incoming ISDN call is disconnected. If the ISDN link a
+ pre-connect incoming call is using fails or is reset, the
+ outgoing leg may not hang up or be delayed in hanging up.
+ (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER,
+ PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
+ PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the
+ incoming call leg hangs up before connecting for any reason. It
+ makes no sense to send a BUSY or CONGESTION control frame to the
+ outgoing call leg under these circumstances. ........
+ ................ ................
+
+2010-09-10 13:20 +0000 [r285993] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt, CHANGES: Merged revisions 285992 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r285992 | diruggles | 2010-09-10 09:13:16 -0400 (Fri, 10 Sep
+ 2010) | 1 line Added missing documentation for ExternalIVR
+ feature added in January 2010 ........
+
+2010-09-10 05:33 +0000 [r285932-285963] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * include/asterisk/select.h, /: Merged revisions 285962 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285962 | tilghman | 2010-09-10 00:32:18 -0500
+ (Fri, 10 Sep 2010) | 13 lines Merged revisions 285961 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010)
+ | 6 lines Another fix for Mac OS X. While trying to fix this the
+ "right" way, I wandered into dependency hell. Two hours later, I
+ backed out, and just removed the offending code. ast_inline_api
+ only goes one level deep and then it breaks. Ouch. ........
+ ................
+
+ * include/asterisk/select.h, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ tests/test_poll.c: Merged revisions 285931 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285931 | tilghman | 2010-09-09 20:25:50 -0500
+ (Thu, 09 Sep 2010) | 21 lines Merged revisions 285930 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285930 | tilghman | 2010-09-09 20:16:32 -0500
+ (Thu, 09 Sep 2010) | 14 lines Merged revisions 285889 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010)
+ | 7 lines Fix Mac OS X build. This also fixes a rather grievous
+ calculation error for the offset of ast_fdset, which was masked
+ on Linux and FreeBSD, because these platforms check the first 256
+ FDs regardless of the bitmask setting (due to backwards
+ compatibility). ........ ................ ................
+
+2010-09-09 22:53 +0000 [r285820] Paul Belanger <pabelanger@digium.com>
+
+ * /, codecs/gsm/Makefile: Merged revisions 285819 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285819 | pabelanger | 2010-09-09 18:52:31 -0400
+ (Thu, 09 Sep 2010) | 22 lines Merged revisions 285818 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285818 | pabelanger | 2010-09-09 18:49:19 -0400
+ (Thu, 09 Sep 2010) | 15 lines Merged revisions 285817 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep
+ 2010) | 8 lines GCC 4.2.x optimizations result in improper
+ behavior of GSM codec (closes issue #17688) Reported by:
+ pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by
+ pprindeville (license 347) Tested by: mkeuter, pprindeville
+ ........ ................ ................
+
+2010-09-09 20:13 +0000 [r285746] Jason Parker <jparker@digium.com>
+
+ * main/channel.c, /: Merged revisions 285745 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285745 | qwell | 2010-09-09 15:11:06 -0500
+ (Thu, 09 Sep 2010) | 23 lines Merged revisions 285744 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285744 | qwell | 2010-09-09 15:09:23 -0500
+ (Thu, 09 Sep 2010) | 16 lines Merged revisions 285742 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) |
+ 9 lines Transmit silence when reading DTMF in ast_readstring.
+ Otherwise, you could get issues with DTMF timeouts causing
+ hangups. (closes issue #17370) Reported by: makoto Patches:
+ channel-readstring-silence-generator.patch uploaded by makoto
+ (license 38) ........ ................ ................
+
+2010-09-09 18:53 +0000 [r285641-285712] Brett Bryant <bbryant@digium.com>
+
+ * main/pbx.c, /: Merged revisions 285711 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285711 | bbryant | 2010-09-09 14:51:52 -0400
+ (Thu, 09 Sep 2010) | 15 lines Merged revisions 285710 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010)
+ | 8 lines Fixes an issue with dialplan pattern matching where the
+ specificity for pattern ranges and pattern special characters was
+ inconsistent. (closes issue #16903) Reported by: Nick_Lewis
+ Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license
+ 657) Tested by: Nick_Lewis ........ ................
+
+ * /, res/res_musiconhold.c: Merged revisions 285640 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285640 | bbryant | 2010-09-09 13:23:28 -0400
+ (Thu, 09 Sep 2010) | 21 lines Merged revisions 285639 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285639 | bbryant | 2010-09-09 13:22:25 -0400
+ (Thu, 09 Sep 2010) | 14 lines Merged revisions 285638 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09 Sep 2010)
+ | 7 lines Fixes an issue with MOH where it doesn't recover
+ cleanly when it can't play a file and would just stop, instead of
+ continuing to find the next playable file in the MOH class.
+ (closes issue #17807) Reported by: kshumard Review:
+ https://reviewboard.asterisk.org/r/910/ ........ ................
+ ................
+
+2010-09-08 22:15 +0000 [r285565-285569] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 285568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285568 | dvossel | 2010-09-08 17:14:19 -0500
+ (Wed, 08 Sep 2010) | 16 lines Merged revisions 285567 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285567 | dvossel | 2010-09-08 17:11:28 -0500
+ (Wed, 08 Sep 2010) | 9 lines Merged revisions 285566 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08
+ Sep 2010) | 2 lines In retrans_pkt, do not unlock pvt until the
+ end of the function on a transmit failure. ........
+ ................ ................
+
+ * /, channels/chan_sip.c: Merged revisions 285564 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285564 | dvossel | 2010-09-08 16:48:37 -0500
+ (Wed, 08 Sep 2010) | 60 lines Merged revisions 285563 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010)
+ | 54 lines Fixes interoperability problems with session timer
+ behavior in Asterisk. CHANGES: 1. Never put "timer" in "Require"
+ header. This is not to our benefit and RFC 4028 section 7.1 even
+ warns against it. It is possible for one endpoint to perform
+ session-timer refreshes while the other endpoint does not support
+ them. If in this case the end point performing the refreshing
+ puts "timer" in the Require field during a refresh, the dialog
+ will likely get terminated by the other end. 2. Change the
+ behavior of 'session-timer=accept' in sip.conf (which is the
+ default behavior of Asterisk with no session timer configuration
+ specified) to only run session-timers as result of an incoming
+ INVITE request if the INVITE contains an "Session-Expires"
+ header... Asterisk is currently treating having the "timer"
+ option in the "Supported" header as a request for session timers
+ by the UAC. I do not agree with this. Session timers should only
+ be negotiated in "accept" mode when the incoming INVITE supplies
+ a "Session-Expires" header, otherwise RFC 4028 says we should
+ treat a request containing no "Session-Expires" header as a
+ session with no expiration. Below I have outlined some situations
+ and what Asterisk's behavior is. The table reflects the behavior
+ changes implemented by this patch. SITUATIONS: -Asterisk as UAS
+ 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS
+ "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO
+ "Session-Expires". 200 Ok Response HAS "Session-Expires" header
+ 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO
+ "Session-Expires" header 5. Outgoing INVITE: HAS
+ "Session-Expires". Active - Asterisk will have an active refresh
+ timer regardless if the other endpoint does. Inactive - Asterisk
+ does not have an active refresh timer regardless if the other
+ endpoint does. XXXXXXX - Not possible for mode.
+ ______________________________________ |SITUATIONS |
+ 'session-timer' MODES | |___________|________________________| |
+ | originate | accept | |-----------|------------|-----------| |1.
+ | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX |
+ Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX |
+ -------------------------------------- (closes issue #17005)
+ Reported by: alexrecarey ........ ................
+
+2010-09-08 21:00 +0000 [r285534] Brett Bryant <bbryant@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 285533 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285533 | bbryant | 2010-09-08 16:58:43 -0400
+ (Wed, 08 Sep 2010) | 15 lines Merged revisions 285532 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010)
+ | 8 lines Fixes a bug with MeetMe where after announcing the
+ amount of time left in a conference, if music on hold was
+ playing, it doesn't restart. (closes issue #17408) Reported by:
+ sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by
+ sysreq (license 1009) Tested by: sysreq ........ ................
+
+2010-09-08 20:43 +0000 [r285528-285531] Jason Parker <jparker@digium.com>
+
+ * /, include/asterisk/astobj2.h, res/res_musiconhold.c: Merged
+ revisions 285530 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285530 | qwell | 2010-09-08 15:43:10 -0500
+ (Wed, 08 Sep 2010) | 9 lines Merged revisions 285529 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r285529 | qwell | 2010-09-08 15:42:44 -0500 (Wed, 08 Sep
+ 2010) | 1 line Follow coding guidelines in moh rescan fix. Also
+ fix the documentation that got me in trouble. ........
+ ................
+
+ * /, res/res_musiconhold.c: Merged revisions 285527 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285527 | qwell | 2010-09-08 15:32:13 -0500
+ (Wed, 08 Sep 2010) | 15 lines Merged revisions 285526 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285526 | qwell | 2010-09-08 15:31:43 -0500 (Wed, 08 Sep 2010) |
+ 8 lines Fixes issue where moh files were no longer rescanned
+ during a reload. (closes issue #16744) Reported by: pj Patches:
+ 16744-reload.diff uploaded by qwell (license 4) Tested by: qwell
+ ........ ................
+
+2010-09-08 07:15 +0000 [r285485] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, funcs/func_channel.c: Merged revisions 285484 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r285484 | tilghman | 2010-09-08 02:14:17 -0500 (Wed, 08
+ Sep 2010) | 2 lines Documentation only ........
+
+2010-09-07 22:23 +0000 [r285394-285456] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 285455 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r285455 | qwell | 2010-09-07 17:22:14 -0500 (Tue, 07 Sep 2010) |
+ 8 lines Don't automatically add domains for wildcard bindaddrs.
+ (closes issue #17832) Reported by: oej Patches:
+ 17832-wildcard.diff uploaded by qwell (license 4) Tested by:
+ qwell ........
+
+ * /, channels/chan_sip.c: Merged revisions 285369 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r285369 | qwell | 2010-09-07 15:58:34 -0500 (Tue, 07 Sep 2010) |
+ 7 lines Add note to 'sip show settings' regarding dual-stack
+ support, and a :: bindaddress. (closes issue #17831) Reported by:
+ oej Patches: 17831-v6wildcardbind.diff uploaded by qwell (license
+ 4) ........
+
+2010-09-07 21:21 +0000 [r285374-285390] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * pbx/pbx_spool.c, /: Merged revisions 285386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r285386 | tilghman | 2010-09-07 16:20:16 -0500 (Tue, 07 Sep 2010)
+ | 13 lines Don't notify on attribute changes, and change how the
+ queuing mechanism works. Fixes call spools in 1.8. (closes issue
+ #17337) Reported by: loloski Patches:
+ 20100827__issue17337.diff.txt uploaded by tilghman (license 14)
+ (closes issue #17924) Reported by: mkeuter Tested by: mkeuter
+ ........
+
+ * /, funcs/func_channel.c: Merged revisions 285373 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r285373 | tilghman | 2010-09-07 16:14:03 -0500 (Tue, 07
+ Sep 2010) | 7 lines Add CHANNEL(checkhangup) to check whether a
+ channel is in the process of being hanged up. (closes issue
+ #17652) Reported by: kobaz Patches: func_channel.patch uploaded
+ by kobaz (license 834) ........
+
+2010-09-07 21:12 +0000 [r285372] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 285371 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r285371 | rmudgett | 2010-09-07 16:08:35 -0500 (Tue, 07 Sep 2010)
+ | 1 line Fix cut-n-paste error. ........
+
+2010-09-07 20:56 +0000 [r285269-285368] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, pbx/pbx_config.c: Merged revisions 285367 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285367 | tilghman | 2010-09-07 15:56:07 -0500
+ (Tue, 07 Sep 2010) | 23 lines Merged revisions 285366 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285366 | tilghman | 2010-09-07 15:31:41 -0500
+ (Tue, 07 Sep 2010) | 16 lines Merged revisions 285365 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010)
+ | 9 lines Catch invalid extensions at the parser, instead of
+ making the core deal with them. (closes issue #17794) Reported
+ by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded
+ by tilghman (license 14) 20100820__issue17794__1.4.diff.txt
+ uploaded by tilghman (license 14) Tested by: PavelL ........
+ ................ ................
+
+ * /, include/asterisk/compiler.h, addons/ooh323c/src/ooSocket.h:
+ Merged revisions 285336 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r285336 | tilghman | 2010-09-07 14:38:12 -0500 (Tue, 07 Sep 2010)
+ | 2 lines Fix build on FreeBSD 8.0, take 2. ........
+
+ * /, main/poll.c: Merged revisions 285268 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285268 | tilghman | 2010-09-07 14:08:09 -0500
+ (Tue, 07 Sep 2010) | 18 lines Merged revisions 285267 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285267 | tilghman | 2010-09-07 14:07:17 -0500
+ (Tue, 07 Sep 2010) | 11 lines Merged revisions 285266 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010)
+ | 4 lines Use poll, if indicated to do so, in the ast_poll2
+ implementation. This fixes the unit tests on FreeBSD 8.0.
+ ........ ................ ................
+
+2010-09-07 17:57 +0000 [r285199] Brett Bryant <bbryant@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 285197 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285197 | bbryant | 2010-09-07 13:54:21 -0400
+ (Tue, 07 Sep 2010) | 24 lines Merged revisions 285196 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285196 | bbryant | 2010-09-07 13:49:07 -0400
+ (Tue, 07 Sep 2010) | 17 lines Merged revisions 285194 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010)
+ | 10 lines Fixes voicemail.conf issues where mailboxes with
+ passwords that don't precede a comma would throw unnecessary
+ error messages. (closes issue #15726) Reported by: 298 Patches:
+ M15726.diff uploaded by junky (license 177) Tested by: junky
+ Review: [full review board URL with trailing slash] ........
