[svn-commits] bebuild: tag 12.2.0-rc1 r411558 - in /tags/12.2.0-rc1: ./ contrib/realtime/my...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Mar 28 13:36:57 CDT 2014
Author: bebuild
Date: Fri Mar 28 13:36:53 2014
New Revision: 411558
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=411558
Log:
Importing files for 12.2.0-rc1 release.
Added:
tags/12.2.0-rc1/.lastclean (with props)
tags/12.2.0-rc1/.version (with props)
tags/12.2.0-rc1/ChangeLog (with props)
tags/12.2.0-rc1/contrib/realtime/mysql/mysql_config.sql (with props)
tags/12.2.0-rc1/contrib/realtime/mysql/mysql_voicemail.sql (with props)
tags/12.2.0-rc1/contrib/realtime/oracle/oracle_config.sql (with props)
tags/12.2.0-rc1/contrib/realtime/oracle/oracle_voicemail.sql (with props)
tags/12.2.0-rc1/contrib/realtime/postgresql/postgresql_config.sql (with props)
tags/12.2.0-rc1/contrib/realtime/postgresql/postgresql_voicemail.sql (with props)
tags/12.2.0-rc1/contrib/realtime/sqlserver/mssql_config.sql (with props)
tags/12.2.0-rc1/contrib/realtime/sqlserver/mssql_voicemail.sql (with props)
Added: tags/12.2.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/12.2.0-rc1/.lastclean?view=auto&rev=411558
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Added: tags/12.2.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/12.2.0-rc1/ChangeLog?view=auto&rev=411558
==============================================================================
--- tags/12.2.0-rc1/ChangeLog (added)
+++ tags/12.2.0-rc1/ChangeLog Fri Mar 28 13:36:53 2014
@@ -1,0 +1,25963 @@
+2014-03-28 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 12.2.0-rc1 Released.
+
+2014-03-28 18:09 +0000 [r411534] Matthew Jordan <mjordan at digium.com>
+
+ * include/asterisk/res_hep.h (added), res/res_hep_pjsip.c (added),
+ res/res_hep.exports.in (added), CHANGES, configs/hep.conf.sample
+ (added), res/res_hep.c (added): res_hep/res_hep_pjsip: Add a
+ HEPv3 capture agent module and a logger for PJSIP This patch adds
+ the following: (1) A new module, res_hep, which implements a
+ generic packet capture agent for the Homer Encapsulation Protocol
+ (HEP) version 3. Note that this code is based on a patch provided
+ by Alexandr Dubovikov; I basically just wrapped it up, added
+ configuration via the configuration framework, and threw in a
+ taskprocessor. (2) A new module, res_hep_pjsip, which forwards
+ all SIP message traffic that passes through the res_pjsip stack
+ over to res_hep for encapsulation and transmission to a HEPv3
+ capture server. Much thanks to Alexandr for his Asterisk patch
+ for this code and for a *lot* of patience waiting for me to port
+ it to 12/trunk. Due to some dithering on my part, this has taken
+ the better part of a year to port forward (I still blame CDRs for
+ the delay). ASTERISK-23557 #close Review:
+ https://reviewboard.asterisk.org/r/3207/
+
+2014-03-28 17:52 +0000 [r411532] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c,
+ addons/chan_ooh323.c, /: process stack command even if gatekeeper
+ client isn't register don't destroy gatekeeper client if it is
+ not started don't destroy gatekeeper client in some sort of
+ gatekeeper errors signal rtp create condition when call cleared
+ before rtp structure created (closes issue ASTERISK-23460)
+ Reported by: Dmitry Melekhov Patches: ASTERISK-23460-2.patch
+ Tested by: Dmitry Melekhov ........ Merged revisions 411531 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-28 17:35 +0000 [r411529] Matthew Jordan <mjordan at digium.com>
+
+ * rest-api/api-docs/applications.json,
+ rest-api/api-docs/playbacks.json, UPGRADE.txt,
+ rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
+ rest-api/resources.json, CHANGES, include/asterisk/manager.h,
+ rest-api/api-docs/bridges.json,
+ rest-api/api-docs/recordings.json,
+ rest-api/api-docs/deviceStates.json,
+ rest-api/api-docs/endpoints.json,
+ rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
+ rest-api/api-docs/asterisk.json: Update API versions and
+ UPGRADE/CHANGES for 12.2.0 This patch does the following: * It
+ updates the AMI version to 2.2.0 to indicate backwards compatible
+ changes have been made since the last release * It updates the
+ ARI version to 1.2.0 to indicate backwards compatible changes
+ have been made since the last release * It updates the
+ UPGRADE/CHANGES files with changes that were not mentioned
+
+2014-03-28 17:08 +0000 [r411514] Mark Michelson <mmichelson at digium.com>
+
+ * contrib/ast-db-manage/config/versions/3855ee4e5f85_add_missing_pjsip_options.py
+ (added): Add alembic script that adds contact user_agent and
+ endpoint message_context.
