[asterisk-commits] bebuild: tag 12.6.0-rc1 r423609 - in /tags/12.6.0-rc1: ./ contrib/realtime/my...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Sep 19 15:38:58 CDT 2014
Author: bebuild
Date: Fri Sep 19 15:38:55 2014
New Revision: 423609
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=423609
Log:
Importing files for 12.6.0-rc1 release.
Added:
tags/12.6.0-rc1/.lastclean (with props)
tags/12.6.0-rc1/.version (with props)
tags/12.6.0-rc1/ChangeLog (with props)
tags/12.6.0-rc1/contrib/realtime/mysql/mysql_cdr.sql (with props)
tags/12.6.0-rc1/contrib/realtime/mysql/mysql_config.sql (with props)
tags/12.6.0-rc1/contrib/realtime/mysql/mysql_voicemail.sql (with props)
tags/12.6.0-rc1/contrib/realtime/oracle/oracle_cdr.sql (with props)
tags/12.6.0-rc1/contrib/realtime/oracle/oracle_config.sql (with props)
tags/12.6.0-rc1/contrib/realtime/oracle/oracle_voicemail.sql (with props)
tags/12.6.0-rc1/contrib/realtime/postgresql/postgresql_cdr.sql (with props)
tags/12.6.0-rc1/contrib/realtime/postgresql/postgresql_config.sql (with props)
tags/12.6.0-rc1/contrib/realtime/postgresql/postgresql_voicemail.sql (with props)
tags/12.6.0-rc1/contrib/realtime/sqlserver/mssql_cdr.sql (with props)
tags/12.6.0-rc1/contrib/realtime/sqlserver/mssql_config.sql (with props)
tags/12.6.0-rc1/contrib/realtime/sqlserver/mssql_voicemail.sql (with props)
Added: tags/12.6.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/12.6.0-rc1/.lastclean?view=auto&rev=423609
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URL: http://svnview.digium.com/svn/asterisk/tags/12.6.0-rc1/ChangeLog?view=auto&rev=423609
==============================================================================
--- tags/12.6.0-rc1/ChangeLog (added)
+++ tags/12.6.0-rc1/ChangeLog Fri Sep 19 15:38:55 2014
@@ -1,0 +1,30657 @@
+2014-09-19 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 12.6.0-rc1 Released.
+
+2014-09-19 19:50 +0000 [r423579] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_notify.c: res_pjsip_notify: Fix crash on
+ unload/load and don't say the module doesn't exist on reload.
+ When unloading the module did not unregister the CLI commands
+ causing a crash upon load when they were registered again. When
+ reloading the module the return value from the config options
+ framework was not checked to determine if an error occurred or
+ not. This caused a message to be output saying the module did not
+ exist when reloading if no changes were present. AST-1433 #close
+ AST-1434 #close
+
+2014-09-19 15:10 +0000 [r423525] Jonathan Rose <jrose at digium.com>
+
+ * main/stasis_channels.c: Stasis_channels: Resolve unfinished Dials
+ when doing masquerades Masquerades into channels that are in the
+ dialing state don't end their dial and this goes against the
+ model for things like CDRs and generating Dial end manager
+ actions and such. ASTERISK-24237 #close Reported by: Richard
+ Mudgett Review: https://reviewboard.asterisk.org/r/3990/
+
+2014-09-19 12:30 +0000 [r423503] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_pjsip_t38.c, include/asterisk/framehook.h,
+ main/framehook.c: PJSIP: Prevent T38 framehook being put on wrong
+ channel This change gives framehooks a reverse-direction
+ masquerade callback in addition to chan_fixup_cb similar to the
+ callback added to datastores to handle the same situation. The
+ new callback provides the same parameters as the fixup callback,
+ but is called on the new channel's framehooks before moving
+ framehooks from the old channel to the new channel. This gives
+ the framehooks an oppurtunity to decide whether they should
+ remain on the new channel or be removed. This new callback is
+ used to prevent the PJSIP T.38 framehook from remaining on a
+ masqueraded channel if the new channel is not also a PJSIP
+ channel. This was causing a crash when a local channel was
+ masqueraded into a PJSIP channel and the framehook was executed
+ on the local channel since the channel's tech private data was
+ not structured as expected. Review:
+ https://reviewboard.asterisk.org/r/4001/
+
+2014-09-18 19:29 +0000 [r423481] Sean Bright <sean at malleable.com>
+
+ * res/res_pjsip/config_auth.c: res_pjsip: Don't require a password
+ when doing userpass authentication. An empty password is valid
+ for username/password authentication so we should allow password
+ to be empty/not supplied. Review:
+ https://reviewboard.asterisk.org/r/3988
+
+2014-09-18 19:21 +0000 [r423476] George Joseph <george.joseph at fairview5.com>
+
+ * main/utils.c, include/asterisk/strings.h, tests/test_strings.c:
+ utils: Create ast_strsep function that ignores separators inside
+ quotes This function acts like strsep with three exceptions... *
+ The separator is a single character instead of a string. *
+ Separators inside quotes are treated literally instead of like
+ separators. * You can elect to have leading and trailing
+ whitespace and quotes stripped from the result and have '\'
+ sequences unescaped. Like strsep, ast_strsep maintains no
+ internal state and you can call it recursively using different
+ separators on the same storage. Also like strsep, for consistent
+ results, consecutive separators are not collapsed so you may get
+ an empty string as a valid result. Tested by: George Joseph
+ Review: https://reviewboard.asterisk.org/r/3989/
+
+2014-09-18 16:44 +0000 [r423417] Jonathan Rose <jrose at digium.com>
+
+ * res/res_pjsip_endpoint_identifier_ip.c:
+ res_pjsip_endpoint_identifier_ip: Fix parsing of match value with
+ CIDR Also fixes comma separates match lists ASTERISK-24290 #close
+ Reported by: Ray Crumrine Review:
+ https://reviewboard.asterisk.org/r/3995/
+
+2014-09-18 16:39 +0000 [r423416] Richard Mudgett <rmudgett at digium.com>
+
+ * main/astobj2.c, contrib/scripts/refcounter.py, /:
+ astobj2.c/refcounter.py: Fix to deal with invalid object refs. *
+ Make astob2 REF_DEBUG output an invalid object line when an
+ invalid ao2 object ref/unref is attempted. This is similar to the
+ constructor/destructor lines. * Fixed refcounter.py to handle
+ skewed objects that have constructor/destructor states. * Made
+ refcounter.py highlight the invalid ao2 object refs by putting
+ them in their own section of the processed output file. * Made
+ refcounter.py highlight unreffing an object by more than one that
+ results in a negative ref count and the object being destroyed.