+ ................ ................
+
+2010-09-07 17:55 +0000 [r285198] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 285195 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285195 | rmudgett | 2010-09-07 12:47:34 -0500
+ (Tue, 07 Sep 2010) | 20 lines Merged revisions 285193 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 285192 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3 ........
+ r285192 | rmudgett | 2010-09-07 11:58:57 -0500 (Tue, 07 Sep 2010)
+ | 8 lines COLP/CONP and chan_misdn missing update chan_misdn does
+ not update the caller id of the channel if a new connected number
+ or ECT-INFORM (w/ new peer number on call transfer) is received.
+ JIRA ABE-2502 JIRA SWP-2058 ........ ........ ................
+
+2010-09-06 20:10 +0000 [r285163] Russell Bryant <russell@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 285161-285162 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r285161 | russell | 2010-09-06 15:10:03 -0500 (Mon, 06 Sep 2010)
+ | 4 lines Fix libsrtp -fPIC check for when non-standard prefix is
+ used. Thanks to loompek in #asterisk for reporting the issue and
+ testing this patch. ........ r285162 | russell | 2010-09-06
+ 15:10:24 -0500 (Mon, 06 Sep 2010) | 1 line regenerate configure
+ script. ........
+
+2010-09-06 06:57 +0000 [r285091] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, BSDmakefile (added), makeopts.in: Merged revisions 285090 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r285090 | tilghman | 2010-09-06 01:56:07 -0500
+ (Mon, 06 Sep 2010) | 16 lines Merged revisions 285089 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285089 | tilghman | 2010-09-06 01:55:17 -0500
+ (Mon, 06 Sep 2010) | 9 lines Merged revisions 285088 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06
+ Sep 2010) | 2 lines Silly convenience script for BSD platforms.
+ ........ ................ ................
+
+2010-09-04 18:10 +0000 [r285058] Russell Bryant <russell@digium.com>
+
+ * /, include/asterisk/cli.h: Merged revisions 285057 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r285057 | russell | 2010-09-04 13:08:19 -0500 (Sat, 04
+ Sep 2010) | 2 lines Add a C++ compatible version of
+ AST_CLI_DEFINE(). ........
+
+2010-09-03 23:23 +0000 [r285029] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 285017 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r285017 | twilson | 2010-09-03 18:19:54 -0500 (Fri, 03 Sep 2010)
+ | 4 lines Call correct lock function as transferer is a sip_pvt
+ not a channel Both functions are #defined to ao2_lock, but
+ still... ........
+
+2010-09-03 22:23 +0000 [r285007] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample,
+ channels/sip/include/sip.h: Merged revisions 285006 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r285006 | dvossel | 2010-09-03 17:21:50 -0500 (Fri, 03
+ Sep 2010) | 9 lines Disables auth_options_request option by
+ default. The auth_options_request option was created to do
+ authentication on OPTIONS request just like INVITES are done.
+ Since it has been noted that some endpoints use OPTIONS requests
+ as a way of qualifying a peer and that a 401 authentication
+ response could result in interoperability issues, this option has
+ been disabled by default. ........
+
+2010-09-03 18:21 +0000 [r284973] Brett Bryant <bbryant@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 284967 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r284967 | bbryant | 2010-09-03 14:19:53 -0400
+ (Fri, 03 Sep 2010) | 15 lines Merged revisions 284958 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r284958 | bbryant | 2010-09-03 14:15:49 -0400 (Fri, 03 Sep 2010)
+ | 8 lines This is a patch provided for issue #17935 to add the
+ ActionID to the IAXregistry AMI response. (closes issue #17935)
+ Reported by: alexkuklin Patches: iaxshowreg uploaded by
+ alexkuklin (license 1115) Tested by: alexkuklin ........
+ ................
+
+2010-09-03 18:04 +0000 [r284951-284953] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 284952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r284952 | dvossel | 2010-09-03 13:03:23 -0500 (Fri, 03 Sep 2010)
+ | 2 lines During OPTIONS authentication, the authpeer does not
+ need to be returned for any reason. ........
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
+ channels/sip/include/sip.h: Merged revisions 284950 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r284950 | dvossel | 2010-09-03 12:29:02 -0500 (Fri, 03
+ Sep 2010) | 14 lines authenticate OPTIONS requests just like we
+ would an INVITE OPTIONS requests should be treated the same as an
+ INVITE This includes authentication. This patch adds the ability
+ for incoming out of dialog OPTION requests to be authenticated
+ before providing a response indicating whether an extension is
+ available or not. The authentication routine works the exact same
+ way as it does for incoming INVITEs. This means that if a peer
+ has 'insecure=invite' in their peer definition, the same will be
+ true for the processing of the OPTIONS request. Review:
+ https://reviewboard.asterisk.org/r/881/ ........
+
+2010-09-03 16:42 +0000 [r284922] Terry Wilson <twilson@digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 284921 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r284921 | twilson | 2010-09-03 11:28:18 -0500
+ (Fri, 03 Sep 2010) | 19 lines Merged revisions 284897 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284897 | twilson | 2010-09-03 11:20:45 -0500
+ (Fri, 03 Sep 2010) | 12 lines Merged revisions 284881 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010)
+ | 5 lines Properly detect when a sound file doesn't exist
+ ast_fileexists returns -1 for error and 0 for a non-existant
+ file. The existing code treated missing files as though they
+ existed. ........ ................ ................
+
+2010-09-03 13:09 +0000 [r284851-284853] Jan Kalab <pitlicek@gmail.com>
+
+ * /, res/res_calendar_ews.c: Merge of strdupa() fix for calendars
+ categories priorities
+
+ * res/res_calendar_icalendar.c, /, res/res_calendar_caldav.c,
+ include/asterisk/calendar.h, res/res_calendar_ews.c,
+ res/res_calendar.c: Support for calendar events priorities and
+ categories (with ISO C90 fix) See RFC 5545 ch. 3.8.1.2 and 9.
+ (closes issue #17837) Review:
+ https://reviewboard.asterisk.org/r/880/
+
+2010-09-02 21:08 +0000 [r284782] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c:
+ Merged revisions 284779-284780 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r284779 | rmudgett | 2010-09-02 15:59:12 -0500 (Thu, 02 Sep 2010)
+ | 8 lines Made output libpri event names if pri debugging is
+ enabled when sig_pri processes them. * Simplified CLI "pri debug
+ xx span xx" command code and removed redundant debugging enabled
+ messages. * Made CLI "pri debug xx span xx" command only close
+ the debugging log file if it was opened. ........ r284780 |
+ rmudgett | 2010-09-02 16:02:54 -0500 (Thu, 02 Sep 2010) | 2 lines
+ Simplified pri_dchannel() poll timeout duration code. ........
+
+2010-09-02 16:57 +0000 [r284706] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 284705 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r284705 | dvossel | 2010-09-02 11:56:43 -0500
+ (Thu, 02 Sep 2010) | 20 lines Merged revisions 284704 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284704 | dvossel | 2010-09-02 11:48:51 -0500
+ (Thu, 02 Sep 2010) | 13 lines Merged revisions 284703 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010)
+ | 7 lines Removed relatedpeer code from sip_autodestruct Handling
+ of the relatedpeer structure associated with a sip_pvt should be
+ done during the final sip_destruction function, not in
+ sip_autodestruct. ........ ................ ................
+
+2010-09-02 16:44 +0000 [r284702] Jason Parker <jparker@digium.com>
+
+ * /, formats/format_wav.c: Merged revisions 284701 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r284701 | qwell | 2010-09-02 11:43:09 -0500 (Thu, 02 Sep
+ 2010) | 8 lines Add slin16 support for format_wav (new wav16 file
+ extension) (closes issue #15029) Reported by: andrew Patches:
+ wav16.patch uploaded by andrew (license 240) Tested by: qwell,
+ andrew ........
+
+2010-09-02 16:36 +0000 [r284700] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, addons/ooh323c/src/oochannels.c: Merged revisions 284696 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r284696 | tilghman | 2010-09-02 11:27:52 -0500 (Thu, 02 Sep 2010)
+ | 2 lines Fixing build ........
+
+2010-09-02 16:35 +0000 [r284699] Richard Mudgett <rmudgett@digium.com>
+
+ * /, doc/tex/channelvariables.tex, doc/tex/partymanip.tex (added),
+ doc/tex/asterisk.tex: Merged revisions 284698 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r284698 | rmudgett | 2010-09-02 11:34:32 -0500 (Thu, 02 Sep 2010)
+ | 5 lines Added documentation for CONNECTEDLINE and REDIRECTING
+ functions. (closes issue #17808) Reported by: jtodd Review:
+ https://reviewboard.asterisk.org/r/875/ ........
+
+2010-09-02 16:12 +0000 [r284598-284667] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * channels/chan_usbradio.c, /: Merged revisions 284666 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r284666 | tilghman | 2010-09-02 11:11:15 -0500
+ (Thu, 02 Sep 2010) | 9 lines Merged revisions 284665 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r284665 | tilghman | 2010-09-02 11:07:19 -0500 (Thu, 02
+ Sep 2010) | 2 lines Fixing build. ........ ................
+
+ * /, apps/app_queue.c: Merged revisions 284632 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r284632 | tilghman | 2010-09-02 00:31:02 -0500
+ (Thu, 02 Sep 2010) | 14 lines Merged revisions 284631 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010)
+ | 7 lines Don't reset queue stats on a module reload. (closes
+ issue #17535) Reported by: raarts Patches:
+ 20100819__issue17535.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+ * /, channels/chan_sip.c, channels/chan_agent.c,
+ channels/chan_iax2.c, apps/app_queue.c, apps/app_getcpeid.c,
+ apps/app_followme.c, apps/app_speech_utils.c, main/loader.c,
+ pbx/pbx_loopback.c, channels/chan_dahdi.c, funcs/func_aes.c,
+ include/asterisk/module.h, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
+ apps/app_stack.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
+ apps/app_voicemail.c: Merged revisions 284610 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010)
+ | 10 lines When optional_api is non-optional, force dependent
+ modules to be loaded. (closes issue #17707) Reported by: ira
+ Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman Review:
+ https://reviewboard.asterisk.org/r/876/ ........
+
+ * main/stun.c, res/res_ais.c, /, include/asterisk/autoconfig.h.in,
+ configure.ac, channels/console_video.c,
+ include/asterisk/channel.h, res/res_jabber.c, res/res_pktccops.c,
+ main/poll.c, channels/chan_usbradio.c, include/asterisk/select.h
+ (added), channels/chan_phone.c, channels/chan_misdn.c, configure,
+ main/features.c, include/asterisk/poll-compat.h,
+ tests/test_poll.c (added), addons/ooh323c/src/oochannels.c,
+ main/asterisk.c, addons/ooh323c/src/ooSocket.h: Merged revisions
+ 284597 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r284597 | tilghman | 2010-09-02 00:00:34 -0500
+ (Thu, 02 Sep 2010) | 29 lines Merged revisions 284593,284595 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284593 | tilghman | 2010-09-01 17:59:50 -0500
+ (Wed, 01 Sep 2010) | 18 lines Merged revisions 284478 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010)
+ | 11 lines Ensure that all areas that previously used select(2)
+ now use poll(2), with implementations that need poll(2)
+ implemented with select(2) safe against 1024-bit overflows. This
+ is a followup to the fix for the pthread timer in 1.6.2 and
+ beyond, fixing a potential crash bug in all supported releases.
+ (closes issue #17678) Reported by: russell Branch:
+ https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
+ Review: https://reviewboard.asterisk.org/r/824/ ........
+ ................ r284595 | tilghman | 2010-09-01 22:57:43 -0500
+ (Wed, 01 Sep 2010) | 2 lines Failed to rerun bootstrap.sh after
+ last commit ................ ................
+
+2010-09-01 21:48 +0000 [r284562] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 284561 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r284561 | dvossel | 2010-09-01 16:47:01 -0500 (Wed, 01 Sep 2010)
+ | 9 lines During request to dialog matching, verify init_ruri is
+ present before comparing. During request to dialog matching, we
+ attempt a best effort routine for fork detection which requires
+ several elements to be in place. The dialog's initial request uri
+ is one of those elements. Since it is best effort, if the
+ init_ruri is not present for some reason we can not proceed with
+ that routine. ........
+
+2010-09-01 18:52 +0000 [r284479] Terry Wilson <twilson@digium.com>
+
+ * res/res_rtp_asterisk.c, include/asterisk/res_srtp.h,
+ main/rtp_engine.c, /, channels/chan_sip.c, res/res_srtp.c: Merged
+ revisions 284477 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010)
+ | 17 lines Fix SRTP for changing SSRC and multiple a=crypto SDP
+ lines Adding code to Asterisk that changed the SSRC during
+ bridges and masquerades broke SRTP functionality. Also broken was
+ handling the situation where an incoming INVITE had more than one
+ crypto offer. This patch caches the SRTP policies the we use so
+ that we can change the ssrc and inform libsrtp of the new
+ streams. It also uses the first acceptable a=crypto line from the
+ incoming INVITE. (closes issue #17563) Reported by: Alexcr
+ Patches: srtp.diff uploaded by twilson (license 396) Tested by:
+ twilson Review: https://reviewboard.asterisk.org/r/878/ ........
