[asterisk-commits] bebuild: tag certified-11.2-cert1-rc1 r382747 - /certified/tags/11.2-cert1-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Mar 8 14:26:27 CST 2013
Author: bebuild
Date: Fri Mar 8 14:26:24 2013
New Revision: 382747
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=382747
Log:
Importing files for 11.2-cert1-rc1 release.
Modified:
certified/tags/11.2-cert1-rc1/.version
certified/tags/11.2-cert1-rc1/ChangeLog
Modified: certified/tags/11.2-cert1-rc1/.version
URL: http://svnview.digium.com/svn/asterisk/certified/tags/11.2-cert1-rc1/.version?view=diff&rev=382747&r1=382746&r2=382747
==============================================================================
--- certified/tags/11.2-cert1-rc1/.version (original)
+++ certified/tags/11.2-cert1-rc1/.version Fri Mar 8 14:26:24 2013
@@ -1,1 +1,1 @@
-11.2.0
+11.2-cert1-rc1
Modified: certified/tags/11.2-cert1-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/certified/tags/11.2-cert1-rc1/ChangeLog?view=diff&rev=382747&r1=382746&r2=382747
==============================================================================
--- certified/tags/11.2-cert1-rc1/ChangeLog (original)
+++ certified/tags/11.2-cert1-rc1/ChangeLog Fri Mar 8 14:26:24 2013
@@ -1,3 +1,460 @@
+2013-03-08 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.2-cert1-rc1 Released.
+
+2013-03-07 17:57 +0000 [r382576-382618] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_voicemail.c, /: Let vm_mailbox_snapshot combine "Urgent"
+ when no folder is specified r381835 fixed a bug in
+ vm_mailbox_snapshot where combining INBOX and Old forgot that
+ Urgent also "counts" as new messages. This fixed the problem when
+ any of the three folders was specified and the combine option was
+ used. It missed the case where the folder isn't specified and we
+ build a snapshot of all folders. This patch corrects that.
+ ........ Merged revisions 382617 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_rtp_asterisk.c, /: Add a 'secret' probation strictrtp
+ mode to handle delayed changes in RTP source Often, Asterisk may
+ realize that a change in the source of an RTP stream is about to
+ occur and ask that the RTP engine reset it's lock on the current
+ RTP source. In certain scenarios, it may take awhile for the new
+ remote system to send RTP packets, while the old remote system
+ may continue providing RTP during that time period. This causes
+ Asterisk to re-lock onto the old source, thereby rejecting the
+ new source when the old source stops sending RTP and the new
+ source begins. This patch prevents that by having a constant
+ secondary, 'secret' probation mode enabled when an RTP source has
+ been chosen. RTP packets from other sources are always
+ considered, but never chosen unless the current RTP source stops
+ sending RTP. Review: https://reviewboard.asterisk.org/r/2364
+ (closes issue AST-1124) Reported by: John Bigelow Tested by: John
+ Bigelow (closes issue AST-1125) Reported by: John Bigelow Tested
+ by: John Bigelow ........ Merged revisions 382573 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-06 19:36 +0000 [r382536] Kinsey Moore <kmoore at digium.com>
+
+ * /, apps/app_page.c: app_page: Fixup application XML documentation
+ typos and inaccuracies. (closes issue AST-1116) ........ Merged
+ revisions 380869 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-04 21:15 +0000 [r382393] Jason Parker <jparker at digium.com>
+
+ * /, main/event.c: Fix comparison of presence state in event
+ subsystem. Several new IEs were not given types (or names),
+ causing the comparison function to improperly succeed. This adds
+ those. (closes issue AST-1128) ........ Merged revisions 382390
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-04 20:21 +0000 [r382387] kharwell <kharwell at localhost>:
+
+ * /, apps/app_confbridge.c: Confbridge CLI new record file name
+ check. This fix checks to make sure that if a confbridge record
+ start command is issued from the CLI it will always use the file
+ name given on the CLI even if it changes between start/stop
+ records for a conference. Previously it had been reusing the same
+ file between start/stops even if a new filename was given. (issue
+ AST-1088) Reported by: John Bigelow ........ Merged revisions
+ 382385 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-28 16:54 +0000 [r382231] Matthew Jordan <mjordan at digium.com>
+
+ * /, apps/app_meetme.c, UPGRADE.txt: Let channels joining a MeetMe
+ conference opt out of the denoiser For some channel drivers,
+ specifically those that have a varying rate in the number of
+ audio samples, the audio quality for a MeetMe conference can be
+ exceedingly poor. This is due to a unilateral application of the
+ DENOISE function in func_speex to channels joining the
+ conference. The denoiser function in the speex library is
+ initialized with the number of audio samples in each sample that
+ will be provided to it. If the number of audio samples changes,
+ the denoiser has to be thrown away and re-initialized. While this
+ could be worked around by removing func_speex, that doesn't help
+ if you actually use the denoiser with other channels on the
+ system. This patches does the following: * Checks for the
+ presence of func_speex as opposed to codec_speex when determining
+ if the DENOISE function is present (which is where the function
+ is actually implemented) * Adds an option to MeetMe 'n' that
+ causes the denoiser to not be applied to a channel when it joins.
