[svn-commits] bebuild: tag certified-13.1-cert1-rc2 r430393 - in /certified/tags/13.1-cert1...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Jan 8 12:30:52 CST 2015
Author: bebuild
Date: Thu Jan 8 12:30:50 2015
New Revision: 430393
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=430393
Log:
Importing files for 13.1-cert1-rc2 release.
Modified:
certified/tags/13.1-cert1-rc2/.version
certified/tags/13.1-cert1-rc2/ChangeLog
certified/tags/13.1-cert1-rc2/contrib/realtime/mysql/mysql_config.sql
certified/tags/13.1-cert1-rc2/contrib/realtime/oracle/oracle_config.sql
certified/tags/13.1-cert1-rc2/contrib/realtime/postgresql/postgresql_config.sql
certified/tags/13.1-cert1-rc2/contrib/realtime/sqlserver/mssql_config.sql
Modified: certified/tags/13.1-cert1-rc2/.version
URL: http://svnview.digium.com/svn/asterisk/certified/tags/13.1-cert1-rc2/.version?view=diff&rev=430393&r1=430392&r2=430393
==============================================================================
--- certified/tags/13.1-cert1-rc2/.version (original)
+++ certified/tags/13.1-cert1-rc2/.version Thu Jan 8 12:30:50 2015
@@ -1,1 +1,1 @@
-13.1.0
+13.1-cert1-rc2
Modified: certified/tags/13.1-cert1-rc2/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/certified/tags/13.1-cert1-rc2/ChangeLog?view=diff&rev=430393&r1=430392&r2=430393
==============================================================================
--- certified/tags/13.1-cert1-rc2/ChangeLog (original)
+++ certified/tags/13.1-cert1-rc2/ChangeLog Thu Jan 8 12:30:50 2015
@@ -1,3 +1,431 @@
+2015-01-08 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 13.1-cert1-rc2 Released.
+
+2015-01-07 03:29 +0000 [r430253-430293] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_mp3.c, channels/chan_alsa.c, res/res_timing_kqueue.c,
+ channels/chan_unistim.c, res/res_config_pgsql.c,
+ res/res_phoneprov.c, utils/smsq.c, apps/app_morsecode.c,
+ cdr/cdr_pgsql.c, res/res_config_sqlite.c,
+ res/res_pjsip_phoneprov_provider.c, pbx/pbx_ael.c,
+ res/res_statsd.c, apps/app_sms.c, formats/format_jpeg.c,
+ utils/streamplayer.c, utils/check_expr.c, apps/app_jack.c,
+ apps/app_adsiprog.c, cel/cel_radius.c, channels/chan_sip.c,
+ cel/cel_tds.c, apps/app_festival.c, agi/eagi-test.c,
+ res/res_hep_pjsip.c, channels/chan_console.c, cdr/cdr_radius.c,
+ apps/app_getcpeid.c, apps/app_talkdetect.c, channels/chan_oss.c,
+ utils/stereorize.c, apps/app_osplookup.c, channels/chan_misdn.c,
+ channels/chan_skinny.c, funcs/func_frame_trace.c, apps/app_amd.c,
+ pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_url.c,
+ channels/chan_nbs.c, utils/extconf.c, apps/app_externalivr.c,
+ apps/app_zapateller.c, cdr/cdr_odbc.c, channels/chan_mgcp.c,
+ cel/cel_pgsql.c, utils/muted.c, apps/app_test.c, utils/astman.c,
+ apps/app_ices.c, utils/conf2ael.c, cdr/cdr_csv.c,
+ channels/chan_phone.c, funcs/func_pitchshift.c,
+ apps/app_waitforring.c, funcs/func_audiohookinherit.c,
+ formats/format_vox.c, res/res_timing_pthread.c,
+ apps/app_minivm.c, cel/cel_sqlite3_custom.c, res/res_hep_rtcp.c,
+ res/res_config_ldap.c, apps/app_nbscat.c,
+ cdr/cdr_sqlite3_custom.c, res/res_hep.c, res/res_snmp.c,
+ apps/app_dictate.c, apps/app_waitforsilence.c,
+ apps/app_dahdiras.c, pbx/pbx_lua.c, apps/app_alarmreceiver.c,
+ res/res_ael_share.c, apps/app_image.c, cdr/cdr_tds.c: Disable
+ extended support modules
+
+ * /,
+ contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py:
+ contrib/ast-db-manage: Correct down_revision path for
+ user_eq_phone When the user_eq_phone patch was backported to 13,
+ it referenced the downward revision that the PJSIP optimistic
+ encryption option also references. This creates a multi-path
+ upgrade Exception when generating the SQL files. This patch
+ corrects this in the 13 branch. Note that trunk, which already
+ contained both of these features, is unaffected by this problem.