+
+2010-09-01 18:19 +0000 [r284440-284474] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * res/res_config_pgsql.c, /: Merged revisions 284473 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r284473 | tilghman | 2010-09-01 13:16:37 -0500
+ (Wed, 01 Sep 2010) | 12 lines Merged revisions 284472 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r284472 | tilghman | 2010-09-01 13:13:35 -0500 (Wed, 01 Sep 2010)
+ | 5 lines Don't warn on floats and timestamps (closes issue
+ #17082) Reported by: coolmig ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 284415 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r284415 | tilghman | 2010-08-31 15:22:10 -0500
+ (Tue, 31 Aug 2010) | 21 lines Merged revisions 284399 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284399 | tilghman | 2010-08-31 15:18:32 -0500
+ (Tue, 31 Aug 2010) | 14 lines Merged revisions 284393 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010)
+ | 7 lines Don't send a devstate change on poke_noanswer if the
+ state did not change. (closes issue #17741) Reported by: schmidts
+ Patches: chan_sip.c.patch uploaded by schmidts (license 1077)
+ ........ ................ ................
+
+2010-08-31 19:01 +0000 [r284315-284319] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/say.conf.sample: Merged revisions 284318 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r284318 | lmadsen | 2010-08-31 14:00:15 -0500
+ (Tue, 31 Aug 2010) | 22 lines Merged revisions 284317 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284317 | lmadsen | 2010-08-31 13:59:31 -0500
+ (Tue, 31 Aug 2010) | 15 lines Merged revisions 284316 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010)
+ | 7 lines Update say.conf.sample to match the rules in say.c
+ (closes issue #17835) Reported by: RoadKill Patches:
+ say.conf.sample.patch.rules uploaded by RoadKill (license 933)
+ Tested by: RoadKill ........ ................ ................
+
+ * channels/chan_sip.c: Add trustrpid and sendrpid global values to
+ 'sip show settings' (closes issue #17860) Reported by: jtodd
+ Patches: __20100816-chan_sip-sip-show-settings.txt uploaded by
+ lmadsen (license 10) Tested by: lmadsen, russell
+
+2010-08-30 22:30 +0000 [r284282] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, apps/app_festival.c: Merged revisions 284281 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r284281 | tilghman | 2010-08-30 17:28:47 -0500
+ (Mon, 30 Aug 2010) | 18 lines Merged revisions 284280 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010)
+ | 11 lines Fix 3 coding errors: 1) After we close FD, we should
+ not be trying to write to it. 2) Call _exit(0), not exit(0), to
+ avoid running shutdown routines in a child. 3) Use endian, not
+ processor, detection to ensure bytes are written in the correct
+ order. (closes issue #15706) Reported by: modelnine Patches:
+ asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine
+ (license 865) Tested by: gmartinez ........ ................
+
+2010-08-30 09:32 +0000 [r284189-284248] Olle Johansson <oej@edvina.net>
+
+ * main/file.c: Small doxygen fix and doc addition
+
+ * main/say.c: Clean upp doxygen documentation
+
+ * include/asterisk/say.h: Doxygen formatting You can't write "same
+ as above" in hypertext documentation. Above doesn't make sense in
+ hyperspace.
+
+ * apps/app_playback.c: Add doxygen documentation
+
+2010-08-29 07:06 +0000 [r284097-284159] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, configs/res_curl.conf.sample (added): Merged revisions 284158
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r284158 | tilghman | 2010-08-29 02:05:27 -0500 (Sun, 29 Aug 2010)
+ | 2 lines Missed adding this file ........
+
+ * /, sounds: Merged revisions 284127 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r284127 | tilghman | 2010-08-29 00:17:37 -0500 (Sun, 29 Aug 2010)
+ | 2 lines Also ignore the checksums ........
+
+ * cel/cel_odbc.c (added), /, configs/cel_adaptive_odbc.conf.sample
+ (removed), configs/cel_odbc.conf.sample (added),
+ cel/cel_adaptive_odbc.c (removed): Merged revisions 284096 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r284096 | tilghman | 2010-08-28 21:51:14 -0500 (Sat, 28 Aug 2010)
+ | 3 lines Rename CEL adaptive driver to plain driver, since there
+ isn't another ODBC driver (and the other CEL drivers have
+ adaptive capabilities, anyway). ........
+
+2010-08-28 21:30 +0000 [r284066] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /: Merged revisions 284065 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r284065 | russell | 2010-08-28 16:29:45 -0500 (Sat, 28 Aug 2010)
+ | 13 lines Be more flexible with whitespace on AMI action
+ headers. Previously, this code required exactly one space to be
+ after the ':' in headers for an AMI action. This now makes
+ whitespace optional, and allows whitespace that is there to vary
+ in amount. (closes issue #17862) Reported by: cmoye Patches:
+ manager.c.patch_trunk uploaded by cmoye (license 858)
+ manager.c.patch_1.8 uploaded by cmoye (license 858) Tested by:
+ cmoye ........
+
+2010-08-27 22:39 +0000 [r284033] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 284032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r284032 | dvossel | 2010-08-27 17:37:11 -0500
+ (Fri, 27 Aug 2010) | 21 lines Merged revisions 284002 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284002 | dvossel | 2010-08-27 17:27:50 -0500
+ (Fri, 27 Aug 2010) | 14 lines Merged revisions 283960 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010)
+ | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests.
+ (closes issue #17758) Reported by: ibc Patches:
+ multiple_accept_headers_1.4.diff uploaded by dvossel (license
+ 671) ........ ................ ................
+
+2010-08-27 21:50 +0000 [r283958] Russell Bryant <russell@digium.com>
+
+ * /, pbx/pbx_realtime.c: Merged revisions 283951 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283951 | russell | 2010-08-27 16:33:55 -0500 (Fri, 27 Aug 2010)
+ | 2 lines Print exten@context:priority in verbose messages from
+ pbx_realtime. ........
+
+2010-08-27 20:32 +0000 [r283883] Jason Parker <jparker@digium.com>
+
+ * res/res_config_odbc.c, /, main/config.c,
+ addons/res_config_mysql.c: Merged revisions 283882 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r283882 | qwell | 2010-08-27 15:31:55 -0500
+ (Fri, 27 Aug 2010) | 22 lines Merged revisions 283881 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283881 | qwell | 2010-08-27 15:30:27 -0500
+ (Fri, 27 Aug 2010) | 15 lines Merged revisions 283880 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) |
+ 8 lines Fix issue with decoding ^-escaped characters in realtime.
+ (closes issue #17790) Reported by: denzs Patches:
+ 17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell,
+ denzs ........ ................ ................
+
+2010-08-27 14:01 +0000 [r283803] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c: Doxygen formatting changes
+
+2010-08-26 23:51 +0000 [r283771] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, res/res_musiconhold.c: Merged revisions 283770 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r283770 | tilghman | 2010-08-26 18:47:02 -0500 (Thu, 26
+ Aug 2010) | 8 lines Convert MOH to use generic timers. (closes
+ issue #17726) Reported by: lmadsen Patches:
+ 20100825__issue17726__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: tilghman ........
+
+2010-08-26 15:28 +0000 [r283693] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r283692 | dvossel | 2010-08-26 10:26:37 -0500
+ (Thu, 26 Aug 2010) | 32 lines Merged revisions 283691 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283691 | dvossel | 2010-08-26 10:24:40 -0500
+ (Thu, 26 Aug 2010) | 25 lines Merged revisions 283690 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010)
+ | 19 lines Fixed how Asterisk destroys a dialog on channel hangup
+ before invite receives a response. If an ast_channel with a SIP
+ tech pvt hangs up before the sip dialog gets a response to its
+ outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is
+ not rfc compliant and results in confusion at the other endpoint.
+ sip_pretend_ack will ack and remove all the packets in the
+ retransmit queue. This means that the INVITE will stop
+ retransmitting, and that any response to that INVITE that comes
+ after the pretend_ack occurs will be ignored. Instead of faking
+ any sort of acknowledgement for an outgoing INVITE during an
+ internal hangup, we should let the protocol stack process the
+ INVITE transaction and terminate the dialog properly. This is
+ achieved by setting the PENDING_BYE flag. When this flag is used,
+ once the dialog proceeds to an escapable state the transaction
+ will either be canceled with a SIP_CANCEL or completed followed
+ immediately by a BYE. Attempting to do this any other way is
+ incorrect. If the endpoint is not responding to the INVITE
+ request, the INVITE must continue to be retransmitted until it
+ times out which will result in the dialog being destroyed.
+ ........ ................ ................
+
+2010-08-26 13:28 +0000 [r283628-283660] Russell Bryant <russell@digium.com>
+
+ * /, res/res_odbc.c: Merged revisions 283659 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283659 | russell | 2010-08-26 08:26:14 -0500 (Thu, 26 Aug 2010)
+ | 2 lines Slight improvement to a debug message. ........
+
+ * Makefile, /, keys/iaxtel.pub (removed), keys/freeworlddialup.pub
+ (removed): Merged revisions 283629 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283629 | russell | 2010-08-26 07:48:45 -0500 (Thu, 26 Aug 2010)
+ | 2 lines Remove public keys that are no longer useful. ........
+
+ * /, configs/manager.conf.sample: Merged revisions 283627 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283627 | russell | 2010-08-26 07:26:22 -0500 (Thu, 26 Aug 2010)
+ | 2 lines Move httptimeout out from in between port and bindaddr.
+ ........
+
+2010-08-25 22:59 +0000 [r283596] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283595 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r283595 | dvossel | 2010-08-25 17:57:56 -0500
+ (Wed, 25 Aug 2010) | 14 lines Merged revisions 283594 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010)
+ | 7 lines Add to and from tags to NOTIFY dialog-info xml body so
+ pickup can occur. When pedantic mode is used, the dialog-info xml
+ generated during a ringing event must contain the to and from tag
+ values. Otherwise if a pickup occurs using INVITE with replaces,
+ Astrisk will not be able to locate the subscription. ........
+ ................
+
+2010-08-25 16:14 +0000 [r283562] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, res/res_odbc.c: Merged revisions 283561 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283561 | tilghman | 2010-08-25 11:12:43 -0500 (Wed, 25 Aug 2010)
+ | 5 lines Initialize connect timeout on each time through the
+ loop. (closes issue #17911) Reported by: wurstsalat ........
+
+2010-08-25 15:56 +0000 [r283560] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
+ revisions 283559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r283559 | dvossel | 2010-08-25 10:54:11 -0500
+ (Wed, 25 Aug 2010) | 16 lines Merged revisions 283558 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010)
+ | 10 lines Asterisk will not advertise session timers are
+ supported when 'session-timers=refuse' is used. Asterisk now
+ dynamically builds the "Supported" header depending on what is
+ enabled/disabled in sip.conf. Session timers used to always be
+ advertised as being supported even when they were disabled in the
+ configuration. This caused problems with some end points. (issue
+ #17005) ........ ................
+
+2010-08-25 14:55 +0000 [r283528] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283527 | russell | 2010-08-25 09:55:00 -0500 (Wed, 25 Aug 2010)
+ | 2 lines Convert ast_log(LOG_DEBUG, ...) to ast_debug(...)
+ ........
+
+2010-08-24 20:42 +0000 [r283495] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Ignore redial hard button when no
+ previous number. (closes issue #17887) Reported by: salecha
+ Patches: skinny.redial.diff uploaded by wedhorn (license 30)
+ Tested by: wedhorn, salecha
+
+2010-08-24 20:36 +0000 [r283494] David Vossel <dvossel@digium.com>
+
+ * /, configs/sip.conf.sample, UPGRADE-1.8.txt,
+ channels/sip/include/sip.h: Merged revisions 283493 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r283493 | dvossel | 2010-08-24 15:34:03 -0500 (Tue, 24
+ Aug 2010) | 2 lines Changes the default behavior for sip.conf's
+ pedantic option from "no" to "yes". ........
+
+2010-08-24 18:58 +0000 [r283458] Leif Madsen <lmadsen@digium.com>
+
+ * res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions
+ 283457 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283457 | lmadsen | 2010-08-24 13:56:29 -0500 (Tue, 24 Aug 2010)
+ | 9 lines Fix issue where TOS is no longer set on RTP packets.
+ Fix issue where the tos is no longer being set on RTP packets
+ through res_rtp_asterisk. (closes issue #17890) Reported by:
+ elguero Patches: qos_18.diff uploaded by elguero (license 37)
+ Review: https://reviewboard.asterisk.org/r/868 ........
+
+2010-08-24 18:45 +0000 [r283383-283456] David Vossel <dvossel@digium.com>
+
+ * res/res_stun_monitor.c: This fix downgrades the ERROR message
+ indicating no res_stun_monitor.conf to a WARNING message.
+
+ * /, channels/chan_sip.c: Merged revisions 283382 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r283382 | dvossel | 2010-08-24 11:11:18 -0500
+ (Tue, 24 Aug 2010) | 25 lines Merged revisions 283381 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283381 | dvossel | 2010-08-24 11:07:37 -0500
+ (Tue, 24 Aug 2010) | 18 lines Merged revisions 283380 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010)
+ | 11 lines This fix makes sure the ast_channel hangs up correctly
+ when the dialog's PENDING_BYE flag is set. When the pending bye
+ flag is used, it is possible that the dialog will terminate and
+ leave the sip_pvt->owner channel up. This is because we never
+ hangup the ast_channel after sending the SIP_BYE request. When we
+ receive the response for the SIP_BYE we set need_destroy which we
+ would expect to destroy the dialog on the next do_monitor loop,
+ but this is not the case. The dialog will only be destroyed once
+ the owner is hungup even with the need_destroy flag set. This
+ patch sets the softhangup flag on the ast_channel when a SIP_BYE
+ request is sent as a result of the pending bye flag. ........
+ ................ ................
+
+2010-08-24 12:51 +0000 [r283351] Russell Bryant <russell@digium.com>
+
+ * /, funcs/func_odbc.c: Merged revisions 283350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283350 | russell | 2010-08-24 07:49:41 -0500 (Tue, 24 Aug 2010)
+ | 2 lines Don't attempt to release a NULL ODBC handle. ........