+ This keeps the current behavior the default, but let's users
+ disable the denoiser if it causes problems on their system.
+ Review: https://reviewboard.asterisk.org/r/2358 (closes issue
+ AST-1062) Reported by: Thomas Arimont ........ Merged revisions
+ 382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 382230 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-26 16:46 +0000 [r382073-382084] Matthew Jordan <mjordan at digium.com>
+
+ * apps/confbridge/conf_config_parser.c,
+ configs/confbridge.conf.sample, /: Ensure that the default
+ bridge/user profiles are always available ConfBridge and Page
+ require that there always be a default bridge and user profile
+ available. While properties of the default profiles can be
+ overriden in the configuration file, removing them can create
+ situations where neither application can function properly. This
+ patch ensures that if an administrator removes the profiles from
+ the confbridge.conf configuration file, the profiles are added
+ upon load. Documentation clarifying this has been added to the
+ confbridge.conf.sample file. Review:
+ https://reviewboard.asterisk.org/r/2356/ (closes issue AST-1115)
+ Reported by: John Bigelow Tested by: John Bigelow ........ Merged
+ revisions 382066 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/app_confbridge.c: Multiple revisions 379478,382068-382069
+ ........ r379478 | kmoore | 2013-01-18 15:46:58 -0600 (Fri, 18
+ Jan 2013) | 13 lines Fix regression in Confbridge user count When
+ the restructuring work got committed to Confbridge in r375470 to
+ fix many open issues, it caused a regression in the reported
+ count of users when conference information was requested via CLI
+ or manager. This corrects the user count and user information
+ displayed when listing conference information from the CLI and
+ manager. (closes issue ASTERISK-20938) Reported By: Timo Teras
+ Patches: confbridge-list.patch uploaded by Timo Teras (license
+ 5409) ........ r382068 | mjordan | 2013-02-26 09:35:05 -0600
+ (Tue, 26 Feb 2013) | 26 lines Clean up ConfBridge commands to
+ account for wait_marked users When ConfBridge was refactored to
+ better handle the concept of marked, wait_marked, and normal
+ users co-existing in a conference (thereby implementing a state
+ machine for the conference), the wait_marked users were put into
+ their own list of conference participants, separate from the
+ active users. This list is used for wait_marked users when they
+ are waiting in a conference but no marked user has joined; normal
+ users may have joined at this point however. There are several
+ AMI/CLI commands that affect conference users that were not
+ checking the wait_marked users list: * CLI/AMI commands that
+ mute/unmute a participant. In this case, wait_marked users have
+ to remain in their particular state and should not be affected -
+ however, the commands would return "Channel not found" as opposed
+ to the appropriate error condition. * CLI/AMI commands that kick
+ a participant. An admin should always be able to kick a
+ participant out of the conference. This patch fixes both sets of
+ commands, and cleans up the CLI commands slightly by allowing
+ them to complete a participant name (this was supposed to have
+ been added, but the function call was commented out and wasn't
+ implemented). Review: https://reviewboard.asterisk.org/r/2346/
+ (closes issue AST-1114) Reported by: John Bigelow Tested by: John
+ Bigelow ........ r382069 | mjordan | 2013-02-26 09:38:05 -0600
+ (Tue, 26 Feb 2013) | 3 lines Fix typo in r382068 Well, that was
+ embarrassing. Removed an '-l' that somehow got in there. ........
+ Merged revisions 379478,382068-382069 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-20 19:15 +0000 [r381832-381836] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_voicemail.c, /: Let vm_mailbox_snapshot_create's combine
+ option apply to "Urgent" as well The vm_mailbox_snapshot_create
+ function has an option that combines the contents of INBOX and
+ Old into a single snapshot. The intent of this is that both 'new'
+ messages and 'deleted' messages are given in a single snapshot,
+ as some applications prefer this view of the voicemail world.
+ Unfortunately, the initial implementation ignored the "Urgent"
+ folder. The "Urgent" folder is a pseudo-INBOX, in that new
+ messages left with the 'U' flag will be placed in that folder as
+ opposed to INBOX. Thus, the option failed the intent with which
+ it was added. This patch makes it so that the "Urgent" folder is
+ included in the snapshot when that option is used. ........