+ ........ Merged revisions 430252 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-06 19:53 +0000 [r430245] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * /, main/bridge_basic.c: bridge: avoid leaking channel during
+ blond transfer pt2 A blond transfer to a failed destination, when
+ followed by a recall attempt, lead to a leak of the reference to
+ the destination channel. In addition to correcting the regression
+ on the previous attempt (r429826) this fixes the leak and two
+ additional reference leaks on failures of bridge_import.
+ ASTERISK-24513 #close Review:
+ https://reviewboard.asterisk.org/r/4302/ ........ Merged
+ revisions 430199 from
+ http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+ revisions 430200 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2014-12-24 15:27 +0000 [r430085-430094] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_agi.c, /: res/res_agi: Make Verbose message for 'stream
+ file' match other playbacks The Verbose message displayed when a
+ file is played back via 'stream file' was formatted differently
+ than other playbacks: * It didn't include the channel name * It
+ didn't include the channel language It does, however, include the
+ playback offset as well as any escape digits. That information
+ was kept; however, this patch updates the formatting to more
+ closely match the Verbose messages displayed when a file is
+ played back by 'control stream file', Playback, ControlPlayback,
+ or any other file playback operation. ........ Merged revisions
+ 429519 from http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * res/res_pjsip.c,
+ contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py
+ (added), /: res_pjsip: Backport missing commits for user_eq_phone
+ This backports the following from trunk, which were missed:
+ r427257 | file | 2014-11-04 16:31:16 -0600 (Tue, 04 Nov 2014) | 2
+ lines res_pjsip: Allow + at the beginning of a phone number when
+ user_eq_phone is enabled. r427259 | file | 2014-11-04 16:51:32
+ -0600 (Tue, 04 Nov 2014) | 2 lines res_pjsip: Apply the
+ 'user_eq_phone' setting to the To header as well. It also adds
+ the Alembic script for the option. ASTERISK-24643 ........ Merged
+ revisions 430092 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, tests/test_stasis_channels.c: Stasis: Update unittest for
+ channel snapshots This adjusts the unit test for channel
+ snapshots to take the new language key into account. ........
+ Merged revisions 429352 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, configs/samples/pjsip.conf.sample, CHANGES, res/res_pjsip.c,
+ include/asterisk/res_pjsip.h, res/res_pjsip_keepalive.c (added),
+ res/res_pjsip/config_global.c: res_pjsip_keepalive: Add runtime
+ configurable keepalive module for connection-oriented transports.
+ Note that this is backport from trunk of r425825. This change
+ adds a module which is configurable using the keep_alive_interval
+ setting in the global section that will send a CRLF keep alive to
+ all active connection-oriented transports at the provided
+ interval. This is useful because it can help keep connections
+ open through NATs. This functionality also exists within PJSIP
+ but can not be controlled at runtime and requires recompiling it.