+
+2010-08-23 21:35 +0000 [r283320] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c,
+ cel/cel_adaptive_odbc.c: Merged revisions 283319 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r283319 | tilghman | 2010-08-23 16:33:47 -0500
+ (Mon, 23 Aug 2010) | 9 lines Merged revisions 283318 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23
+ Aug 2010) | 2 lines CDR drivers depend upon res_odbc, not
+ directly on the ODBC libraries ........ ................
+
+2010-08-23 20:50 +0000 [r283287-283289] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Hack to allow easy debugging of skinny in
+ trunk.
+
+ * channels/chan_skinny.c: Add additional AST_CONTROL_ states to
+ control2str.
+
+ * channels/chan_skinny.c: Fixes display issues on 7910 and older
+ phones. Also correct the callinfo provided in skinny_answer.
+ (closes issue #17876) Reported by: salecha Patches:
+ skinny_cnd3.diff uploaded by wedhorn (license 30) Tested by:
+ salecha, wedhorn Review: NA
+
+2010-08-23 13:35 +0000 [r283178-283242] Russell Bryant <russell@digium.com>
+
+ * configs/cel.conf.sample, /: Merged revisions 283241 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r283241 | russell | 2010-08-23 08:35:35 -0500 (Mon, 23
+ Aug 2010) | 2 lines Add sample configuration for cel_radius.
+ ........
+
+ * /, include/asterisk/cel.h, main/cel.c: Merged revisions 283230
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283230 | russell | 2010-08-23 08:23:12 -0500 (Mon, 23 Aug 2010)
+ | 7 lines Make the AST_CEL_AMA enum match up with the AST_CDR_
+ ama flag values. Really, having 2 enums for this is silly and
+ error prone, demonstrated by the crash that I hit because there
+ was an assumption in the code that the values in each matched up.
+ However, this is a quick fix to get them to match up so it will
+ work. ........
+
+ * /, main/cel.c: Merged revisions 283209 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283209 | russell | 2010-08-23 08:06:57 -0500 (Mon, 23 Aug 2010)
+ | 2 lines Don't blow up on an invalid AMA flag. ........
+
+ * /, configs/cel_custom.conf.sample: Merged revisions 283207 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283207 | russell | 2010-08-23 07:31:20 -0500 (Mon, 23 Aug 2010)
+ | 2 lines Tack on ${eventextra} to the sample cel_custom.conf.
+ ........
+
+ * /, configs/cel_custom.conf.sample: Merged revisions 283177 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283177 | russell | 2010-08-23 07:12:53 -0500 (Mon, 23 Aug 2010)
+ | 2 lines Cut down on excessive quotation. ........
+
+2010-08-23 12:09 +0000 [r283176] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, res/res_stun_monitor.c: Merged revisions 283175 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r283175 | tilghman | 2010-08-23 07:06:26 -0500 (Mon, 23
+ Aug 2010) | 2 lines Don't fail to start if the config file is
+ missing. ........
+
+2010-08-23 11:59 +0000 [r283174] Russell Bryant <russell@digium.com>
+
+ * /, configs/cel_custom.conf.sample: Merged revisions 283173 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283173 | russell | 2010-08-23 06:58:34 -0500 (Mon, 23 Aug 2010)
+ | 5 lines Expand cel_custom.conf.sample. Include the usage of
+ CSV_QUOTE() to ensure data has valid CSV formatting. Also list
+ the special CEL variables that are available for use in the
+ mapping. ........
+
+2010-08-20 15:39 +0000 [r283051] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 283050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r283050 | rmudgett | 2010-08-20 10:35:38 -0500
+ (Fri, 20 Aug 2010) | 36 lines Merged revisions 283049 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283049 | rmudgett | 2010-08-20 10:31:03 -0500
+ (Fri, 20 Aug 2010) | 29 lines Merged revisions 283048 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010)
+ | 22 lines Q931 - Sending PROGRESS after sending ALERTING is a
+ protocol error The PRI layer in chan_dadhi will check if a
+ PROGRESS message has already been sent, and not allow sending
+ another (although that is technically allowed by the Q931 spec),
+ however it does not protect against sending an ALERTING and then
+ sending a PROGRESS message, which is a violation of the
+ specification. Most switches don't seem to care too deeply about
+ this, but some do, and will disconnect the call when receiving
+ this invalid sequence. Protocol specification reference:
+ T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
+ protocol control (network side) point-point (sheet 3 of 8)"
+ (closes issue #17874) Reported by: nic_bellamy Patches:
+ asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
+ nic bellamy (license 299)
+ asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299)
+ asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299) ........ ................
+ ................
+
+2010-08-20 12:45 +0000 [r282980-283015] Russell Bryant <russell@digium.com>
+
+ * /, configs/cel_adaptive_odbc.conf.sample: Merged revisions 283013
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r283013 | russell | 2010-08-20 07:45:12 -0500 (Fri, 20 Aug 2010)
+ | 2 lines Fix a typo in a column name. ........
+
+ * /, apps/app_celgenuserevent.c: Merged revisions 282979 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282979 | russell | 2010-08-20 06:52:37 -0500 (Fri, 20 Aug 2010)
+ | 2 lines Add an argument missing from the CELGenUserEvent
+ documentation. ........
+
+ * channels/chan_multicast_rtp.c, /: Merged revisions 282638 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282638 | russell | 2010-08-18 07:30:40 -0500 (Wed, 18 Aug 2010)
+ | 4 lines Split _all_ arguments before parsing them. This fixes
+ multicast RTP paging using linksys mode. ........
+
+2010-08-19 21:08 +0000 [r282892-282896] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282895 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r282895 | dvossel | 2010-08-19 16:07:20 -0500
+ (Thu, 19 Aug 2010) | 25 lines Merged revisions 282894 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282894 | dvossel | 2010-08-19 16:05:54 -0500
+ (Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010)
+ | 11 lines tos_sip option was not being set correctly When
+ tos_sip is used, the tos of the sip socket is only set correctly
+ if the socket binding changes on a reload. If the binding stays
+ the same but the TOS changes, the new tos value would not take
+ into effect. This patch fixes that. (closes issue #17712)
+ Reported by: nickb ........ ................ ................
+
+ * /, channels/chan_sip.c: Merged revisions 282891 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r282891 | dvossel | 2010-08-19 15:34:41 -0500
+ (Thu, 19 Aug 2010) | 11 lines Merged revisions 282890 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010)
+ | 5 lines fixes sip peer memory leaks in the peer_by_ip table
+ (issue #17798) ........ ................
+
+2010-08-19 20:02 +0000 [r282861] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282860 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r282860 | mnicholson | 2010-08-19 15:01:11 -0500
+ (Thu, 19 Aug 2010) | 30 lines Merged revisions 282859 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282859 | mnicholson | 2010-08-19 14:44:00 -0500
+ (Thu, 19 Aug 2010) | 23 lines Merged revisions 277944 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul
+ 2010) | 16 lines Regression with T.38 negotiation Prior to
+ 1.4.26.3 T.38 negotiation worked properly, in the case of the
+ reporter. (issue #16852) Reported by: cfc (closes issue #16705)
+ Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
+ by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
+ samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........
+ ................ ................
+
+2010-08-19 14:46 +0000 [r282827] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/netsock2.c, /: Merged revisions 282826 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282826 | tilghman | 2010-08-19 09:44:51 -0500 (Thu, 19 Aug 2010)
+ | 2 lines Only output debugging if the debug level is on.
+ ........
+
+2010-08-19 12:13 +0000 [r282798] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/cel.h: Add a todo item for CEL.
+
+2010-08-19 02:20 +0000 [r282751] Terry Wilson <twilson@digium.com>
+
+ * /, configs/sip.conf.sample: Merged revisions 282740 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r282740 | twilson | 2010-08-18 21:18:50 -0500
+ (Wed, 18 Aug 2010) | 16 lines Merged revisions 282730 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282730 | twilson | 2010-08-18 21:14:28 -0500
+ (Wed, 18 Aug 2010) | 9 lines Merged revisions 282729 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18
+ Aug 2010) | 2 lines Add some documentation about codec
+ negotiation to sip.conf ........ ................
+ ................
+
+2010-08-18 21:34 +0000 [r282701] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Cleanup: consolidate offhook (new call).
+ Consolidates all offhook (new call with dialtone) to
+ setsubstate_offhook. This should be roughly equivalent to
+ existing code, although a couple of calls now run through the
+ full offhook sequence rather than an abbreviated one. (closes
+ issue #17812) Reported by: wedhorn Patches:
+ cleanup.stateoffhook.diff uploaded by wedhorn (license 30) Tested
+ by: salecha, wedhorn Review: NA
+
+2010-08-18 15:35 +0000 [r282673] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, /: Merged revisions
+ 282671-282672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282671 | rmudgett | 2010-08-18 10:27:51 -0500 (Wed, 18 Aug 2010)
+ | 1 line Use the correct operator when calculating the PRI span
+ devstate. ........ r282672 | rmudgett | 2010-08-18 10:28:27 -0500
+ (Wed, 18 Aug 2010) | 1 line Use the correct type for
+ aoce_delayhangup bit field. ........
+
+2010-08-18 13:11 +0000 [r282640] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282639 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282639 | mnicholson | 2010-08-18 08:10:39 -0500 (Wed, 18 Aug
+ 2010) | 13 lines Properly handle 200 and unknown responses
+ conatined in NOTIFY requests received in response to REFER
+ requests. This patch fixes the way asterisk handles NOTIFY
+ requests received in response to REFER requests. These changes to
+ NOTIFY handler were first introduced in r217482. This new change
+ properly handles the 200 response by queueing an
+ AST_TRANSFER_SUCCESS control frame and also prevents that control
+ frame from being queued when provisional and unknown responses
+ are received. (issue #17486) Reported by: davidw Tested by:
+ mnicholson (issue #12713) Reported by: davidw Review:
+ https://reviewboard.asterisk.org/r/860/ ........
+
+2010-08-18 07:50 +0000 [r282609] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, channels/sig_pri.c: Merged revisions 282608 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r282608 | tilghman | 2010-08-18 02:49:04 -0500
+ (Wed, 18 Aug 2010) | 16 lines Merged revisions 282607 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010)
+ | 9 lines Don't warn on callerid when completely text, instead of
+ numeric with localdialplan prefixes. (closes issue #16770)
+ Reported by: jamicque Patches: 20100413__issue16770.diff.txt
+ uploaded by tilghman (license 14) 20100811__issue16770.diff.txt
+ uploaded by tilghman (license 14) Tested by: jamicque ........
+ ................
+
+2010-08-17 21:37 +0000 [r282544-282578] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r282577 | dvossel | 2010-08-17 16:36:57 -0500
+ (Tue, 17 Aug 2010) | 16 lines Merged revisions 282576 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010)
+ | 9 lines fixes no default transport for temp peer creation in
+ chan_sip (closes issue #17829) Reported by: falves11 Patches:
+ issue_17829.rev1.txt uploaded by russell (license 2)
+ issue_17829.diff uploaded by dvossel (license 671) Tested by:
+ falves11 ........ ................
+
+ * /, channels/chan_iax2.c: Merged revisions 282545 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r282545 | dvossel | 2010-08-17 15:08:56 -0500 (Tue, 17
+ Aug 2010) | 6 lines ACCEPT message should respond with the new
+ FORMAT2 ie (closes issue #17804) Reported by: tpanton ........
+
+ * /, include/asterisk/unaligned.h: Merged revisions 282543 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282543 | dvossel | 2010-08-17 14:34:06 -0500 (Tue, 17 Aug 2010)
+ | 4 lines fixes truncated uint64_t value in
+ put_unaligned_uint64_t() function (issue #17804) ........
+
+2010-08-16 20:40 +0000 [r282502] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Merged revisions 282468 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r282468 | twilson | 2010-08-16 12:53:44 -0500
+ (Mon, 16 Aug 2010) | 30 lines Merged revisions 282467 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282467 | twilson | 2010-08-16 12:32:01 -0500
+ (Mon, 16 Aug 2010) | 23 lines Merged revisions 282430 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010)
+ | 16 lines Send a SRCCHANGE indication when we masquerade
+ Masquerading a channel means that the src of the audio is
+ potentially changing, so send a SRCCHANGE so that RTP-based media
+ streams can get a new SSRC generated to reflect the change.
+ Original patch by addix (along with lots of testing--thanks!).
+ (closes issue #17007) Reported by: addix Patches:
+ 1001-reset-SSRC-original-channel.diff uploaded by addix (license
+ 1006) srcchange.diff uploaded by twilson (license 396) Tested by:
+ addix, twilson Review: https://reviewboard.asterisk.org/r/862/
+ ........ ................ ................
+
+2010-08-16 18:02 +0000 [r282471] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/sounds.tex (added), /, doc/tex/asterisk.tex: Merged
+ revisions 282470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r282470 | lmadsen | 2010-08-16 13:01:00 -0500
+ (Mon, 16 Aug 2010) | 15 lines Merged revisions 282469 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010)
+ | 7 lines Add information about creating sounds files using the
+ sounds tools publically available so that others can create their
+ own sounds prompts using the same tools we use to generate sounds
+ releases. This allows people creating their own prompts to sound
+ consistent with the prompts available from the open source
+ project. SWP-595 ........ ................
+
+2010-08-15 13:08 +0000 [r282397] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * utils/muted.c, configure, main/Makefile, configure.ac,
+ main/acl.c, channels/chan_oss.c, main/netsock.c: Support for
+ GNU/kFreeBSD kFreeBSD is GNU (with glibc) on to of a FreeBSD
+ kernel. See http://glibc-bsd.alioth.debian.org/porting/PORTING
+ This patch gets Asterisk close to building on Debian kFreeBSD
+ i386, mainly by adding an extra test for __GLIBC__ in one or two
+ (or more) places. OSARCH is set to 'kfreebsd-gnu' DAHDI support
+ (and support for chan_vpb) was not tested. Review:
+ https://reviewboard.asterisk.org/r/858/
+
+2010-08-14 04:58 +0000 [r282367] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, include/asterisk/sched.h, channels/chan_iax2.c: Merged
+ revisions 282366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282366 | tilghman | 2010-08-13 23:53:58 -0500 (Fri, 13 Aug 2010)
+ | 4 lines Fix our FRACKing issue with chan_iax2 a different way.