+ Merged revisions 381835 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: Ensure Min-SE is included in outbound
+ INVITEs Asterisk now includes Min-SE in outbound INVITEs when the
+ value is not 90 (the default) and session timers are not
+ disabled. This has the effect of Asterisk following RFC4028 more
+ closely with regard to 422 responses and preventing situations in
+ which Asterisk would be forced to temporarily accept a call to
+ tear it down based on a Session-Expires below the locally
+ configured Min-SE. (issue SWP-5051) Review:
+ https://reviewboard.asterisk.org/r/2222/ Reported-by: Kinsey
+ Moore Patch-by: Kinsey Moore ........ Merged revisions 377946
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 377947 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377948 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * asterisk-11.2.0-summary.html (removed),
+ asterisk-11.2.0-summary.txt (removed): Remove the release
+ summaries from the branch
+
+2013-02-20 16:16 +0000 [r381705-381823] kharwell <kharwell at localhost>:
+
+ * /: Updated merge properties to reflect correct trace.
+
+ * apps/app_confbridge.c: Confbridge channels staying active when
+ all participants leave. If you started/stopped recording of a
+ conference multiple times channels would remain active even when
+ all participants left the conference. This was due to the fact
+ that a reference to the confbridge was being added every time a
+ start record command was issued, but when the recording was
+ stopped there was no matching de-reference thus keeping the
+ conference alive. Made sure only a single reference is added for
+ the record thread no matter how many times recording is
+ started/stopped. A de-reference is issued upon thread ending.
+ Note, this issue is being fixed under AST-1088 since it relates
+ to it and should have been corrected along with those
+ modifications. (issue AST-1088) Reported by: John Bigelow
+ ........ Merged revisions 381737 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/app_confbridge.c: Fixed Confbridge file recording
+ deadlock and appending. A deadlock occurred after
+ starting/stopping and then restarting a confbridge recording.
+ Upon starting a recording a record thread is created that holds a
+ lock until just before exiting. Stopping the recording does not
+ stop/exit the thread or release the lock. The thread waits until
+ recording begins again. Starting a stopped recording signals the
+ thread to continue and start recording again. However restarting
+ the recording also created another record thread resulting in a
+ deadlock. The fix was to make sure the record thread was only
+ created once. Also it was noted that filenames for the recordings
+ were being concatenated for each start/stop. This was fixed by
+ creating a new file for each conference session and appending the
+ actual recorded data within the file (e.g. passing the 'a' option
+ to MixMonitor). (issue AST-1088) Reported by: John Bigelow
+ Review: http://reviewboard.digium.internal/r/374/ ........ Merged
+ revisions 381702 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-16 16:31 +0000 [r381596-381616] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: Don't send presencestate information if
+ the state is invalid Previously, presencestate information was
+ sent whenever the state was not NOT_SET. When r381594 actually
+ returned INVALID presence state in all the places it was supposed
+ to, it caused chan_sip to start adding presence state information
+ to NOTIFY requests that it previously would not have added.
+ chan_sip shouldn't be adding presence state information when the
+ provider is in an invalid state; users can't set the state to
+ invalid and an invalid state always implies that the provider is
+ in an error condition. (issue AST-1084) ........ Merged revisions
+ 381613 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * funcs/func_presencestate.c, main/manager.c, /,
+ main/presencestate.c: Fix crash in PresenceState AMI action when
+ specifying an invalid provider This patch fixes a crash in
+ Asterisk that could be caused by using the PresenceState AMI
+ action while providing an invalid provider. This patch also adds
+ some additional warnings when a user attempts to provide the
+ PresenceState action with invalid data, and removes some NOTICE
+ statements that were still lurking in the code from testing.
+ (closes issue AST-1084) Reported by: John Bigelow Tested by: John
+ Bigelow ........ Merged revisions 381594 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-14 18:46 +0000 [r381400-381447] Matthew Jordan <mjordan at digium.com>
+
+ * main/channel.c, /: Multiple revisions 378121,378459 ........