+
+2014-03-28 16:48 +0000 [r411512] Matthew Jordan <mjordan at digium.com>
+
+ * /, res/res_odbc.exports.in, UPGRADE.txt, res/res_odbc.c,
+ configs/res_odbc.conf.sample, include/asterisk/res_odbc.h,
+ res/res_config_odbc.c: res_config_odbc/res_odbc: Fix handling of
+ non-text columns updates with empty values. This patch fixes
+ setting nullable integer columns to NULL instead of an empty
+ string, which fails for PostgreSQL, for example. The current code
+ is supposed to do so, but the check is broken. The patch also
+ allows the first column in the list to be a nullable integer.
+ This patch also adds a compatibility setting in res_odbc.conf,
+ allow_empty_string_in_nontext. It is enabled by default. It
+ should be disabled for database backends (such as PostgreSQL)
+ that require NULL instead of an empty string for Integer columns.
+ Review: https://reviewboard.asterisk.org/r/3375 (issue
+ ASTERISK-23459) Reported by: zvision patches:
+ res_config_odbc.diff uploaded by zvision (License 5755) ........
+ Merged revisions 411399 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 411408 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-28 16:17 +0000 [r411465] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/tcptls.c, main/manager.c, /, main/http.c: http: response
+ body often missing after specific request This patch works around
+ a problem with the HTTP body being dropped from the response to a
+ specific client and under specific circumstances: a) Client
+ request comes from node.js user agent "Shred" via use of
+ swagger-client library. b) Asterisk and Client are *not* on the
+ same host or TCP/IP stack In testing this problem, it has been
+ determined that the write of the HTTP body is lost, even if the
+ data is written using low level write function. The only solution
+ found is to instruct the TCP stack with the shutdown function to
+ flush the last write and finish the transmission. See review for
+ more details. ASTERISK-23548 #close (closes issue ASTERISK-23548)
+ Reported by: Sam Galarneau Review:
+ https://reviewboard.asterisk.org/r/3402/ ........ Merged
+ revisions 411462 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 411463 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-28 16:00 +0000 [r411374-411461] Matthew Jordan <mjordan at digium.com>
+
+ * /: Remove block on 411408
+
+ * /, UPGRADE.txt: UPGRADE: Note IAX2 compatibility issue between
+ 1.4 and 1.8+ systems. ........ Merged revisions 411457 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 411458 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * contrib/realtime/mysql/voicemail_messages.sql (removed),
+ contrib/realtime/postgresql/realtime.sql (removed),
+ contrib/realtime/mysql/voicemail_data.sql (removed),
+ contrib/realtime/mysql/musiconhold.sql (removed),
+ contrib/realtime/mysql/queue_log.sql (removed),
+ contrib/realtime/mysql/voicemail.sql (removed),
+ contrib/realtime/mysql/sippeers.sql (removed),
+ contrib/realtime/mysql/iaxfriends.sql (removed),
+ contrib/realtime/mysql/meetme.sql (removed): contrib/realtime:
+ Remove empty SQL script files Since the relatime scripts are now
+ managed by Alembic, the previous realtime scripts were previously
+ removed. However, the removal process messed up, as the files
+ were still in the repository. The contents were just empty. This
+ removes the files from the tree.
+
+ * channels/sip/include/sip.h, /: chan_sip: Add MESSAGE request to
+ allowed methods The allowed methods advertised by chan_sip did
+ not previously note the MESSAGE request. Even in Asterisk 1.8, we
+ do accept in-dialog MESSAGE requests; we should advertise that we
+ support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504
+ #comment Reported by: Martin Kontsek ASTERISK-23504 #comment
+ Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)
+ Review: https://reviewboard.asterisk.org/r/3396/ ........ Merged
+ revisions 411372 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 411373 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-27 19:15 +0000 [r411311-411315] Corey Farrell <git at cfware.com>
+
+ * main/message.c, apps/app_jack.c, funcs/func_dialplan.c,
+ channels/chan_sip.c, funcs/func_math.c,
+ funcs/func_jitterbuffer.c, res/res_mutestream.c,
+ funcs/func_global.c, apps/app_speech_utils.c,
+ res/res_pjsip_header_funcs.c, funcs/func_callcompletion.c,
+ funcs/func_blacklist.c, funcs/func_cdr.c, funcs/func_channel.c,
+ apps/app_stack.c, funcs/func_callerid.c, res/res_calendar.c,
+ apps/app_voicemail.c, funcs/func_speex.c, /,
+ funcs/func_strings.c, res/res_xmpp.c, res/res_jabber.c,
+ main/features_config.c, channels/chan_iax2.c,
+ apps/confbridge/conf_config_parser.c,
+ channels/pjsip/dialplan_functions.c, funcs/func_groupcount.c,
+ funcs/func_pitchshift.c, funcs/func_odbc.c, funcs/func_volume.c,
+ funcs/func_frame_trace.c: Fix dialplan function NULL channel
+ safety issues (closes issue ASTERISK-23391) Reported by: Corey
+ Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........