+ Review: https://reviewboard.asterisk.org/r/4084/ ASTERISK-24644
+ #close ........ Merged revisions 430084 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, res/res_pjsip/pjsip_configuration.c,
+ res/res_pjsip_caller_id.c, CHANGES, res/res_pjsip.c,
+ include/asterisk/res_pjsip.h: res_pjsip: Add 'user_eq_phone'
+ option to add a 'user=phone' parameter when applicable. Note that
+ this is a backport of r425804 from trunk. This change adds a
+ configuration option which adds a 'user=phone' parameter if the
+ user portion of the request URI or the From URI is determined to
+ be a number. Review: https://reviewboard.asterisk.org/r/4073/
+ ASTERISK-24643 #close ........ Merged revisions 430083 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2014-12-22 21:22 +0000 [r430030-430046] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/bridge_basic.c: DTMF atxfer: Setup recall channels as if
+ the transferee initiated the call. After the initial DTMF atxfer
+ call attempt to the transfer target fails to answer during a
+ blonde transfer, the recall callback channels do not get setup
+ with information from the initial transferrer channel. As a
+ result, the recall callback to the transferrer does not have
+ callid, channel variables, datastores, accountcode, peeraccount,
+ COLP, and CLID setup. A similar situation happens with the recall
+ callback to the transfer target but it is less visible. The
+ recall callback to the transfer target does not have callid,
+ channel variables, datastores, accountcode, peeraccount, and COLP
+ setup. * Added missing information to the recall callback
+ channels before initiating the call. callid, channel variables,
+ datastores, accountcode, peeraccount, COLP, and CLID * Set callid
+ of the transferrer channel on the DTMF atxfer controller thread
+ attended_transfer_monitor_thread(). * Added missing channel
+ unlocks and props unref to off nominal paths in
+ attended_transfer_properties_alloc(). ASTERISK-23841 #close
+ Reported by: Richard Mudgett Review:
+ https://reviewboard.asterisk.org/r/4259/ ........ Merged
+ revisions 430034 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * main/logger.c, include/asterisk/_private.h, main/asterisk.c, /:
+ queue_log: Post QUEUESTART entry when Asterisk fully boots. The
+ QUEUESTART log entry has historically acted like a fully booted
+ event for the queue_log file. When the QUEUESTART entry was
+ posted to the log was broken by the change made by
+ ASTERISK-15863. * Made post the QUEUESTART queue_log entry when
+ Asterisk fully boots. This restores the intent of that log entry
+ and happens after realtime has had a chance to load. AST-1444
+ #close Reported by: Denis Martinez Review:
+ https://reviewboard.asterisk.org/r/4282/ ........ Merged
+ revisions 430009 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 430010 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2014-12-22 18:35 +0000 [r430007-430008] bebuild <bebuild at localhost>:
+
+ * /, res/res_pjsip/pjsip_options.c: Multiple revisions
+ 429128,429246 ........ r429128 | kmoore | 2014-12-09 08:00:50
+ -0600 (Tue, 09 Dec 2014) | 12 lines PJSIP: Stagger outbound
+ qualifies This change staggers initiation of outbound qualify
+ (OPTIONS) attempts to reduce instantaneous server load and
+ prevent network congestion. Review:
+ https://reviewboard.asterisk.org/r/4246/ ASTERISK-24342 #close
+ Reported by: Richard Mudgett ........ Merged revisions 429127
+ from http://svn.asterisk.org/svn/asterisk/branches/12 ........
+ r429246 | kmoore | 2014-12-10 07:14:56 -0600 (Wed, 10 Dec 2014) |
+ 8 lines PJSIP: Fix assert on initial mass qualify This fixes the
+ MWI test regressions caused by r429127 and ensures that contacts
+ have non-zero qualify_frequency before attempting scheduling.
+ ........ Merged revisions 429245 from
+ http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+ revisions 429128,429246 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * main/manager.c, /: Prevent possible race condition on dual
+ redirect of channels in the same bridge. The
+ AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT flag was created to prevent
+ bridges from prematurely acting on orphaned channels in bridges.