+ Review: https://reviewboard.asterisk.org/r/861/ ........
+
+2010-08-13 23:57 +0000 [r282335] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 282334 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r282334 | rmudgett | 2010-08-13 18:53:36 -0500 (Fri, 13
+ Aug 2010) | 6 lines PRI CCSS may use a stale dial string for the
+ recall dial string. If an outgoing call negotiates a different B
+ channel than initially requested, the saved original dial string
+ was not transferred to the new B channel. CCSS uses that dial
+ string to generate the recall dial string. ........
+
+2010-08-13 22:27 +0000 [r282237-282304] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
+ UPGRADE-1.8.txt: Merged revisions 282302 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010)
+ | 10 lines remove current STUN support from chan_sip.c This patch
+ removes the current broken/useless stun support from chan_sip.
+ (closes issue #17622) Reported by: philipp2 Review:
+ https://reviewboard.asterisk.org/r/855/ ........
+
+ * /, CHANGES: Merged revisions 282271 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282271 | dvossel | 2010-08-13 15:11:58 -0500 (Fri, 13 Aug 2010)
+ | 2 lines res_stun_monitor and corresponding options CHANGES
+ documentation ........
+
+ * configs/iax.conf.sample, /, channels/chan_sip.c,
+ include/asterisk/event_defs.h, res/res_stun_monitor.c (added),
+ configs/res_stun_monitor.conf.sample (added),
+ configs/sip.conf.sample, channels/chan_iax2.c: Merged revisions
+ 282269 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282269 | dvossel | 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010)
+ | 4 lines res_stun_monitor for monitoring network changes behind
+ a NAT device Review: https://reviewboard.asterisk.org/r/854
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 282236 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r282236 | dvossel | 2010-08-13 13:58:10 -0500
+ (Fri, 13 Aug 2010) | 23 lines Merged revisions 282235 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010)
+ | 16 lines only do magic pickup when notifycid is enabled A new
+ way of doing BLF pickup was introduced into 1.6.2. This feature
+ adds a call-id value into the XML of a SIP_NOTIFY message sent to
+ alert a subscriber that a device is ringing. This option should
+ only be enabled when the new 'notifycid' option is set... but
+ this was not the case. Instead the call-id value was included for
+ every RINGING Notify message, which caused a regression for
+ people who used other methods for call pickup. (closes issue
+ #17633) Reported by: urosh Patches: chan_sip.txt uploaded by
+ urosh (license ) blf_cid_issue.diff uploaded by dvossel (license
+ 671) Tested by: dvossel, urosh, okrief, alecdavis ........
+ ................
+
+2010-08-13 16:08 +0000 [r282202] Terry Wilson <twilson@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 282200-282201 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282200 | twilson | 2010-08-13 11:00:02 -0500 (Fri, 13 Aug 2010)
+ | 10 lines Detect when libsrtp cannot be linked in a shared
+ library The libsrtp build system currently does not produce a
+ shared library or a static library compiled with -fPIC, so on
+ 64-bit systems it is possible that we will get a compile error if
+ libsrtp is installed and res_srtp is selected in menuselect. This
+ patch attempts to detect this situation and provide the user with
+ instructions to work around the problem. ........ r282201 |
+ twilson | 2010-08-13 11:02:20 -0500 (Fri, 13 Aug 2010) | 2 lines
+ Whitespace fix :-/ ........
+
+2010-08-12 22:52 +0000 [r282132] Jason Parker <jparker@digium.com>
+
+ * /, pbx/pbx_config.c: Merged revisions 282131 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r282131 | qwell | 2010-08-12 17:51:44 -0500
+ (Thu, 12 Aug 2010) | 16 lines Merged revisions 282130 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282130 | qwell | 2010-08-12 17:50:54 -0500
+ (Thu, 12 Aug 2010) | 9 lines Merged revisions 282129 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug
+ 2010) | 1 line Register CLI commands before parsing config, in
+ case there is a config error. ........ ................
+ ................
+
+2010-08-12 22:10 +0000 [r282099] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/ccss.c, include/asterisk/ccss.h: Merged revisions 282098
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282098 | rmudgett | 2010-08-12 17:06:06 -0500 (Thu, 12 Aug 2010)
+ | 7 lines Separate call completion config parameter allocation
+ and default initialization. If you ever have a need to reset the
+ call completion config parameters to defaults, now you can. And
+ no Virginia, C++ idioms do not always work in C. ........
+
+2010-08-12 20:44 +0000 [r282067] Russell Bryant <russell@digium.com>
+
+ * /, CHANGES, main/cli.c: Merged revisions 282066 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282066 | russell | 2010-08-12 15:41:17 -0500 (Thu, 12 Aug 2010)
+ | 4 lines Add a "core reload" CLI command. Review:
+ https://reviewboard.asterisk.org/r/859/ ........
+
+2010-08-12 20:17 +0000 [r282048] David Vossel <dvossel@digium.com>
+
+ * /, main/translate.c, CHANGES, include/asterisk/translate.h,
+ main/cli.c: Merged revisions 282047 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010)
+ | 35 lines improved translation paths for wideband codecs The
+ problem I'm addressing is that Asterisk's current method of
+ building the least cost translation paths between codecs does not
+ take into account sample rate. For instance, it was possible for
+ siren14 (a 32khz codec), to contain the a translation path to
+ siren7 (a 16khz audio codec) that goes through slin at 8khz. In
+ this case Asterisk takes a 32khz codec, down samples it to 8khz
+ and then up samples it to 16khz which is terrible regardless if
+ it is computationally less expensive. This patch now builds
+ translation paths that give priority to maintaining the best
+ possible sample rate before taking into consideration
+ computational cost. This patch also adds cli commands to expose
+ what translation paths are actually being used. Changes: 1.
+ Translation paths will never contain a step that changes the
+ sample rate unless absolutely necessary. 2. When choosing the
+ best codec to make two channels compatible. Shared codecs with
+ the highest sample rate are given priority. 3. A new cli command
+ to show all translation paths available for a specific codec
+ 'core show translation paths [codec name]' has been added. 4.
+ 'core show translation' which displays the translation matrix now
+ includes the new higher bit audio codecs in the table. 5. 'core
+ show channel [channel name]' now displays the translation paths
+ if translation is used. (closes issue #16841) Reported by:
+ dvossel Review: https://reviewboard.asterisk.org/r/842/ ........
+
+2010-08-12 18:04 +0000 [r281983-282016] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, /: Merged revisions 282015 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r282015 | russell | 2010-08-12 13:03:56 -0500 (Thu, 12 Aug 2010)
+ | 2 lines Put back pointer value output for ast_debug(), such
+ that it is only removed for verbose output. ........
+
+ * main/pbx.c, /: Merged revisions 281982 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r281982 | russell | 2010-08-12 11:33:30 -0500 (Thu, 12 Aug 2010)
+ | 5 lines Remove debugging output from verbose messages. Pointer
+ values to internal objects is not terribly useful to users in the
+ verbose messages about adding extensions and contexts. ........
+
+2010-08-12 03:08 +0000 [r281914] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 281913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r281913 | jpeeler | 2010-08-11 22:03:37 -0500
+ (Wed, 11 Aug 2010) | 34 lines Merged revisions 281912 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281912 | jpeeler | 2010-08-11 22:01:38 -0500
+ (Wed, 11 Aug 2010) | 27 lines Merged revisions 281911 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010)
+ | 20 lines Ensure SSRC is changed when media source is changed to
+ resolve audio delay. This change causes the SSRC to change right
+ before the channels are bridged, which is what used to happen. It
+ seems that fixes were made to attempt limiting SSRC changes,
+ targeted mainly at sending DTMF. DTMF is not affecting the SSRC
+ with this change. There are two other control frames sent in
+ ast_channel_bridge that probably should also be changed to
+ AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change
+ up to the discretion of resolving issue #17007. For reference -
+ old review implementing new control frame SRCCHANGE:
+ https://reviewboard.asterisk.org/r/540 (closes issue #17404)
+ Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler
+ (license 325) Tested by: sdolloff ........ ................
+ ................
+
+2010-08-11 21:13 +0000 [r281877] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/say.conf.sample: Merged revisions 281875 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r281875 | lmadsen | 2010-08-11 16:12:13 -0500
+ (Wed, 11 Aug 2010) | 21 lines Merged revisions 281873 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281873 | lmadsen | 2010-08-11 16:09:47 -0500
+ (Wed, 11 Aug 2010) | 14 lines Merged revisions 281819 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010)
+ | 6 lines Add Danish support to say.conf.sample (closes issue
+ #17836) Reported by: RoadKill Patches: say.conf.sample.patch.dk
+ uploaded by RoadKill (license 933) ........ ................
+ ................
+
+2010-08-11 21:12 +0000 [r281876] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 281874 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r281874 | mnicholson | 2010-08-11 16:11:54 -0500 (Wed, 11 Aug
+ 2010) | 10 lines handle all possible responses to REFER requests
+ (closes issue #17486) Reported by: davidw Patches:
+ Issue17486-counterbid.diff.txt uploaded by davidw (license 780)
+ Tested by: davidw Review: https://reviewboard.asterisk.org/r/837/
+ ........
+
+2010-08-11 20:38 +0000 [r281871] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_analog.c, /, channels/sig_analog.h: Merged revisions
+ 281870 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r281870 | rmudgett | 2010-08-11 15:30:29 -0500 (Wed, 11 Aug 2010)
+ | 4 lines Fix a call to analog_set_pulsedial() not setting 0 or 1
+ only. * Also a couple minor tweaks. ........
+
+2010-08-11 17:55 +0000 [r281765] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/say.conf.sample: Merged revisions 281764 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r281764 | lmadsen | 2010-08-11 12:54:56 -0500
+ (Wed, 11 Aug 2010) | 21 lines Merged revisions 281763 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281763 | lmadsen | 2010-08-11 12:54:09 -0500
+ (Wed, 11 Aug 2010) | 14 lines Merged revisions 281762 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010)
+ | 6 lines Allow say.conf to handle large numbers ending with
+ multiple zeros. (closes issue #17833) Reported by: RoadKill
+ Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill
+ (license 933) ........ ................ ................
+
+2010-08-11 17:29 +0000 [r281761] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 281760 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r281760 | mnicholson | 2010-08-11 12:27:59 -0500 (Wed, 11 Aug
+ 2010) | 4 lines Avoid a deadlock in add_header_max_forwards().
+ Related to r276951 ........
+
+2010-08-11 15:20 +0000 [r281726] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, apps/app_readexten.c: Merged revisions 281723 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r281723 | tilghman | 2010-08-11 10:18:40 -0500
+ (Wed, 11 Aug 2010) | 14 lines Merged revisions 281722 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r281722 | tilghman | 2010-08-11 10:17:20 -0500 (Wed, 11 Aug 2010)
+ | 7 lines Only set status TIMEOUT, if we have no digits. (closes
+ issue #15188) Reported by: jcovert Patches:
+ app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license
+ 551) ........ ................
+
+2010-08-11 13:31 +0000 [r281688] <simon.perreault@viagenie.ca>
+
+ * main/netsock2.c, /, include/asterisk/netsock2.h,
+ configs/sip.conf.sample, channels/sip/config_parser.c: Merged
+ revisions 281687 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r281687 | simon.perreault | 2010-08-11 09:30:59 -0400 (Wed, 11
+ Aug 2010) | 9 lines Fix parsing of IPv6 address literals in
+ outboundproxy (closes issue #17757) Reported by: oej Patches:
+ 17757.diff uploaded by sperreault (license 252) sip.conf.diff
+ uploaded by sperreault (license 252) Tested by: oej ........
+
+2010-08-10 21:50 +0000 [r281530-281651] Russell Bryant <russell@digium.com>
+
+ * /, configs/sip.conf.sample, UPGRADE-1.8.txt,
+ channels/sip/include/sip.h: Merged revisions 281650 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r281650 | russell | 2010-08-10 16:47:31 -0500 (Tue, 10
+ Aug 2010) | 5 lines Change the default value for alwaysauthreject
+ in sip.conf to "yes". (closes issue #17756) Reported by: oej
+ ........
+
+ * /, main/sched.c: Merged revisions 281575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r281575 | russell | 2010-08-10 13:05:07 -0500
+ (Tue, 10 Aug 2010) | 16 lines Merged revisions 281574 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010)
+ | 9 lines Don't move the time threshold for running scheduled
+ events on every iteration. Instead, only calculate the time
+ threshold each time ast_sched_runq() is called. (closes issue
+ #17742) Reported by: schmidts Patches: sched.c.patch uploaded by
+ schmidts (license 1077) ........ ................
+
+ * apps/app_dial.c, /: Merged revisions 281568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r281568 | russell | 2010-08-10 12:48:42 -0500
+ (Tue, 10 Aug 2010) | 22 lines Merged revisions 281567 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281567 | russell | 2010-08-10 12:47:13 -0500
+ (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
+ | 8 lines Reset visible indication after answer. (closes issue
+ #17641) Reported by: klaus3000 Patches:
+ ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
+ klaus3000 (license 65) Tested by: schmidts ........
+ ................ ................
+
+ * /, channels/chan_sip.c: Merged revisions 281532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r281532 | russell | 2010-08-10 11:54:20 -0500 (Tue, 10 Aug 2010)
+ | 8 lines Ensure that the proper external address is used for the
+ RTP destination. (closes issue #17044) Reported by: ebroad Tested
+ by: ebroad Review: https://reviewboard.asterisk.org/r/566/
+ ........