+ Merged revisions 411313 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 411314 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * include/asterisk.h, /, main/format.c: main/formats: Fix crash in
+ ast_format_cmp during non-clean shutdown. * Update asterisk.h to
+ reflect availability of ast_register_cleanup in 11.9. * Use
+ ast_register_cleanup for format_attr_shutdown. (closes issue
+ ASTERISK-23103) Reported by: JoshE ........ Merged revisions
+ 411310 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-27 14:20 +0000 [r411295] Mark Michelson <mmichelson at digium.com>
+
+ * main/sorcery.c: Give sorcery instances a reference to their
+ wizards. On graceful shutdown, sorcery wizards are all killed
+ off, but it is possible for sorcery instances to still have
+ dangling pointers after this, possibly causing a crash. Giving
+ the sorcery instances a reference to their wizards ensures that
+ the wizard reference will remain valid for the lifetime of the
+ sorcery instance. Review: https://reviewboard.asterisk.org/r/3401
+
+2014-03-26 22:44 +0000 [r411245] Joshua Colp <jcolp at digium.com>
+
+ * /, main/say.c: say: Fix a bug where SayNumber in Polish tries to
+ play incorrect sound. This change fixes a bug where calling
+ SayNumber with a number divisible by 100 using the Polish
+ language would cause the code to attempt to play a sound file
+ with an empty name. (closes issue ASTERISK-23509) Reported by:
+ zvision Review: https://reviewboard.asterisk.org/r/3378/ ........
+ Merged revisions 411243 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 411244 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-26 16:07 +0000 [r411193] Jonathan Rose <jrose at digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: chan_sip: Send
+ real CallerID information with P-Assserted-Identity (RFC-3325)
+ Prior too this patch, the P-Asserted-Identity header would
+ include anonymous caller id information which seems to go against
+ the point of the P-Asserted-Identity header. Now the real caller
+ ID information will be included in this header. Also, no privacy
+ header would be included. This patch adds 'Privacy: id' to
+ outgoing SIP messages that include the P-Asserted-Identity
+ header. (closes issue AST-1301) ........ Merged revisions 411189
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 411190 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-26 16:03 +0000 [r411191] Richard Mudgett <rmudgett at digium.com>
+
+ * contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py:
+ Fix 'alembic branches' merge conflict as described by the web
+ page.
+
+2014-03-25 18:43 +0000 [r411173] Sean Bright <sean at malleable.com>
+
+ * res/ari/config.c: ARI: Don't complain about missing ARI users
+ when we aren't enabled Currently, if ARI is not enabled it will
+ still complain that there are no configured users. This patch
+ checks to see if ARI is enabled before logging and error or
+ iterating the container to validate the users. Review:
+ https://reviewboard.asterisk.org/r/3391/
+
+2014-03-25 17:52 +0000 [r411157-411159] Mark Michelson <mmichelson at digium.com>
+
+ * tests/test_sorcery.c, tests/test_sorcery_realtime.c,
+ main/sorcery.c, res/res_mwi_external.c,
+ res/res_pjsip/config_system.c, configs/sorcery.conf.sample,
+ main/bucket.c, include/asterisk/sorcery.h,
+ res/res_pjsip/pjsip_configuration.c, tests/test_sorcery_astdb.c:
+ Prevent duplicate sorcery wizards from being applied to sorcery
+ object types. This commit contains several changes to sorcery: 1)
+ Application of sorcery configuration based on module name is
+ automatically performed when sorcery is opened for a module. 2)
+ Sorcery will not attempt to apply the same wizard to an object
+ type more than once. 3) Sorcery gives more exact results when
+ attempting to apply a wizard, whether as the default or based on
+ configuration. Sorcery unit tests still pass for me after making
+ these changes. Review: https://reviewboard.asterisk.org/r/3326
+
+ * res/res_pjsip/pjsip_configuration.c, UPGRADE.txt,
+ res/res_pjsip_messaging.c, res/res_pjsip.c,
+ include/asterisk/res_pjsip.h: Add a "message_context" option for
+ PJSIP endpoints.