+ The problem with the AMI redirect action was that it was setting
+ this flag on channels based on the presence of a PBX, not whether
+ the channel was in a bridge. Whether a channel has a PBX is
+ irrelevant, so the condition has been altered to check if the
+ channel is in a bridge. ASTERISK-24536 #close Reported by Niklas
+ Larsson Review: https://reviewboard.asterisk.org/r/4268 ........
+ Merged revisions 429741 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2014-12-19 21:52 +0000 [r429855-429892] bebuild <bebuild at localhost>:
+
+ * /, rest-api/api-docs/channels.json, res/ari/resource_channels.c,
+ CHANGES, res/res_ari_channels.c, res/ari/resource_channels.h:
+ ari: Add support for specifying an originator channel when
+ originating. If an originator channel is specified when
+ originating a channel the linked ID of it will be applied to the
+ newly originated outgoing channel. This allows an association to
+ be made between the two so it is known that the originator has
+ dialed the originated channel. ASTERISK-24552 #close Reported by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/4243/
+ ........ Merged revisions 429153 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, main/stasis_channels.c, rest-api/api-docs/channels.json,
+ res/ari/ari_model_validators.c, main/manager_channels.c,
+ res/ari/ari_model_validators.h: ARI/AMI: Include language in
+ standard channel snapshot output The channel "language" was
+ already part of a channel snapshot, however is was not sent out
+ over AMI or ARI. This patch makes it so the channel "language" is
+ included in the appropriate AMI or ARI events. ASTERISK-24553
+ #close Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/4245/ ........ Merged
+ revisions 429204 from
+ http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+ revisions 429206 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, res/res_pjsip_session.c: res_pjsip_session: Fix issue where a
+ declined media stream in a re-INVITE would fail SDP negotiation.
+ In the past the SDP negotiation within res_pjsip_session was made
+ more tolerant of certain situations. The only case where SDP
+ negotiation will fail is when a major error occurs during
+ negotiation. Receiving an already declined media stream is not
+ considered a major error. When producing the local SDP the logic
+ took this into account so on the initial INVITE the declined
+ media stream did not cause an SDP negotiation failure.
+ Unfortunately the logic for handling media streams with a handler
+ did not mirror this logic and considered an already declined
+ media stream an error and thus failed the SDP negotiation. This
+ change makes the logic between both situations match so only
+ under major errors will the SDP negotiation fail. ASTERISK-24607
+ #close Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/4254/ ........ Merged
+ revisions 429407 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * main/format.c, /, main/codec.c, include/asterisk/format.h: media:
+ Fix crash when determining sample count of a frame during
+ shutdown. When shutting down Asterisk the codecs are cleaned up.
+ As a result anything attempting to get a codec based on ID or
+ details will find that no codec exists. This currently occurs
+ when determining the sample count of a frame. This code did not
+ take this situation into account. This change fixes this by
+ getting the codec directly from the format and eliminates the
+ lookup. This is both faster and also provides a guarantee that
+ the codec will exist and will be valid. ASTERISK-24604 #close
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/4260/ ........ Merged
+ revisions 429497 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, res/res_pjsip_outbound_registration.c: Prevent potential
+ infinite outbound authentication loops in registration. Prior to
+ this patch, Asterisk would always respond to 401 responses to
+ registration attempts by trying to provide a registration with
+ authentication credentials. Even if subsequent attempts were
+ rejected with 401 responses, Asterisk would continue this
+ behavior. If authentication credentials were incorrect, this
+ could continue forever. With this patch, we keep track of whether
+ we have attempted authentication on an outbound registration
+ attempt. If we already have, we don not try again until the next
+ attempt. This prevents the infinite loop scenario. Review:
+ https://reviewboard.asterisk.org/r/4273 ........ Merged revisions
+ 429761 from http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * res/res_pjsip_outbound_publish.c, /: res_pjsip_outbound_publish:
+ stack overflow when using non-default sorcery wizard When using a