+
+ * /, main/cli.c: Merged revisions 281529 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r281529 | russell | 2010-08-10 11:21:58 -0500 (Tue, 10 Aug 2010)
+ | 8 lines Resolve a problem with channel name tab completion.
+ Hitting tab without typing any part of a channel name resulted in
+ no results. This now results in getting a full list of active
+ channels, just as it did in previous versions of Asterisk.
+ Review: https://reviewboard.asterisk.org/r/818/ ........
+
+2010-08-10 07:26 +0000 [r281498] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Fixed the issue caused by EXTEN including
+ user parameters.
+
+2010-08-09 23:04 +0000 [r281467] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 281466 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r281466 | jpeeler | 2010-08-09 18:04:02 -0500 (Mon, 09
+ Aug 2010) | 2 lines Add some more stuff to copy from 281429.
+ ........
+
+2010-08-09 20:49 +0000 [r281433] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 281432 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r281432 | dvossel | 2010-08-09 15:47:53 -0500
+ (Mon, 09 Aug 2010) | 20 lines Merged revisions 281430 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010)
+ | 13 lines fixes SIP peers memory leak We zeroed out the peer's
+ addr before it was removed from the peers_by_ip container. This
+ made it impossible to be removed from the container as the addr
+ is the key used by the container to find the peer. (closes issue
+ #17774) Reported by: kkm Patches:
+ 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
+ 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
+ ........ ................
+
+2010-08-09 20:46 +0000 [r281431] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 281429 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r281429 | jpeeler | 2010-08-09 15:43:54 -0500
+ (Mon, 09 Aug 2010) | 27 lines Merged revisions 281391 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281391 | jpeeler | 2010-08-09 15:07:29 -0500
+ (Mon, 09 Aug 2010) | 20 lines Merged revisions 281390 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010)
+ | 13 lines Prevent loss of Caller ID information set on local
+ channel after masquerade. Caller ID set on the channel before a
+ masquerade occurs when using a local channel would cause the
+ information to be lost. The problem was that the information was
+ set on a channel destined to be hung up. The somewhat confusing
+ fix is to detect if any Caller ID has been set on the channel and
+ if so preswap the Caller ID data so that basically the masquerade
+ puts the data back. (closes issue #17138) Reported by: kobaz
+ Review: https://reviewboard.asterisk.org/r/847/ ........
+ ................ ................
+
+2010-08-09 14:52 +0000 [r281359] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 281358 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r281358 | mnicholson | 2010-08-09 09:49:38 -0500 (Mon, 09 Aug
+ 2010) | 4 lines Validate minrate, maxrate, and modem settings
+ before attempting a fax session. FAX-224 ........
+
+2010-08-09 14:32 +0000 [r281357] <simon.perreault@viagenie.ca>
+
+ * /, configs/sip.conf.sample: Merged revisions 281356 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r281356 | simon.perreault | 2010-08-09 10:31:40 -0400
+ (Mon, 09 Aug 2010) | 2 lines Added comment about IPv4-mapped IPv6
+ addresses and the output of netstat. ........
+
+2010-08-09 12:52 +0000 [r281295-281326] Russell Bryant <russell@digium.com>
+
+ * /, configs/cdr.conf.sample: Merged revisions 281325 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r281325 | russell | 2010-08-09 07:51:43 -0500 (Mon, 09
+ Aug 2010) | 2 lines Add a couple of default values to the
+ documentation of cdr.conf. ........
+
+ * /, configs/cdr.conf.sample: Merged revisions 281294 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r281294 | russell | 2010-08-09 07:14:34 -0500 (Mon, 09
+ Aug 2010) | 5 lines Reorder some options in cdr.conf.sample. Put
+ all of the options that affect the contents of CDRs together,
+ instead of having the batch mode options in the middle of them.
+ ........
+
+2010-08-07 22:36 +0000 [r281226-281257] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fix up handling and indications during
+ transfer. Cleaned up handling of onhook indications and added
+ indications if more than one sub on device. Also fixes issue in
+ 12324 so that the phone can call itself without locking up.
+ (closes issue #17692) Reported by: jmhunter Patches:
+ chan_skinny-transfer-v4.txt uploaded by DEA (license 3)
+ skinnytransfver.v8.diff uploaded by wedhorn (license 30) Tested
+ by: jmhunter, salecha, wedhorn Review: NA
+
+ * channels/chan_skinny.c: Move call answering stuff into new
+ setsubstate_connected. Move call answering stuff into new
+ setsubstate_connected. Also add sub->substate var and set it to
+ SUBSTATE_CONNECTED in setsubstate_connected. (closes issue
+ #17772) Reported by: wedhorn Patches:
+ cleanup.stateconnected2.diff uploaded by wedhorn (license 30)
+ Tested by: wedhorn, salecha Review: NA
+
+ * channels/chan_skinny.c: Start rtp on answer before the answer is
+ queued (closes issue #17770) Reported by: salecha Patches:
+ skinny.answercrash.diff uploaded by wedhorn (license 30) Tested
+ by: salecha Review: NA
+
+2010-08-06 18:58 +0000 [r281086] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/utils.c: Merged revisions 281085 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r281085 | tilghman | 2010-08-06 13:57:10 -0500 (Fri, 06 Aug 2010)
+ | 8 lines Fix alignment of stringfields on the SPARC architecture
+ (closes issue #17789) Reported by: Ian Mason Patches:
+ 20100806__issue17789__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: Ian_Mason ........
+
+2010-08-05 13:19 +0000 [r281054] Russell Bryant <russell@digium.com>
+
+ * main/cdr.c, /: Merged revisions 281052 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r281052 | russell | 2010-08-05 08:16:11 -0500
+ (Thu, 05 Aug 2010) | 16 lines Merged revisions 281051 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010)
+ | 9 lines Cleanup default option value handling for cdr.conf
+ [general]. The default values would differ depending on whether
+ or not cdr.conf exists. That is no longer the case. Apply a
+ default value to the unanswered option. Define all default values
+ as named constants. ........ ................
+
+2010-08-05 07:47 +0000 [r280985] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 280984
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r280984 | tilghman | 2010-08-05 02:46:36 -0500
+ (Thu, 05 Aug 2010) | 22 lines Merged revisions 280983 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r280983 | tilghman | 2010-08-05 02:40:47 -0500
+ (Thu, 05 Aug 2010) | 15 lines Merged revisions 280982 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010)
+ | 8 lines Change context lock back to a mutex, because
+ functionality depends upon the lock being recursive. (closes
+ issue #17643) Reported by: zerohalo Patches:
+ 20100726__issue17643.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo ........ ................ ................
+
+2010-08-04 15:22 +0000 [r280910] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 280909 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r280909 | mnicholson | 2010-08-04 10:11:13 -0500 (Wed, 04 Aug
+ 2010) | 2 lines Initialize FAXOPT() status variables in sendfax
+ and receivefax instead of when the details structure is created.
+ ........
+
+2010-08-04 14:05 +0000 [r280810-280880] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, channels/chan_mgcp.c: Merged revisions 280879 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r280879 | tilghman | 2010-08-04 09:04:07 -0500 (Wed, 04
+ Aug 2010) | 14 lines Check cur value before attempting a deref.
+ (closes issue #17775) Reported by: svinson Patches:
+ 20100804__issue17775.diff.txt uploaded by tilghman (license 14)
+ Tested by: svinson (closes issue #17743) Reported by: tgruenberg
+ Patches: 20100804__issue17775.diff.txt uploaded by tilghman
+ (license 14) Tested by: tgruenberg ........
+
+ * /, funcs/func_strings.c, CHANGES: Merged revisions 280809 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r280809 | tilghman | 2010-08-03 15:25:10 -0500 (Tue, 03 Aug 2010)
+ | 12 lines Sneak FIELDNUM() into 1.8. Returns a 1-based index
+ into a list of a specified item. Matches up with FIELDQTY() and
+ CUT(). (closes issue #17713) Reported by: gareth Patches:
+ svn-279754.diff uploaded by gareth (license 208) Tested by:
+ gareth, tilghman Review: https://reviewboard.asterisk.org/r/810/
+ ........
+
+2010-08-03 19:59 +0000 [r280745-280780] <simon.perreault@viagenie.ca>
+
+ * /, channels/chan_sip.c: Merged revisions 280778 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r280778 | simon.perreault | 2010-08-03 15:54:03 -0400 (Tue, 03
+ Aug 2010) | 9 lines Fixed IPv6-related SIP parsing bugs. (closes
+ issue #17663) Reported by: oej Patches: diff uploaded by
+ sperreault (license 252) diff2 uploaded by sperreault (license
+ 252) get_domain.diff uploaded by sperreault (license 252)
+ ........
+
+ * /, configs/sip.conf.sample: Merged revisions 280777 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r280777 | simon.perreault | 2010-08-03 15:53:07 -0400
+ (Tue, 03 Aug 2010) | 8 lines Better documentation related to
+ IPv6. (closes issue #17737) Reported by: oej Patches: doc.diff
+ uploaded by sperreault (license 252) Tested by: mmichelson
+ ........
+
+ * contrib/realtime/mysql/voicemail.sql, channels/chan_sip.c,
+ contrib/realtime/mysql/iaxfriends.sql,
+ contrib/realtime/mysql/meetme.sql,
+ contrib/realtime/postgresql/realtime.sql,
+ configs/sip.conf.sample, contrib/realtime/mysql/sipfriends.sql:
+ Reverted r280706 and r280707. Will commit in branch 1.8 and merge
+ to trunk properly.
+
+2010-08-03 18:50 +0000 [r280743] Russell Bryant <russell@digium.com>
+
+ * contrib/scripts/get_mp3_source.sh (added), /, addons/Makefile,
+ addons/mp3 (removed): Merged revisions 280742 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r280742 | russell | 2010-08-03 13:48:45 -0500 (Tue, 03 Aug 2010)
+ | 4 lines Remove the MP3 decoder source code and replace it with
+ a small shell script. Review:
+ https://reviewboard.asterisk.org/r/836/ ........
+
+2010-08-03 18:43 +0000 [r280741] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, doc/asterisk.8, doc/Makefile (added), doc/asterisk.sgml:
+ Merged revisions 280740 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r280740 | tilghman | 2010-08-03 13:42:24 -0500
+ (Tue, 03 Aug 2010) | 9 lines Merged revisions 280739 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03
+ Aug 2010) | 2 lines Document -B and -W flags and regenerate
+ manpage from sgml ........ ................
+
+2010-08-03 16:52 +0000 [r280706-280707] <simon.perreault@viagenie.ca>
+
+ * channels/chan_sip.c: Fixed IPv6-related SIP parsing bugs. (closes
+ issue #17663) Reported by: oej Patches: diff uploaded by
+ sperreault (license 252) diff2 uploaded by sperreault (license
+ 252) get_domain.diff uploaded by sperreault (license 252)
+
+ * configs/sip.conf.sample: Better documentation related to IPv6.
+
+2010-08-02 21:28 +0000 [r280629-280673] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, apps/app_voicemail.c: Merged revisions 280672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r280672 | tilghman | 2010-08-02 16:27:25 -0500
+ (Mon, 02 Aug 2010) | 9 lines Merged revisions 280671 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02
+ Aug 2010) | 2 lines Allow the pipe, but also allow the comma
+ ........ ................
+
+ * /, main/Makefile: Merged revisions 280628 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r280628 | tilghman | 2010-08-02 09:41:46 -0500 (Mon, 02 Aug 2010)
+ | 2 lines Make this a little more deterministic... we want the
+ latest value, not just a 1 somewhere. ........
+
+2010-08-02 14:30 +0000 [r280627] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: if totag is not present for an ACK request,
+ do not send an error response
+
+2010-08-02 14:28 +0000 [r280626] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/Makefile: Merged revisions 280624 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r280624 | tilghman | 2010-08-02 09:27:20 -0500 (Mon, 02 Aug 2010)
+ | 2 lines Apparently, the values in makeopts are sometimes 1:1
+ and sometimes 1. Compensate for this. ........
+
+2010-07-30 09:12 +0000 [r280589] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Cleanup transmit_ for handle_register and
+ keepalives Moved inline packet sending to transmit_ subs. Removed
+ handle_keep_alive and handle_register_message to inline in
+ handle_message. Also moved transmit_response(d) to
+ transmit_response_bysessions(s) and created a wrapper
+ transmit_response(d) that calls
+ transmit_response_bysession(d->session). (closes issue #16980)
+ Reported by: wedhorn Patches: skinny-clean06b.diff uploaded by
+ wedhorn (license 30) Tested by: wedhorn, DEA Review: NA
+
+2010-07-29 21:08 +0000 [r280559] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 280557 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r280557 | mnicholson | 2010-07-29 16:07:21 -0500 (Thu, 29 Jul
+ 2010) | 4 lines Fix regression introduced in r1664. Give the fax
+ stack time to shutdown and populate the FAXOPT output variables.
+ FAX-222 ........
+
+2010-07-29 21:06 +0000 [r280555] Paul Belanger <pabelanger@digium.com>
+
+ * CHANGES, channels/chan_iax2.c: PeerStatus now includes Address
+ and Port (closes issue #17730) Reported by: jkroon Patches:
+ iax2-peerstate-address.patch uploaded by jkroon (license 714)
+ Tested by: lmadsen
+
+2010-07-29 20:44 +0000 [r280553] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 280552 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r280552 | dvossel | 2010-07-29 15:43:47 -0500
+ (Thu, 29 Jul 2010) | 17 lines Merged revisions 280551 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010)
+ | 11 lines fixes wrong SRV query for TLS connection (closes issue
+ #17612) Reported by: marcelloceschia Patches:
+ chan-sip_srvQuery.patch uploaded by marcelloceschia (license
+ 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
+ chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia
+ (license 1079) Tested by: marcelloceschia, st, pabelanger
+ ........ ................