+
+2014-03-25 16:55 +0000 [r411141] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/res_pjsip.h, res/res_pjsip/pjsip_options.c,
+ res/res_pjsip.c: res_pjsip: Fix contact authenticate_qualify
+ endpoint lookup when qualifing a contact. * Fixed bad use of
+ ao2_find() in on_endpoint(). * Replaced use of find_endpoints()
+ with find_an_endpoint() since only the first found endpoint is
+ ever needed. * Fixed qualify_contact_cb() to update the contact
+ with the aor authenticate_qualify setting. Otherwise, permanent
+ contacts in the aor type sections would have a config line order
+ dependancy. * Fixed off nominal path contact ref leak in
+ qualify_contact(). The comment saying the unref is not needed was
+ wrong. * Fixed off nominal path use of the endpoint parameter if
+ it is NULL in send_out_of_dialog_request(). * Added missing off
+ nominal path unref of pjsip tdata in
+ send_out_of_dialog_request(). * Fixed off nominal path failing to
+ call the callback in send_request_cb() when the request is
+ challenged for authentication. * Eliminated silly RAII_VAR() use
+ in qualify_contact_cb(). * Updated ast_sip_send_request() doxygen
+ to better reflect reality. (closes issue ASTERISK-23254) Reported
+ by: rmudgett Review: https://reviewboard.asterisk.org/r/3381/
+
+2014-03-25 16:04 +0000 [r411091] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Fix incorrect use of timers If
+ update_provisional_keepalive() is called while
+ send_provisional_keepalive_full() is waiting on the PVT lock,
+ then pvt->provisional_keepalive_sched_id will be changed to a new
+ sched_id value by update_provisional_keepalive(), but that new
+ sched_id then may be overwritten with -1 by
+ send_provisional_keepalive_full(), killing the pvt's reference to
+ a schedule and "leaking" the reference. (closes issue
+ ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/
+ Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
+ Patches: provisional_keepalive_fix.diff uploaded by Steve Davies
+ (license 5012) ........ Merged revisions 411088 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 411089 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-25 15:44 +0000 [r411086] Jonathan Rose <jrose at digium.com>
+
+ * res/res_stasis.c: ARI: Resolve a subscription leak against
+ implicit bridge subscriptions When a channel in a stasis
+ application is joined to a bridge, a subscription for that bridge
+ is created implicitly for the stasis application serving the
+ channel. Prior to this patch, subsequent removals of the channel
+ from the bridge would leave the subscription open. Review:
+ https://reviewboard.asterisk.org/r/3380/
+
+2014-03-24 21:38 +0000 [r411023] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Always use fromdomain if set
+ for domain, even if callerid is set to restricted. (closes issue
+ ASTERISK-20841) Reported by: Kelly Goedert ........ Merged
+ revisions 411021 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 411022 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-21 16:01 +0000 [r410995] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_pjsip_registrar.c: res_pjsip_registrar.c: Miscellaneous
+ cleanup in rx_task(). * Fix variable shadowing of 'updated' by
+ renaming it to 'contact_update'. * Checked 'contact_update' for
+ ast_sorcery_copy() failure. * Removed silly use of RAII_VAR() for
+ 'contact_update'.
+
+2014-03-20 22:54 +0000 [r410966] Jonathan Rose <jrose at digium.com>
+
+ * /, apps/app_confbridge.c: app_confbridge: Fix bug - users with
+ startmuted set don't start muted (closes issue ASTERISK-23461)
+ Reported by: Chico Manobela Review:
+ https://reviewboard.asterisk.org/r/3373/ ........ Merged
+ revisions 410965 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-20 16:27 +0000 [r410949] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/channel.h, res/ari/resource_channels.c,
+ res/res_stasis_snoop.c, include/asterisk/rtp_engine.h,
+ main/dial.c, main/manager.c, main/channel_internal_api.c,
+ main/core_unreal.c: assigned-uniqueids: Miscellaneous cleanup and
+ fixes. * Fix memory leak in ast_unreal_new_channels(). Made it
+ generate the ;2 uniqueid on a stack variable instead of mallocing
+ it. * Made send error response to ARI and AMI requests instead of
+ just logging excessive uniqueid length and allowing truncation.