+ non-default sorcery wizard (in this instance realtime) for
+ outbound publishes Asterisk will crash after a stack overflow
+ occurs due to the code infinitely recursing. The fix entails
+ removing the outbound publish state dependency from the outbound
+ publish sorcery object and instead keeping an in memory container
+ that can be used to lookup the state when needed. ASTERISK-24514
+ #close Reported by: Mark Michelson Review:
+ https://reviewboard.asterisk.org/r/4178/ ........ Merged
+ revisions 429175 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, res/res_pjsip_sdp_rtp.c: PJSIP: Allow use of 'inactive'
+ streams for hold This allows use of the 'inactive' stream
+ direction identifier to be used for hold where 'sendonly' is
+ normally used. Some Seimens phones use 'inactive' and this change
+ allows music on hold to operate properly. Review:
+ https://reviewboard.asterisk.org/r/4252/ Reported by: Steve Pitts
+ ........ Merged revisions 429432 from
+ http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+ revisions 429433 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * channels/chan_pjsip.c, res/res_pjsip_session.c,
+ include/asterisk/res_pjsip_session.h, /,
+ res/res_pjsip_session.exports.in: res_pjsip_session: Delay
+ sending BYE if a re-INVITE transaction is in progress. Given the
+ scenario where a PJSIP channel is in a native RTP bridge with
+ direct media and the channel is then hung up the code will
+ currently re-INVITE the channel back to Asterisk and send a BYE
+ at the same time. Many SIP implementations dislike this greatly.
+ This change makes it so that if a re-INVITE transaction is in
+ progress the BYE is queued to occur after the completion of the
+ transaction (be it through normal means or a timeout). Review:
+ https://reviewboard.asterisk.org/r/4248/ ........ Merged
+ revisions 429409 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, channels/chan_pjsip.c: chan_pjsip: Race between channel answer
+ and bridge setup when using direct media When direct media is
+ enabled and a pjsip channel is answered a race would occur
+ between the handling of the answer and bridge setup. Sometimes
+ the media negotiation would take place after the native bridge
+ was setup. This resulted in a NULL media address, which in turn
+ resulted in Asterisk using its address as the remote media
+ address when sending a reinvite. This patch makes the chan_pjsip
+ answer handler synchronous thus alleviating the race condition
+ (the bridge won't start setting things up until after it
+ returns). ASTERISK-24563 #close Reported by: Steve Pitts Review:
+ https://reviewboard.asterisk.org/r/4257/ ........ Merged
+ revisions 429477 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, /,
+ channels/chan_sip.c, include/asterisk/rtp_engine.h: Direct Media
+ calls within private network sometimes get one way audio When
+ endpoints with direct_media enabled, behind a firewall (Asterisk
+ on a separate network) and were bridged sometimes Asterisk would
+ send the ip address of the firewall in the sdp to one of the
+ phones in the reinvite resulting in one way audio. When sending
+ the reinvite Asterisk will retrieve the media address from the
+ associated rtp instance, but if frames were being read this can
+ be overwritten with another address (in this case the
+ firewall's). This patch ensures that Asterisk uses the original
+ device address when using direct media. ASTERISK-24563 Reported
+ by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4216/
+ ........ Merged revisions 429195 from
+ http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+ revisions 429196 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, channels/pjsip/dialplan_functions.c: Ensure the correct value
+ is returned for CHANNEL(pjsip, secure) Prior to this patch, we
+ were using the PJSIP dialog's secure flag to determine if a
+ secure transport was being used. Unfortunately, the dialog's
+ secure flag was only set if a SIPS URI were in use, as required
+ by RFC 3261 sections 12.1.1 and 12.1.2. What we're interested in
+ is not dialog security, but transport security. This code change
+ switches to a model where we use the dialog's target URI to
+ determine what transport would be used to communicate, and then
+ check if that transport is secure. AST-1450 #close Reported by
+ John Bigelow Review: https://reviewboard.asterisk.org/r/4277
+ ........ Merged revisions 429739 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * channels/chan_dahdi.c, /: chan_dahdi: Don't ignore setvar when
+ using configuration section scheme. When the configuration
+ section scheme of chan_dahdi.conf is used (keyword dahdichan
+ instead of channel) all setvar= options are completely ignored.