+ r378121 | kmoore | 2012-12-18 11:41:35 -0600 (Tue, 18 Dec 2012) |
+ 14 lines Add test events for time limit-related hangups This
+ patch adds hangup-related test events in order to support testing
+ of time-limited bridges. This aids in testing the S() and L()
+ bridge options. (issue SWP-4713) ........ Merged revisions 378119
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 378120 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ r378459
+ | kmoore | 2013-01-03 12:48:00 -0600 (Thu, 03 Jan 2013) | 10
+ lines Add missing test event This test event was missing from
+ channel.c causing the dial_LS_options test to fail intermittently
+ because of a race condition where most code paths emitted the
+ test event but this one did not. The dial_LS_options test should
+ stop bouncing now. ........ Merged revisions 378455 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378121,378459 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: Fixed failing test from r380696. When I
+ added my extensive suite of session timer unit tests, apparently
+ one of them was failing and I never noticed. If neither Min-SE
+ nor Session-Expires is set in the header, it was responding with
+ a Session-Expires of the global maxmimum instead of the
+ configured max for the endpoint. (issue ASTERISK-20787) ........
+ Merged revisions 380973 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 380974 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: Process session timers, even if
+ Session-Expires header is missing Previously, Asterisk only
+ processed session timer information if both the 'Supported:
+ timer' and 'Session-Expires' headers were present. However, the
+ Session-Expires header is optional. If we were to receive a
+ request with a Min-SE greater than our configured
+ session-expires, we would respond with a 'Session-Expires' header
+ that was too small. This patch cleans the situation up a bit,
+ always processing timer information if the 'Supported: timer'
+ header is present. (closes issue ASTERISK-20787) Reported by:
+ Mark Michelson Review: https://reviewboard.asterisk.org/r/2299/
+ ........ Merged revisions 380696 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 380698 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/confbridge/conf_state_multi_marked.c: app_confbridge: Fix
+ error messages on exiting conference. A marked user ending a
+ conference with only end_marked users generates error messages:
+ ERROR[0000][C-00000000]: confbridge/conf_state.c:47
+ conf_invalid_event_fn: Invalid event for confbridge user '' * The
+ MULTI_MARKED state was doing too much when it was kicking out the
+ end_marked users from the conference. The kicked out users will
+ clean up after themselves when they exit the conference. (closes
+ issue ASTERISK-20991) Reported by: Jeremy Kister Tested by:
+ rmudgett ........ Merged revisions 380892 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/app_page.c, apps/app_confbridge.c: app_page and
+ app_confbridge: Fix custom announcement on entering conference.
+ The Page and ConfBridge custom announcement did not play when
+ users entered the conference. * Fix the
+ CONFBRIDGE(user,announcement) file not getting played. The code
+ to do this got removed accidentally when the ConfBridge code was
+ restructured to be more state machine like. * Fixed
+ play_prompt_to_user() doxygen comments. * Fixed the Page A(x) and
+ n options for the caller. The caller never played the
+ announcement file and totally ignored the n option. The code to
+ do this was lost when the application was converted to use
+ ConfBridge. * Factored out setup_profile_bridge(),
+ setup_profile_paged(), and setup_profile_caller() routines to
+ setup ConfBridge profiles. Made each profile setup routine use
+ the default template if one has not already been setup by
+ dialplan. (closes issue ASTERISK-20990) Reported by: Jeremy
+ Kister Tested by: rmudgett ........ Merged revisions 380894 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/asterisk.c: Fix astcanary startup problem due to wrong
+ pid value from before daemon call When Asterisk forks itself into
+ the background via a call to daemon, it must re-set the pid value
+ of the new process. Otherwise, astcanary gets the pid value of
+ the process before the fork, which prevents it from running.
+ Asterisk eventually starts lowering its priority, as it can no
+ longer communicate with the proverbial canary in the coal mine.