+ action_originate() and ari_channels_handle_originate_with_id(). *
+ Fixed minor truncating uniqueid hole when generating the ;2
+ uniqueid string length. Created public and internal lengths of
+ uniqueid. The internal length can handle a max public uniqueid
+ plus an appended ;2. * free() and ast_free() are NULL tolerant so
+ they don't need a NULL test before calling. * Made use better
+ struct initialization format instead of the position dependent
+ initialization format. Also anything not explicitly initialized
+ in the struct is initialized to zero by the compiler. * Made
+ ast_channel_internal_set_fake_ids() use the safer
+ ast_copy_string() instead of strncpy(). Review:
+ https://reviewboard.asterisk.org/r/3371/
+
+2014-03-19 17:26 +0000 [r410933] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_pjsip_endpoint_identifier_ip.c: PJSIP: Allow for identify
+ sections to be specified in sorcery.conf. "identify" is a special
+ type of configuration object in PJSIP because unlike the other
+ objects, it is not provided by the base res_pjsip module.
+ Instead, it is provided by the res_pjsip_endpoint_identifier_ip
+ module. If using the default sorcery wizard
+ (config,criteria=type=identify) then things work because the
+ module that applies the default wizard is the correct module.
+ However, if attempting to use sorcery.conf to apply an alternate
+ wizard, it was not possible. If you attempted to specify the
+ identify object type in the res_pjsip section, then the object
+ could not be registered since the object was undocumented for the
+ res_pjsip module. There was no alternate configuration section
+ defined for it, so you were out of luck if you wanted to override
+ the default wizard. With this change, the identify section will
+ properly have a sorcery.conf-based wizard applied when the
+ identify definition is within the
+ res_pjsip_endpoint_identifier_ip section.
+
+2014-03-19 14:24 +0000 [r410904-410918] Joshua Colp <jcolp at digium.com>
+
+ * res/res_stasis.c: res_stasis: Fix a bug where the default bridge
+ type was not set.
+
+ * CHANGES, res/res_stasis.c, rest-api/api-docs/bridges.json,
+ res/ari/resource_bridges.h: res_stasis: Extend bridge type to be
+ a comma separated list of bridge attributes. This change turns
+ the bridge type field into a comma separated list of attributes.
+ These attributes include: mixing, holding, dtmf_events, and
+ proxy_media. By setting the various attributes a user can control
+ the type of bridge created with the behavior they need for their
+ application. (closes issue ASTERISK-23437) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/3359/
+
+2014-03-19 02:29 +0000 [r410890] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_ari.c: res_ari: Fix documentation schema error
+
+2014-03-18 23:31 +0000 [r410876] Rusty Newton <rnewton at digium.com>
+
+ * res/res_ari.c: res_ari: Add notes about Asterisk HTTP server to
+ the "enabled" config option for the res_ari general section Added
+ note and see-also reminding user to enable the HTTP server.
+ (closes issue ASTERISK-22499) Reported by: Rusty Newton
+
+2014-03-18 15:28 +0000 [r410861] Matthew Jordan <mjordan at digium.com>
+
+ * main/cdr.c: cdr: Add asserts for when we don't know about a CDR
+ for a channel In the CDR core, every channel should either be
+ filtered out (due to being an 'internal' channel used as an
+ implementation detail, such as playing media back into a bridge)
+ or it should get a CDR. Even if that CDR ends up being discarded,
+ we still give the channel a CDR in case we end up needing it. If
+ we hit a situation where a channel does not have a CDR, we should
+ blow up in -dev-mode. Asserts are appropriate for that. This
+ patch adds those asserts, as they would have quickly caught the
+ error fixed by r410814.
+
+2014-03-18 14:51 +0000 [r410858] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/http.c: ARI: allow json content type with zero length body
+ When a request was received with a Content-type of json, the body
+ was sent for json parsing - even if it was zero length. This
+ resulted in ARI requests failing that were valid, such as a
+ channel DELETE with no parameters. The code has now been changed
+ to skip json parsing with zero content length. (closes issue
+ SWP-6748) Reported by: Samuel Galarneau Review:
+ https://reviewboard.asterisk.org/r/3360/
+
+2014-03-18 12:45 +0000 [r410844] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip/config_system.c: res_pjsip: Fix memory leak of
+ nameservers in off-nominal resolver creation failure. Thanks
+ Walter Doekes!