+ No variable defined this way appears in the created DAHDI
+ channels. * Move the clearing of setvar values to after the
+ deferred processing of dahdichan. AST-1378 #close Reported by:
+ Guenther Kelleter Patch by: Guenther Kelleter ........ Merged
+ revisions 429825 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 429829 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * include/asterisk/lock.h, main/lock.c, /: DEBUG_THREADS: Fix
+ regression and lock tracking initialization problems. This patch
+ started with David Lee's patch at
+ https://reviewboard.asterisk.org/r/2826/ and includes a
+ regression fix introduced by the ASTERISK-22455 patch. The
+ initialization of a mutex's lock tracking structure was not
+ protected in a critical section. This is fine for any mutex that
+ is explicitly initialized, but a static mutex may have its lock
+ tracking double initialized if multiple threads attempt the first
+ lock simultaneously. * Added a global mutex to properly serialize
+ initialization of the lock tracking structure. The painful global
+ lock can be mitigated by adding a double checked lock flag as
+ discussed on the original review request. * Defer lock tracking
+ initialization until first use. * Don't be "helpful" and
+ initialize an uninitialized lock when DEBUG_THREADS is enabled.
+ Debug code is not supposed to fix or change normal code behavior.
+ We don't need a lock initialization race that would force a
+ re-setup of lock tracking. Lock tracking already handles
+ initialization on first use. * Properly handle allocation
+ failures of the lock tracking structure. * No need to initialize
+ tracking data in __ast_pthread_mutex_destroy() just to turn
+ around and destroy it. The regression introduced by
+ ASTERISK-22455 is the result of manipulating a pthread_mutex_t
+ struct outside of the pthread library code. The pthread_mutex_t
+ struct seems to have a global linked list pointer member that can
+ get changed by other threads. Therefore, saving and restoring the
+ contents of a pthread_mutex_t struct is a bad thing. Thanks to
+ Thomas Airmont for finding this obscure regression. * Don't
+ overwrite the struct ast_lock_track.reentr_mutex member to
+ restore tracking data in __ast_cond_wait() and
+ __ast_cond_timedwait(). The pthread_mutex_t struct must be
+ treated as a read-only opaque variable. Miscellaneous other items
+ fixed by this patch: * Match ast_suspend_lock_info() with
+ ast_restore_lock_info() in __ast_cond_timedwait(). * Made some
+ uninitialized lock sanity checks return EINVAL and try a
+ DO_THREAD_CRASH. * Fix bad canlog initialization expressions.
+ ASTERISK-24614 #close Reported by: Thomas Airmont Review:
+ https://reviewboard.asterisk.org/r/4247/ Review:
+ https://reviewboard.asterisk.org/r/2826/ ........ Merged
+ revisions 429539 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 429540 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, res/res_pjsip_pubsub.c: Activate persistent subscriptions when
+ they are recreated. Prior to this change, recreating persistent
+ subscriptions would create the subscription but would not
+ activate it. This led to subscriptions being listed in the "NULL"
+ state by diagnostics and not sending NOTIFYs when expected.
+ Review: https://reviewboard.asterisk.org/r/4261 ........ Merged
+ revisions 429571 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * asterisk-13.1.0-summary.txt (removed), /,
+ asterisk-13.1.0-summary.html (removed): Update properties; remove
+ old summaries
+
+ * / (added): Create Certified Asterisk 13.1 branch
+
2014-12-15 Asterisk Development Team <asteriskteam at digium.com>
* Asterisk 13.1.0 Released.