+ This patch ensures that the correct process identifier is used by
+ astcanary. Note that this is getting committed to 10 as a
+ regression fix. (closes issue ASTERISK-20947) Reported by: Jakob
+ Hirsch Tested by: mjordan patches:
+ asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch
+ (license 6113) ........ Merged revisions 379509 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379510 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 379513 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * contrib/init.d/rc.mandriva.asterisk,
+ contrib/init.d/rc.debian.asterisk, /,
+ contrib/init.d/rc.redhat.asterisk, UPGRADE.txt,
+ contrib/init.d/rc.gentoo.asterisk,
+ contrib/init.d/rc.slackware.asterisk,
+ contrib/init.d/rc.archlinux.asterisk,
+ contrib/scripts/safe_asterisk, main/asterisk.c,
+ contrib/init.d/rc.suse.asterisk: Update init.d scripts to handle
+ stderr; readd splash screen for remote consoles When r376428 was
+ commited to re-order start up sequences to be more tolerant of
+ forking with thread primitives, a few items were changed that
+ caused changes in behavior on some distros. This includes: * Not
+ displaying the splash screen on a remote console. * Displaying an
+ error message on stderr when a remote console cannot connect to a
+ running instance of Asterisk. In the first case, the splash
+ screen was re-added (thanks to Michael L. Young). In the second
+ case, the various init.d scripts were modified to pipe stderr to
+ /dev/null, as the error message is useful - if you execute a
+ remote console or a remote console command execution and it fail,
+ it should tell you. Note that the error message was always
+ present, it just failed to be printed prior to r376428. Much
+ thanks to the folks who quickly reported this problem, provided
+ solutions, and promptly tested the various init.d scripts on a
+ variety of distros. (closes issue ASTERISK-20945) Reported by:
+ Warren Selby Tested by: Michael L. Young, Jamuel Starkey,
+ kaldemar, Danny Nicholas, mjordan patches:
+ asterisk-20945-remote-intro-msg.diff uploaded by elguero (license
+ 5026) ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan
+ (license 6283) ........ Merged revisions 379760 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379777 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 379790 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_rtp_asterisk.c, /: Reset RTP timestamp; sequence number
+ on SSRC change In r370252 for ASTERISK-18404, Asterisk's handling
+ of RTP was modified to better account for out of order RTP
+ packets. This was accomplished by using the RTP timestamp and
+ sequence number to check for out of order packets. However, when
+ a SSRC change occurs, the timestamp and sequence number will no
+ longer have any relation to the previously received packets. The
+ variables tracking the timestamp and sequence number therefore
+ have to be reset. (closes issue ASTERISK-20906) Reported by:
+ Eelco Brolman patches: dtmf_on_hold.patch uploaded by Eelco
+ Brolman (license #6442) ........ Merged revisions 378967 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378984 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: Perform case insensitive comparisons for
+ T.38 attributes RFC5347 section 2.5.2 states the following: ...
+ The attribute "T38MaxBitRate" was once incorrectly registered
+ with IANA as "T38maxBitRate" (lower-case "m"). In accordance with
+ T.38 examples and common implementation practice, the form
+ "T38MaxBitRate" SHOULD be generated by implementations conforming
+ to this package. In general, it is RECOMMENDED that
+ implementations of this package accept lowercase, uppercase, and
+ mixed upper/lowercase encodings of all the T.38 attributes. ...
+ Asterisk currently does not perform case insensitive matching on
+ the T.38 attributes. This causes the T38MaxBitRate attribute to
+ be negotiated at 2400 baud instead of 14400 (or whatever value
+ you actually wanted). This patch makes it so that when we compare
+ T.38 attributes, we do so in a case insensitive fashion. Note
+ that while the issue reporter did not directly write the patch,
+ they contributed to it (and would have provided one themselves if
+ the license had gone through a tad faster), and hence get
+ attribution for it. Review:
+ https://reviewboard.asterisk.org/r/2298/ (closes issue
+ ASTERISK-20897) Reported by: Eric Hill Tested by: Eric Hill
+ patches: -- uploaded by Eric Hill ........ Merged revisions
+ 380458 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 380465 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/rtp_engine.c, /: Do not allow native RTP bridging if
+ packetization of media streams differs. The RTP engine will no
+ longer allow for local and remote native RTP bridges if
+ packetization of streams differs. Allowing native bridging in
+ this scenario has been known to cause FAX failures. (closes
+ ASTERISK-20650) Reported by: Maciej Krajewski Patches:
+ ASTERISK-20659.patch uploaded by Mark Michelson (License #5049)
+ Review: https://reviewboard.asterisk.org/r/2319 ........ Merged
+ revisions 381281 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 381306 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /: Track merged changes using the standard branch nomenclature
+
+2013-02-08 19:42 +0000 [r381085] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_meetme.c, sounds/Makefile: Merge r379892 into Certified
+ 11.2 ........ r379892 | jrose | 2013-01-22 13:07:42 -0600 (Tue,
+ 22 Jan 2013) | 16 lines app_meetme: Use new prompts for
+ administrator menu The old prompts for the administrator menu
+ were inadequate. They didn't mention that the menu had additional
+ options through the 8 key and pressing the 8 key wouldn't reveal
+ what those options were. This patch fixes all of that while also
+ organizing code pertaining to each individual menu type which was
+ previously all stored in one gigantic function along with many of
+ the basic conference functions. (closes issue AST-996) Reported
+ by: John Bigelow Review:
+ http://reviewboard.digium.internal/r/360/ ........ Merged
+ revisions 379885 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ------------------------------------------------------------------------
+
+2013-01-14 20:23 +0000 [r379063] Matthew Jordan <mjordan at digium.com>
+
+ * / (added): Create branch for Certified Asterisk 11.2.
+
2013-01-14 Asterisk Development Team <asteriskteam at digium.com>
* Asterisk 11.2.0 Released.
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