+
+2014-03-18 11:51 +0000 [r410830] Sean Bright <sean at malleable.com>
+
+ * res/res_fax_spandsp.c, /: res_fax_spandsp: Use g711_free() when
+ available. Per Johann Steinwendtner on the asterisk-dev mailing
+ list:
+ http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html
+ g711_free() was introduced in spandsp 0.0.6pre4 and
+ g711_release() became a noop. I opted not to remove the call to
+ g711_release() since it is harmless and to call g711_free() if we
+ have a sufficiently recent version of spandsp. (issue
+ ASTERISK-20149) Reported by: Alexandr Gordeev ........ Merged
+ revisions 410829 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-18 02:02 +0000 [r410813] Richard Mudgett <rmudgett at digium.com>
+
+ * main/stasis_cache.c: stasis_cache: Use the right variable in the
+ cache entry ao2 cmp function.
+
+2014-03-17 22:53 +0000 [r410793-410795] Joshua Colp <jcolp at digium.com>
+
+ * CHANGES, res/res_pjsip/include/res_pjsip_private.h,
+ res/res_pjsip.c, main/dns.c, res/res_pjsip/config_system.c,
+ include/asterisk/dns.h: res_pjsip: Enable PJSIP DNS client
+ support. This change enables DNS client support within PJSIP.
+ System nameservers are automatically discovered using res_init or
+ res_ninit. If this fails then PJSIP will resort to using
+ gethostbyname for resolution. By enabling this support we gain
+ SRV support, failover, and weight support. (closes issue
+ ASTERISK-23435) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/3343/
+
+ * res/res_pjsip_multihomed.c: res_pjsip_multihomed: Make address
+ replacement less aggressive. This change makes the
+ res_pjsip_multihomed module less aggressive when changing the
+ address in messages. It will now only occur if the transport in
+ use is bound to the any address OR if the system determined
+ source address matches the bound address of the transport in use.
+ Review: https://reviewboard.asterisk.org/r/3369/
+
+2014-03-17 21:56 +0000 [r410747-410750] Russ Meyerriecks <rmeyerreicks at digium.com>
+
+ * /, main/callerid.c: !fixup: callerid: Logic error in checksum
+ processing Fixes syntax error in previous commit :-( ........
+ Merged revisions 410748 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 410749 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/callerid.c, /: callerid: Logic error in checksum processing
+ Callerid checksum-ing was being handled incorrectly here. When
+ the checksum is calculated to be 0x00, it will perform 0x100-0x00
+ which results in 0x100. This value will then fail the otherwise
+ correct callerid message. This patch changes the logic to simply
+ add the calculated checksum to the transmitted 2's compliment
+ checksum. Review: https://reviewboard.asterisk.org/r/3356/
+ (closes issue ASTERISK-23488) ........ Merged revisions 410710
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 410717 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-17 18:36 +0000 [r410673-410696] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_mwi_external.c, res/res_pjsip/config_system.c,
+ configs/sorcery.conf.sample, include/asterisk/sorcery.h,
+ res/res_pjsip/pjsip_configuration.c, tests/test_sorcery_astdb.c,
+ tests/test_sorcery.c, tests/test_sorcery_realtime.c,
+ main/sorcery.c: Revert changes to sorcery that accidentally got
+ committed. These changes were still up for review and have not
+ been approved yet. I must have had the changes in my working copy
+ when making a different change.
+
+ * tests/test_sorcery.c, main/channel.c,
+ res/res_pjsip/config_system.c, res/res_mwi_external.c,
+ include/asterisk/bridge_channel.h, funcs/func_frame_trace.c,
+ configs/sorcery.conf.sample, res/res_pjsip/pjsip_configuration.c,
+ include/asterisk/sorcery.h, tests/test_sorcery_astdb.c,
+ include/asterisk/frame.h, main/bridge_channel.c,
+ tests/test_sorcery_realtime.c, main/sorcery.c,
+ res/res_stasis_playback.c, main/frame.c,
+ bridges/bridge_softmix.c: Fix stuck channel in ARI through the
+ introduction of synchronous bridge actions. Playing back a file
+ to a channel in an ARI bridge would attempt to wait until the
+ playback concluded before returning. The method used involved
+ signaling the waiting thread in the ARI custom playback function.
+ The problem with this is that there were some corner cases that
+ were not accounted for: * If a bridge channel could not be found,
+ then we never would attempt the playback but would still attempt
+ to wait for the playback to complete. * If the bridge playfile
+ action failed to queue, we would still attempt to wait for the
+ playback to complete. * If the bridge playfile action were queued
+ but some circumstance caused the playback not to occur (the
+ bridge dies, the channel is removed from the bridge), then we
+ would never be notified. The solution to this is to move the
+ waiting logic into the bridge code. A new bridge API function is
+ added to queue a synchronous action on a bridge. The waiting
+ thread is notified when the queued frame has been freed, either
+ due to an error occurring or due to successful playback. As a
+ failsafe, the waiting thread has a 10 minute timeout just in case
+ there is a frame leak somewhere. Review:
+ https://reviewboard.asterisk.org/r/3338
+
+2014-03-17 16:42 +0000 [r410671] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/confbridge/conf_chan_announce.c: app_confbridge: Add missing
+ destructor call to announcer channel destructor.