Modified: certified/tags/13.1-cert1-rc2/contrib/realtime/mysql/mysql_config.sql
URL: http://svnview.digium.com/svn/asterisk/certified/tags/13.1-cert1-rc2/contrib/realtime/mysql/mysql_config.sql?view=diff&rev=430393&r1=430392&r2=430393
==============================================================================
--- certified/tags/13.1-cert1-rc2/contrib/realtime/mysql/mysql_config.sql (original)
+++ certified/tags/13.1-cert1-rc2/contrib/realtime/mysql/mysql_config.sql Thu Jan 8 12:30:50 2015
@@ -703,3 +703,9 @@
UPDATE alembic_version SET version_num='eb88a14f2a';
+-- Running upgrade eb88a14f2a -> 371a3bf4143e
+
+ALTER TABLE ps_endpoints ADD COLUMN user_eq_phone ENUM('yes','no');
+
+UPDATE alembic_version SET version_num='371a3bf4143e';
+
Modified: certified/tags/13.1-cert1-rc2/contrib/realtime/oracle/oracle_config.sql
URL: http://svnview.digium.com/svn/asterisk/certified/tags/13.1-cert1-rc2/contrib/realtime/oracle/oracle_config.sql?view=diff&rev=430393&r1=430392&r2=430393
==============================================================================
--- certified/tags/13.1-cert1-rc2/contrib/realtime/oracle/oracle_config.sql (original)
+++ certified/tags/13.1-cert1-rc2/contrib/realtime/oracle/oracle_config.sql Thu Jan 8 12:30:50 2015
@@ -984,7 +984,17 @@
/
-INSERT INTO alembic_version (version_num) VALUES ('eb88a14f2a')
+-- Running upgrade eb88a14f2a -> 371a3bf4143e
+
+ALTER TABLE ps_endpoints ADD user_eq_phone VARCHAR(3 CHAR)
+
+/
+
+ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (user_eq_phone IN ('yes', 'no'))
+
+/
+
+INSERT INTO alembic_version (version_num) VALUES ('371a3bf4143e')
/
Modified: certified/tags/13.1-cert1-rc2/contrib/realtime/postgresql/postgresql_config.sql
URL: http://svnview.digium.com/svn/asterisk/certified/tags/13.1-cert1-rc2/contrib/realtime/postgresql/postgresql_config.sql?view=diff&rev=430393&r1=430392&r2=430393
==============================================================================
--- certified/tags/13.1-cert1-rc2/contrib/realtime/postgresql/postgresql_config.sql (original)
+++ certified/tags/13.1-cert1-rc2/contrib/realtime/postgresql/postgresql_config.sql Thu Jan 8 12:30:50 2015
@@ -733,7 +733,11 @@
ALTER TABLE ps_endpoints ADD COLUMN media_encryption_optimistic yesno_values;
-INSERT INTO alembic_version (version_num) VALUES ('eb88a14f2a');
+-- Running upgrade eb88a14f2a -> 371a3bf4143e
+
+ALTER TABLE ps_endpoints ADD COLUMN user_eq_phone yesno_values;
+
+INSERT INTO alembic_version (version_num) VALUES ('371a3bf4143e');
COMMIT;
Modified: certified/tags/13.1-cert1-rc2/contrib/realtime/sqlserver/mssql_config.sql
URL: http://svnview.digium.com/svn/asterisk/certified/tags/13.1-cert1-rc2/contrib/realtime/sqlserver/mssql_config.sql?view=diff&rev=430393&r1=430392&r2=430393
==============================================================================
--- certified/tags/13.1-cert1-rc2/contrib/realtime/sqlserver/mssql_config.sql (original)
+++ certified/tags/13.1-cert1-rc2/contrib/realtime/sqlserver/mssql_config.sql Thu Jan 8 12:30:50 2015
@@ -982,7 +982,17 @@
GO
-INSERT INTO alembic_version (version_num) VALUES ('eb88a14f2a');
+-- Running upgrade eb88a14f2a -> 371a3bf4143e
+
+ALTER TABLE ps_endpoints ADD user_eq_phone VARCHAR(3) NULL;
+
+GO
+
+ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (user_eq_phone IN ('yes', 'no'));
+
+GO
+
+INSERT INTO alembic_version (version_num) VALUES ('371a3bf4143e');
GO
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