+
+2010-07-29 20:36 +0000 [r280550] Russell Bryant <russell@digium.com>
+
+ * /, configs/ccss.conf.sample: Merged revisions 280549 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r280549 | russell | 2010-07-29 15:35:30 -0500 (Thu, 29
+ Jul 2010) | 5 lines Add header to ccss.conf to appease oej.
+ (closes issue #17755) Reported by: oej ........
+
+2010-07-29 19:48 +0000 [r280520] Sean Bright <sean@malleable.com>
+
+ * /, channels/sig_pri.c: Merged revisions 280519 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r280519 | seanbright | 2010-07-29 15:47:16 -0400 (Thu, 29 Jul
+ 2010) | 7 lines Fix compilation error in chan_dahdi (strdupa ->
+ ast_strdupa). (closes issue #17751) Reported by: b11d Patches:
+ strdupa_oops.diff uploaded by malcolmd (license 924) ........
+
+2010-07-29 19:35 +0000 [r280459-280518] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: respond with 481 when request requiring
+ totag has no totag to match against
+
+ * main/channel.c, /: Merged revisions 280450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r280450 | dvossel | 2010-07-29 14:13:27 -0500
+ (Thu, 29 Jul 2010) | 25 lines Merged revisions 280449 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r280449 | dvossel | 2010-07-29 14:05:25 -0500
+ (Thu, 29 Jul 2010) | 18 lines Merged revisions 280448 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010)
+ | 12 lines fixes issue with translator frame not getting freed A
+ translator frame even if it local storage so the translation path
+ can be freed. This issue prevented g729 licenses from being freed
+ up. (closes issue #17630) Reported by: manvirr Patches:
+ encoder_fix.diff uploaded by dvossel (license 671) Tested by:
+ manvirr, dvossel ........ ................ ................
+
+2010-07-29 18:51 +0000 [r280447] Paul Belanger <pabelanger@digium.com>
+
+ * /, tests/test_utils.c: Merged revisions 280446 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r280446 | pabelanger | 2010-07-29 14:37:32 -0400 (Thu, 29 Jul
+ 2010) | 2 lines Remove res_crypto dependency. ........
+
+2010-07-29 16:47 +0000 [r280416] Jean Galarneau <jgalarneau@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 280346 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r280346 | jeang | 2010-07-29 11:07:16 -0500
+ (Thu, 29 Jul 2010) | 17 lines Merged revisions 280345 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r280345 | jeang | 2010-07-29 11:01:35 -0500
+ (Thu, 29 Jul 2010) | 10 lines Merged revisions 280341 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) |
+ 2 lines Fix a dsp structure leak occuring when a local channel is
+ put into a meetme conference, then masquaraded away. ABE-2422
+ ........ ................ ................
+
+2010-07-29 16:45 +0000 [r280415] Paul Belanger <pabelanger@digium.com>
+
+ * /, tests/test_utils.c: Merged revisions 280414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r280414 | pabelanger | 2010-07-29 12:44:22 -0400 (Thu, 29 Jul
+ 2010) | 2 lines crypto_loaded_test depends on res_crypto, else
+ test will fail. ........
+
+2010-07-29 16:26 +0000 [r280395] Russell Bryant <russell@digium.com>
+
+ * main/rtp_engine.c, /: Merged revisions 280391 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r280391 | russell | 2010-07-29 11:25:43 -0500 (Thu, 29 Jul 2010)
+ | 2 lines Don't blow up if get_codec() was not provided in the
+ RTP glue. ........
+
+2010-07-29 15:58 +0000 [r280308-280344] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_usbradio.c, /: Merged revisions 280343 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r280343 | mnicholson | 2010-07-29 10:57:57 -0500 (Thu,
+ 29 Jul 2010) | 4 lines Use PRIx64 instead of PRId64 in format
+ string. related to r280302 ........
+
+ * channels/chan_usbradio.c, /: Merged revisions 280302 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r280302 | pabelanger | 2010-07-28 19:45:34 -0500 (Wed,
+ 28 Jul 2010) | 2 lines Use PRId64 with format_t ........
+
+ * channels/chan_usbradio.c: Make chan_usbradio.c build on 64bit
+ platforms.
+
+ * main/channel.c, channels/chan_local.c, /: Merged revisions 280307
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r280307 | mnicholson | 2010-07-29 08:56:35 -0500
+ (Thu, 29 Jul 2010) | 11 lines Merged revisions 280306 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul
+ 2010) | 2 lines Implement support for ast_channel_queryoption on
+ local channels. Currently only AST_OPTION_T38_STATE is supported.
+ ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........
+ Additionally, pass AST_CONTROL_T38_PARAMETERS control frames
+ through generic bridges. This change appears to have been
+ unintentionally left out of rev 203699. ................
+
+2010-07-28 20:50 +0000 [r280270] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/sip/reqresp_parser.c: Merged revisions 280269 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r280269 | jpeeler | 2010-07-28 15:49:26 -0500 (Wed, 28 Jul 2010)
+ | 2 lines Give test category missing leading slash ........
+
+2010-07-28 20:19 +0000 [r280247] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 280235 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r280235 | rmudgett | 2010-07-28 15:12:16 -0500
+ (Wed, 28 Jul 2010) | 9 lines Merged revisions 280229 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28
+ Jul 2010) | 2 lines Add missing enum value "unknown" to the SS7
+ called_nai and calling_nai config options. ........
+ ................
+
+2010-07-28 20:04 +0000 [r280234] Jason Parker <jparker@digium.com>
+
+ * /, sounds/Makefile: Merged revisions 280233 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r280233 | qwell | 2010-07-28 15:03:22 -0500
+ (Wed, 28 Jul 2010) | 13 lines Merged revisions 280231 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280231 | qwell | 2010-07-28 15:02:27 -0500 (Wed, 28 Jul 2010) |
+ 6 lines Work around some silly behavior on BSD. A non-zero exit
+ from a subshell should make the build fail. (closes issue #17621)
+ ........ ................
+
+2010-07-28 19:37 +0000 [r280226] Terry Wilson <twilson@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Merged revisions 280225 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r280225 | twilson | 2010-07-28 12:34:42 -0700 (Wed, 28
+ Jul 2010) | 3 lines Do rtp/rtcp debugging when it is turned on
+ w/o filtering ........
+
+2010-07-28 18:25 +0000 [r280196] Jason Parker <jparker@digium.com>
+
+ * /, sounds/Makefile: Merged revisions 280195 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r280195 | qwell | 2010-07-28 13:24:29 -0500
+ (Wed, 28 Jul 2010) | 16 lines Merged revisions 280193 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280193 | qwell | 2010-07-28 13:05:54 -0500 (Wed, 28 Jul 2010) |
+ 9 lines Remove unnecessary subshells. Attempt to make
+ checksumming work. Also improves readability. (issue #17621)
+ Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/
+ ........ ................
+
+2010-07-28 16:53 +0000 [r280162] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_queue.c: Merged revisions 280161 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r280161 | seanbright | 2010-07-28 12:52:12 -0400
+ (Wed, 28 Jul 2010) | 15 lines Merged revisions 280160 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul
+ 2010) | 8 lines Plug a reference leak in app_queue when adding
+ members dynamically. (closes issue #17738) Reported by:
+ bobwienholt Patches: issue17738.patch uploaded by bobwienholt
+ (license 950) Tested by: bobwienholt, seanbright ........
+ ................
+
+2010-07-28 14:14 +0000 [r280093] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Formatting changes
+
+2010-07-28 13:53 +0000 [r280091] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/live_ast, /: Merged revisions 280090 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r280090 | lmadsen | 2010-07-28 08:52:50 -0500
+ (Wed, 28 Jul 2010) | 16 lines Merged revisions 280089 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r280089 | lmadsen | 2010-07-28 08:51:16 -0500
+ (Wed, 28 Jul 2010) | 9 lines Merged revisions 280088 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28
+ Jul 2010) | 1 line Update help text to be less confusing.
+ ........ ................ ................
+
+2010-07-28 13:02 +0000 [r280059] Russell Bryant <russell@digium.com>
+
+ * /, res/res_crypto.c: Merged revisions 280058 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r280058 | russell | 2010-07-28 08:01:15 -0500 (Wed, 28 Jul 2010)
+ | 2 lines s/init keys/keys init/ ........
+
+2010-07-28 01:39 +0000 [r280024] Paul Belanger <pabelanger@digium.com>
+
+ * channels/chan_usbradio.c, /: Merged revisions 280023 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r280023 | pabelanger | 2010-07-27 21:37:10 -0400 (Tue,
+ 27 Jul 2010) | 5 lines Resolve compiler warning about formatting
+ (closes issue #17732) Reported by: pabelanger ........
+
+2010-07-27 21:16 +0000 [r279954] Russell Bryant <russell@digium.com>
+
+ * /, utils, codecs, main/db1-ast/mpool, Makefile.rules, cdr,
+ formats, codecs/gsm/src, bridges, codecs/lpc10, configure,
+ main/editline, channels/sip, pbx, res/ael, channels,
+ main/stdtime, main/editline/np, main/db1-ast/hash, cel, apps,
+ configure.ac, main/db1-ast/db, res/ais, res/snmp, funcs,
+ main/db1-ast/btree, codecs/g722, main, main/db1-ast/recno,
+ makeopts.in, res: Merged revisions 279953 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279953 | russell | 2010-07-27 16:16:05 -0500 (Tue, 27 Jul 2010)
+ | 5 lines Add --enable-coverage option to configure script. This
+ option enables the proper compiler flags for tracking code
+ coverage, which is useful along side automated testing. ........
+
+2010-07-27 20:59 +0000 [r279951] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /, include/asterisk/audiohook.h,
+ main/audiohook.c: Merged revisions 279949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r279949 | dvossel | 2010-07-27 15:57:00 -0500
+ (Tue, 27 Jul 2010) | 31 lines Merged revisions 279946 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500
+ (Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010)
+ | 19 lines remove empty audiohook write list on channel If a
+ channel has an audiohook write list created on it, that list
+ stays on the channel until the channel is destroyed. There is no
+ reason to keep that list on the channel if it becomes empty. If
+ it is empty that just means we are doing needless translating for
+ every ast_read and ast_write. This patch removes the audiohook
+ list from the channel once it is detected to be empty on either a
+ read or write. If a audiohook is added back to the channel after
+ this list is destroyed, the list just gets recreated as if it
+ never existed to begin with. (closes issue #17630) Reported by:
+ manvirr Review: https://reviewboard.asterisk.org/r/799/ ........
+ ................ ................
+
+2010-07-27 19:55 +0000 [r279917] Russell Bryant <russell@digium.com>
+
+ * channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions
+ 279916 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279916 | russell | 2010-07-27 14:50:56 -0500 (Tue, 27 Jul 2010)
+ | 12 lines Fix inband DTMF detection on outgoing ISDN calls. This
+ is a regression from the sig_pri split from chan_dahdi. When a
+ call is first initiated, the inband DTMF detector is not enabled
+ if it's an outgoing ISDN call. However, it needs to be turned on
+ once the media path starts up. This handling was put back in the
+ open_media() callback of chan_dahdi. In sig_pri, open_media()
+ calls were added to a few places where it was needed, including
+ handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and
+ PRI_EVENT_PROCEEDING. Thanks to rmudgett for helping me with the
+ patch! ........
+
+2010-07-27 18:55 +0000 [r279888] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 279887 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279887 | mmichelson | 2010-07-27 13:54:07 -0500 (Tue, 27 Jul
+ 2010) | 16 lines Fix parsing error in sip_sipredirect(). The code
+ was written in a way that did a bad job of parsing the port out
+ of a URI. Specifically, it would do badly when dealing with an
+ IPv6 address. In this particular scenario, there was no value
+ from parsing the port out, so I just removed that logic. And
+ while I was messing around in the function, I changed some
+ variable names to be more descriptive. (closes issue #17661)
+ Reported by: oej Patches: 17661.diff uploaded by mmichelson
+ (license 60) ........
+
+2010-07-27 16:41 +0000 [r279851] Jason Parker <jparker@digium.com>
+
+ * /, sounds/Makefile: Merged revisions 279850 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r279850 | qwell | 2010-07-27 11:40:05 -0500
+ (Tue, 27 Jul 2010) | 9 lines Merged revisions 279849 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r279849 | qwell | 2010-07-27 11:39:16 -0500 (Tue, 27 Jul
+ 2010) | 1 line Simply sounds/Makefile some more. ........
+ ................
+
+2010-07-27 16:11 +0000 [r279818] David Vossel <dvossel@digium.com>
+
+ * main/netsock2.c, /, channels/chan_sip.c: Merged revisions 279817
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279817 | dvossel | 2010-07-27 11:09:15 -0500 (Tue, 27 Jul 2010)
+ | 2 lines fix sip transaction match with authentication, fix
+ confusing log message when using getaddrinfo ........
+
+2010-07-27 16:08 +0000 [r279816] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, channels/chan_dahdi.c, /: Merged revisions
+ 279636,279815 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279636 | russell | 2010-07-26 16:53:30 -0500 (Mon, 26 Jul 2010)
+ | 2 lines Ignore a control subclass of -1 in
+ ast_waitfordigit_full(). ........ r279815 | russell | 2010-07-27
+ 11:06:58 -0500 (Tue, 27 Jul 2010) | 4 lines Support "channels" in
+ addition to "channel" in chan_dahdi.conf. Review:
+ https://reviewboard.asterisk.org/r/804 ........