+
+2014-03-16 20:20 +0000 [r410650] Matthew Jordan <mjordan at digium.com>
+
+ * res/stasis/app.c: stasis/app.c: Add some extra debugging for
+ subscription counts Events are sent to a connected ARI
+ application based on the things that ARI application cares about.
+ These subscriptions can be set up implicitly - such as when that
+ ARI application creates a new object - or explicitly, via the
+ application resource's subscription operations. Debugging *why*
+ something was being sent to an application - or why something was
+ not being sent to an application - was a bit tricky, as there was
+ no debug information for the subscriptions. This patch adds some
+ debug level 3 statements that show the subscription counts for
+ applications. (Level 3 was chosen as it matches the verbose level
+ 3 statements elsewhere)
+
+2014-03-14 21:55 +0000 [r410625] Mark Michelson <mmichelson at digium.com>
+
+ * tests/test_sorcery_realtime.c: Fix failing realtime sorcery
+ tests. The store realtime callback needs to return a positive
+ value for sorcery to treat the store as a success.
+
+2014-03-14 21:28 +0000 [r410623] Jonathan Rose <jrose at digium.com>
+
+ * main/manager.c, /: manager: fix memory leak in manager_add_filter
+ function (closes issue ASTERISK-23420) Reported by: Etienne
+ Lessard Patches: manager_eventfilter_leak uploaded by Etienne
+ Lessard (license 6394) ........ Merged revisions 410609 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-14 20:53 +0000 [r410590-410607] Mark Michelson <mmichelson at digium.com>
+
+ * main/db.c, /: Remove an extra ast_cond_wait() that slipped
+ through the patch. ........ Merged revisions 410606 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/config.c, res/res_sorcery_realtime.c: Handle the return
+ values of realtime updates and stores more accurately. Realtime
+ backends' update and store callbacks return the number of rows
+ affected, or -1 if there was a failure. There were a couple of
+ issues: * The config API was treating 0 as a successful return,
+ and positive values as a failure. Now the config API treats
+ anything >= 0 as a success. * res_sorcery_realtime was treating 0
+ as a successful return from the store procedure, and any positive
+ values as a failure. Now sorcery treats anything > 0 as a
+ success. It still considers 0 a "failure" since there is no
+ change to report to observers. Review:
+ https://reviewboard.asterisk.org/r/3341
+
+ * res/res_pjsip_mwi.c: Prevent conflicts regarding unsolicited and
+ solicited MWI to an endpoint. If an endpoint is receiving
+ unsolicited MWI for a mailbox and then attempts to subscribe to
+ an AOR that provides MWI for the same mailbox, then the SUBSCRIBE
+ is rejected with a 500 response. Review:
+ https://reviewboard.asterisk.org/r/3345
+
+2014-03-14 17:56 +0000 [r410588] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * CHANGES: uniqueid: Update CHANGES to reflect new features Note
+ the new features provided by uniqueid in the CHANGES file. (issue
+ ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3316/
+
+2014-03-14 16:26 +0000 [r410574] Jonathan Rose <jrose at digium.com>
+
+ * CHANGES, res/res_pjsip/config_transport.c,
+ include/asterisk/acl.h, main/acl.c,
+ res/res_pjsip/pjsip_configuration.c: PJSIP: TOS values should be
+ represented as decimals in sorcery objects (closes issue
+ ASTERISK-23235) Reported by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3324/
+
+2014-03-14 16:11 +0000 [r410559] Mark Michelson <mmichelson at digium.com>
+
+ * main/db.c, /: Prevent delayed astdb syncs. The syncing thread
+ sleeps for a second before waiting to be told to attempt to sync
+ again. If a signal were sent during this sleeping period, we
+ would end up having to wait until the next sync signal occurred
+ in order to sync up the astdb. This code rearrangement also
+ ensures that any pending transactions will be synced prior to
+ Asterisk shutting down. Patches: db_sync.patch by John Hardin
+ (License #6512) ........ Merged revisions 410556 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-14 16:05 +0000 [r410558] Jonathan Rose <jrose at digium.com>
+
+ * res/ari/resource_bridges.c: ARI/bridges: Forward
+ Playback/Recording Started/Finished to bridge topic (closes issue
+ ASTERISK-23444) Reported by: Ben Merrills Review:
+ https://reviewboard.asterisk.org/r/3340/
+
+2014-03-14 15:55 +0000 [r410541-410555] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/app.h, res/res_mwi_external.c, main/app.c:
+ res_mwi_external: Clear the stasis cache entry when the external
+ MWI is deleted. One of the things missing when external MWI
+ support was added was the ability to clear the stasis cache entry
+ of deleted external MWI mailboxes. Review:
+ https://reviewboard.asterisk.org/r/3325/
+
+ * main/cdr.c: cdr.c: Add missing aow_unlock(cdr) in off nominal
+ path of handle_dial_message(). * Trivial common code hoisting in
+ handle_bridge_leave_message(). * Some whitespace fixing.
