[svn-commits] bebuild: tag 10.0.0-rc1 r344001 - /tags/10.0.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Nov 8 15:51:13 CST 2011
Author: bebuild
Date: Tue Nov 8 15:51:09 2011
New Revision: 344001
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=344001
Log:
Importing files for 10.0.0-rc1 release.
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tags/10.0.0-rc1/.version (with props)
tags/10.0.0-rc1/ChangeLog (with props)
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+2011-11-08 Asterisk Development Team <asteriskteam at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * res/res_calendar.c: Don't crash on empty notify channel
+
+2011-10-29 04:26 +0000 [r342662] Richard Mudgett <rmudgett at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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
[... 18268 lines stripped ...]
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