+
+2010-07-27 15:16 +0000 [r279786] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 279785 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r279785 | mmichelson | 2010-07-27 10:15:22 -0500
+ (Tue, 27 Jul 2010) | 20 lines Merged revisions 279784 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul
+ 2010) | 14 lines Fix bad behavior of dynamic_exclude_static
+ option in sip.conf. We were attempting to create a contactdeny
+ rule based on the peer's IP address before the peer's IP address
+ had been set. By moving the processing further down in the
+ function, we can ensure stuff works as we expect for it to.
+ (closes issue #17717) Reported by: mmichelson Patches:
+ 17717.patch uploaded by mmichelson (license 60) Tested by:
+ DennisD ........ ................
+
+2010-07-27 03:02 +0000 [r279727-279756] Paul Belanger <pabelanger@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 279755 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r279755 | pabelanger | 2010-07-26 22:57:33 -0400 (Mon,
+ 26 Jul 2010) | 10 lines If dringXcontext is null, fallback to
+ default context value. (closes issue #17693) Reported by:
+ iasgoscouk Patches: issue17693.patch uploaded by pabelanger
+ (license 224) Tested by: iasgoscouk Review:
+ https://reviewboard.asterisk.org/r/803/ ........
+
+ * /, main/http.c: Merged revisions 279726 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279726 | pabelanger | 2010-07-26 21:53:38 -0400 (Mon, 26 Jul
+ 2010) | 9 lines Use ast_sockaddr_setnull() when http is not
+ enabled. Otherwise, ast_tcptls_server_start() will still start
+ http. (closes issue #17708) Reported by: pabelanger Patches:
+ http.patch uploaded by pabelanger (license 224) ........
+
+2010-07-27 01:39 +0000 [r279725] Russell Bryant <russell@digium.com>
+
+ * CHANGES: Make a formatting change. (Demonstrating the commit IRC
+ bot to pabelanger)
+
+2010-07-26 23:35 +0000 [r279692] Paul Belanger <pabelanger@digium.com>
+
+ * /, CHANGES, UPGRADE-1.8.txt: Merged revisions 279689 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r279689 | pabelanger | 2010-07-26 19:29:34 -0400 (Mon,
+ 26 Jul 2010) | 2 lines Updated documentation for FAX logger
+ level. ........
+
+2010-07-26 23:06 +0000 [r279659] Jason Parker <jparker@digium.com>
+
+ * /, sounds/Makefile.380 (removed), configure,
+ include/asterisk/autoconfig.h.in, sounds/Makefile.381 (removed),
+ configure.ac, sounds/Makefile (added): Merged revisions 279658
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r279658 | qwell | 2010-07-26 18:03:38 -0500
+ (Mon, 26 Jul 2010) | 12 lines Merged revisions 279657 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul 2010) |
+ 5 lines Really fix sounds Makefile (and make it readableish).
+ There was a rather large syntax error that should have caused ALL
+ versions of GNU make to fail. I don't know how it worked.
+ ........ ................
+
+2010-07-26 21:21 +0000 [r279602-279624] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, configure, configure.ac: Merged revisions 279619 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r279619 | tilghman | 2010-07-26 16:20:12 -0500
+ (Mon, 26 Jul 2010) | 9 lines Merged revisions 279609 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r279609 | tilghman | 2010-07-26 16:18:17 -0500 (Mon, 26
+ Jul 2010) | 2 lines Dunno why this worked on my machine, but it
+ works better this way. ........ ................
+
+ * /, res/res_config_ldap.c: Merged revisions 279601 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r279601 | tilghman | 2010-07-26 16:07:45 -0500
+ (Mon, 26 Jul 2010) | 19 lines Merged revisions 279597 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26 Jul 2010) |
+ 13 lines Apply all patches in:
+ https://issues.asterisk.org/view.php?id=13573 (closes issue
+ #13573) Reported by: navkumar Patches:
+ res_config_ldap-category.diff uploaded by navkumar (license 580)
+ res_config_ldap.patch uploaded by bencer (license 961)
+ res_config_ldap uploaded by bencer (license 961) Tested by:
+ suretec ........ ................
+
+2010-07-26 21:07 +0000 [r279600] Gavin Henry <ghenry@suretecsystems.com>
+
+ * /: Merged revisions 279598 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279598 | ghenry | 2010-07-26 21:58:12 +0100 (Mon, 26 Jul 2010) |
+ 21 lines Merged revisions 279597 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/1.6.2
+ -----------------------------------------------------------------------
+ r279597 | ghenry | 2010-07-26 15:25:53 -0500 (Mon, 26 Jul 2010) |
+ 13 lines Apply all patches in:
+ https://issues.asterisk.org/view.php?id=13573 [^] (closes issue
+ 0013573) Reported by: navkumar Patches:
+ res_config_ldap-category.diff uploaded by navkumar (license 580)
+ res_config_ldap.patch uploaded by bencer (license 961)
+ res_config_ldap uploaded by bencer (license 961) Tested by:
+ suretec
+ ------------------------------------------------------------------------
+ ........
+
+2010-07-26 20:00 +0000 [r279569] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/reqresp_parser.c,
+ channels/sip/include/sip.h,
+ channels/sip/include/reqresp_parser.h: Merged revisions 279568
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279568 | dvossel | 2010-07-26 14:59:03 -0500 (Mon, 26 Jul 2010)
+ | 21 lines transaction matching using top most Via header This
+ patch modifies the way chan_sip.c does transaction to dialog
+ matching. Asterisk now stores information in the top most Via
+ header of the initial incoming request and compares that against
+ other Requests that have the same call-id. This results in
+ Asterisk being able to detect a forked call in which it has
+ received multiple legs of the fork. I completely stripped out the
+ previous matching code and made the comparisons a little more
+ explicit and easier to understand. My comments in the code should
+ offer all the details involving this patch. This patch also fixes
+ a bug with the usage of the OBJ-MULTIPLE flag to find multiple
+ dialogs with the same call-id. Since the callback function was
+ returning (CMP_MATCH | CMP_STOP) only the first item found was
+ being returned. I fixed this by making a new callback function
+ for finding multiple dialogs that only returns (CMP_MATCH) on a
+ match allowing for multiple items to be returned. Review:
+ https://reviewboard.asterisk.org/r/776/ ........
+
+2010-07-26 19:58 +0000 [r279567] Paul Belanger <pabelanger@digium.com>
+
+ * /, CHANGES, UPGRADE-1.8.txt, configs/logger.conf.sample: Merged
+ revisions 279566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279566 | pabelanger | 2010-07-26 15:51:39 -0400 (Mon, 26 Jul
+ 2010) | 8 lines Add documentation for FAX logger level. (closes
+ issue #17715) Reported by: vrban Patches: 17715.patch uploaded by
+ pabelanger (license 224) Tested by: vrban ........
+
+2010-07-26 19:20 +0000 [r279564] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, sounds/Makefile.380 (added), configure,
+ include/asterisk/autoconfig.h.in, sounds/Makefile.381 (added),
+ configure.ac, sounds/Makefile (removed): Merged revisions 279562
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r279562 | tilghman | 2010-07-26 14:18:26 -0500
+ (Mon, 26 Jul 2010) | 9 lines Merged revisions 279561 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26
+ Jul 2010) | 2 lines Use a special Makefile for noobs who still
+ have GNU Make 3.80. ........ ................
+
+2010-07-26 16:44 +0000 [r279533] Mark Michelson <mmichelson@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sip/reqresp_parser.c: Merged revisions 279504 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279504 | mmichelson | 2010-07-26 11:04:09 -0500 (Mon, 26 Jul
+ 2010) | 14 lines Allow for systems without locale support to be
+ usable. A recent change to SIP URI comparison code added a
+ locale-specific string comparison to the mix, and certain systems
+ do not support such functions. This fix allows for those systems
+ to still use Asterisk 1.8 (closes issue #17697) Reported by:
+ pprindeville Patches: asterisk-trunk-bugid17697.patch uploaded by
+ pprindeville (license 347) Tested by: mmichelson ........
+
+2010-07-26 15:44 +0000 [r279503] Sean Bright <sean@malleable.com>
+
+ * /, autoconf/ast_ext_lib.m4: Merged revisions 279502 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r279502 | seanbright | 2010-07-26 11:43:54 -0400
+ (Mon, 26 Jul 2010) | 12 lines Merged revisions 279501 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r279501 | seanbright | 2010-07-26 11:41:13 -0400 (Mon, 26 Jul
+ 2010) | 5 lines Expand the correct value within AST_OPTION_ONLY.
+ (closes issue #17703) Reported by: stuarth ........
+ ................
+
+2010-07-26 03:28 +0000 [r279473] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, formats/format_sln.c, formats/format_wav.c,
+ formats/format_ogg_vorbis.c, formats/format_wav_gsm.c,
+ formats/format_sln16.c, formats/format_siren7.c,
+ formats/format_ilbc.c, formats/format_vox.c,
+ formats/format_pcm.c, formats/format_h263.c,
+ formats/format_g723.c, formats/format_h264.c,
+ formats/format_siren14.c, formats/format_jpeg.c,
+ formats/format_g726.c, formats/format_gsm.c,
+ formats/format_g719.c, formats/format_g729.c: Merged revisions
+ 279472 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279472 | tilghman | 2010-07-25 22:27:06 -0500 (Sun, 25 Jul 2010)
+ | 2 lines Formats need to load before apps, because some apps
+ call ast_format_str_reduce() at load time. ........
+
+2010-07-25 21:28 +0000 [r279443] Paul Belanger <pabelanger@digium.com>
+
+ * /, tests/test_func_file.c: Merged revisions 279442 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r279442 | pabelanger | 2010-07-25 17:26:42 -0400 (Sun,
+ 25 Jul 2010) | 2 lines Add trailing backslash to silence warning
+ message. ........
+
+2010-07-25 18:22 +0000 [r279391-279413] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, cdr/cdr_odbc.c: Merged revisions 279410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279410 | tilghman | 2010-07-25 13:21:27 -0500 (Sun, 25 Jul 2010)
+ | 8 lines Don't re-register CDR module on reload. (closes issue
+ #17304) Reported by: jnemeth Patches:
+ 20100507__issue17304.diff.txt uploaded by tilghman (license 14)
+ Tested by: jnemeth ........
+
+ * /, main/logger.c: Merged revisions 279390 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279390 | tilghman | 2010-07-25 12:32:21 -0500 (Sun, 25 Jul 2010)
+ | 8 lines Don't assume qlog is open. (closes issue #17704)
+ Reported by: vrban Patches: issue17704.patch uploaded by
+ pabelanger (license 224) Tested by: vrban ........
+
+2010-07-24 20:49 +0000 [r279274-279315] Paul Belanger <pabelanger@digium.com>
+
+ * Makefile, /: Merged revisions 279314 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279314 | pabelanger | 2010-07-24 16:47:52 -0400 (Sat, 24 Jul
+ 2010) | 7 lines Remove duplicate -c flag when using $(INSTALL)
+ (closes issue #17695) Reported by: pabelanger Patches:
+ Makefile.diff uploaded by pabelanger (license 224) ........
+
+ * /, include/asterisk/netsock2.h: Merged revisions 279280 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279280 | pabelanger | 2010-07-24 14:18:43 -0400 (Sat, 24 Jul
+ 2010) | 8 lines Check if ast_sockaddr is NULL then return.
+ (closes issue #17677) Reported by: outcast Patches:
+ issue0017677.patch uploaded by pabelanger (license 224) Tested
+ by: elguero ........
+
+ * main/manager.c, /: Merged revisions 279273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279273 | pabelanger | 2010-07-24 13:36:42 -0400 (Sat, 24 Jul
+ 2010) | 6 lines Default sin_family to AF_INET for TCP / TLS
+ Bindaddress. Otherwise, 'manager show settings' will generate
+ errors if manager is not enabled. ........
+
+2010-07-23 22:24 +0000 [r279156-279245] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 279227 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r279227 | rmudgett | 2010-07-23 17:20:47 -0500
+ (Fri, 23 Jul 2010) | 21 lines Merged revisions 279207 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r279207 | rmudgett | 2010-07-23 17:11:23 -0500
+ (Fri, 23 Jul 2010) | 14 lines Merged revisions 279206 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010)
+ | 7 lines SIP promiscuous redirect could fail to dial the
+ redirect. The ast_channel was created with one variable to
+ ast_request() but the call to ast_call() that initiates the
+ outgoing call was using a different variable. The two variables
+ are not equivalent if the call_forward string included a channel
+ technology specifier. e.g., SIP/200 ........ ................
+ ................
+
+ * channels/chan_dahdi.c: Make "dahdi show channels" show an
+ outgoing called number. The "dahdi show channels" extension
+ column previously only showed the called number of an incoming
+ call. It now shows the called number for an outgoing call as
+ well. (closes issue #17653) Reported by: amazinzay Patches:
+ issue17653_trunk.txt uploaded by rmudgett (license 664)
+
+2010-07-23 19:17 +0000 [r279116-279118] Russell Bryant <russell@digium.com>
+
+ * UPGRADE.txt, UPGRADE-1.8.txt (added): Shuffle UPGRADE.txt files
+ for 1.10.
+
+ * CHANGES: Start a new section in CHANGES for 1.10.
+
+2010-07-23 18:56 +0000 [r279115] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, res/res_odbc.c: Merged revisions 279113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r279113 | tilghman | 2010-07-23 13:56:04 -0500 (Fri, 23 Jul 2010)
+ | 2 lines Silly 64-bit compilers (who uses 64-bit anyway?)
+ ........
+
+2010-07-23 18:22 +0000 [r279063-279084] Russell Bryant <russell@digium.com>
+
+ * /: Remove old properties.
+
+ * /: Add branch-1.8-merged and branch-1.8-blocked properties to
+ trunk.
+
projects / thirdparty / asterisk.git / commitdiff