+
+2014-03-13 19:30 +0000 [r410527] Kinsey Moore <kmoore at digium.com>
+
+ * res/stasis/control.c, res/stasis/control.h, res/res_stasis.c:
+ ARI: Ensure managing application receives ChannelEnteredBridge
+ messages This fixes an issue where a Stasis application running
+ over ARI and subscribed to ari/events could miss the
+ ChannelEnteredBridge event because it did not subscribe to the
+ new bridge fast enough. To accomplish this, it subscribes the
+ application controlling the channel to the new bridge before
+ adding it to that bridge which required the stasis_app_control
+ structure to maintain a reference to the stasis_app. (closes
+ issue ASTERISK-23295) Review:
+ https://reviewboard.asterisk.org/r/3336/
+
+2014-03-13 13:24 +0000 [r410509-410510] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_multihomed.c: res_pjsip_multihomed: Remove change
+ for testing fix.
+
+ * res/res_pjsip_multihomed.c: res_pjsip_multihomed: Fix a bug where
+ the 200 OK for a REGISTER would contain the wrong contact.
+
+2014-03-12 19:05 +0000 [r410491-410493] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_musiconhold.c, main/channel.c: res_musiconhold.c:
+ Generate MOH start/stop events whenever the MOH stream is
+ started/stopped. * Made res_musiconhold.c always post the
+ MusicOnHoldStart/MusicOnHoldStop events when it actually
+ starts/stops the music streams. This allows the events to always
+ happen when MOH starts/stops. The event posting code was moved to
+ the MOH alloc/release routines. * Made channel_do_masquerade()
+ stop any MOH on the original channel before masquerading so the
+ original channel will get a stop event with correct information.
+ * Cleaned up a couple odd codings in moh_files_alloc() and
+ moh_alloc() dealing with the music state variable. (issue
+ ASTERISK-23311) Reported by: Benjamin Keith Ford Review:
+ https://reviewboard.asterisk.org/r/3306/
+
+ * apps/confbridge/conf_state.c,
+ apps/confbridge/conf_state_single.c,
+ apps/confbridge/conf_state_inactive.c,
+ apps/confbridge/conf_state_single_marked.c, /: app_confbridge:
+ Make explicitly stop MOH if a user is kicked or hangs up while
+ MOH is playing. When MOH is playing to a user in a conference and
+ the user is kicked or hangs up from the conference then the AMI
+ MusicOnHoldStop events didn't happen. (Asterisk v11 AMI event:
+ MusicOnHold, state:Stop) (closes issue ASTERISK-23311) Reported
+ by: Benjamin Keith Ford Review:
+ https://reviewboard.asterisk.org/r/3306/ ........ Merged
+ revisions 410490 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-12 12:50 +0000 [r410451-410471] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_multihomed.c: res_pjsip_multihomed: Fix a bug where
+ outgoing messages for TCP would go out using UDP. This change
+ fixes a bug where the code which changes the transport did not
+ check whether the message is going out over UDP or not before
+ changing it. For TCP and TLS transports we don't need to change
+ the transport as the correct one is already chosen.
+
+ * res/res_pjsip_multihomed.c (added): res_pjsip_multihomed: Add
+ module which places the correct address within messages. Due to
+ how messages are handled within PJSIP it is not until a message
+ is actually sent that the destination is reliably known. This
+ means that the addresses placed within the message may not be of
+ the interface the message is being sent out on. This module
+ determines what interface a message is being sent on and updates
+ the message to contain the correct address if applicable. This
+ module was tested by myself in a virtualized environment with
+ multiple interfaces and also by Kinsey Moore in the following
+ configuration: Networks: * 10.24.16.0/21 ** hard phone ** default
[... 28532 lines stripped ...]
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