+ The abnormally destroyed object is reported in the invalid and
+ finalized object sections of the output. Review:
+ https://reviewboard.asterisk.org/r/3971/ ........ Merged
+ revisions 423349 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 423400 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-09-18 16:20 +0000 [r423344-423365] Mark Michelson <mmichelson at digium.com>
+
+ * /, res/res_fax_spandsp.c: res_fax_spandsp: Properly handle
+ cleanup before starting FAXes. If faxing fails at a very early
+ stage, then it is possible for us to pass a NULL t30 state
+ pointer to spandsp, which spandsp is none too pleased with. This
+ patch ensures that we pass the correct pointer to spandsp in the
+ situation where we have not yet set our local t30 state pointer.
+ ASTERISK-24301 #close Reported by Matt Jordan Patches:
+ ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License
+ #5049) ........ Merged revisions 423360 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_pjsip_exten_state.c, include/asterisk/res_pjsip_pubsub.h,
+ res/res_pjsip_pidf_body_generator.c, res/res_pjsip_mwi.c,
+ res/res_pjsip_dialog_info_body_generator.c,
+ res/res_pjsip_xpidf_body_generator.c,
+ res/res_pjsip_mwi_body_generator.c, res/res_pjsip_pubsub.c:
+ res_pjsip_pubsub: Add some type safety when generating NOTIFY
+ bodies. res_pjsip_pubsub has two separate checks that it makes
+ when a SUBSCRIBE arrives. * It checks that there is a
+ subscription handler for the Event * It checks that there are
+ body generators for the types in the Accept header The problem
+ is, there's nothing that ensures that these two things will
+ actually mesh with each other. For instance, Asterisk will accept
+ a subscription to MWI that accepts pidf+xml bodies. That doesn't
+ make sense. With this commit, we add some type information to the
+ mix. Subscription handlers state they generate data of type X,
+ and body generators state that they consume data of type X. This
+ way, Asterisk doesn't end up in some hilariously mismatched
+ situation like the one in the previous paragraph. ASTERISK-24136
+ #close Reported by Mark Michelson Review:
+ https://reviewboard.asterisk.org/r/3877 Review:
+ https://reviewboard.asterisk.org/r/3878
+
+2014-09-18 15:01 +0000 [r423278-423282] George Joseph <george.joseph at fairview5.com>
+
+ * res/res_pjsip/pjsip_configuration.c,
+ res/res_pjsip/config_transport.c, include/asterisk/res_pjsip.h,
+ res/res_pjsip/config_auth.c, res/res_pjsip/location.c,
+ res/res_pjsip_endpoint_identifier_ip.c: res_pjsip: ami: Fix error
+ in AMI output when an endpoint has no transport When no transport
+ is associated to an endpoint, the AMI output for
+ PJSIPShowEndpoint indicates an error instead of silently ignoring
+ the missing transport. This patch causes the error to appear only
+ if a transport was specified on the endpoint and the transport
+ doesn't exist. It also fixes an issue with counting the objects
+ that were actually found. ASTERISK-24161 #close ASTERISK-24331
+ #close Tested by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3998/
+
+ * main/config.c, main/manager.c, /, include/asterisk/config.h:
+ config: bug: Fix SEGV in ast_category_insert when matching
+ category isn't found If you call ast_category_insert with a match
+ category that doesn't exist, the list traverse runs out of 'next'
+ categories and you get a SEGV. This patch adds check for the
+ end-of-list condition and changes the signature to return an int
+ for success/failure indication instead of a void. The only
+ consumer of this function is manager and it was also changed to
+ use the return value. Tested by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3993/ ........ Merged
+ revisions 423276 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 423277 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-09-17 18:04 +0000 [r423151-423254] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Ensure that the
+ thread terminating pj stuff is registered. ........ Merged
+ revisions 423253 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix 100% CPU usage
+ due to timer heap thread spinning. Side note: I need a vacation.
+ ........ Merged revisions 423210 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Fix building when
+ pjproject is not used. ........ Merged revisions 423207 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_pjsip_session.c: res_pjsip_session: Fix usage of wrong
+ memory pool when creating local SDP.
+
+ * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Fix a myriad of TURN
+ client issues. 1. The number of file descriptors an ioqueue
+ instance can handle is fixed, so we now spawn the required number
+ to handle the load. 2. Our transport identifiers were exceeding
+ the range supported by pjnath. 3. The TURN client did not set up
+ client binding causing needless bandwidth usage. 4. The code no
+ longer updates address information on each packet. 5. STUN
+ traffic was getting looped back to Asterisk instead of going
+ through the TURN server. 6. Synchronization now ensures things
+ are completely setup or destroyed. 7. Logging now reflects the
+ target the TURN server is sending to/receiving from on our
+ behalf. ASTERISK-23577 #close Reported by: Jay Jideliov
+ ASTERISK-23634 #close Reported by: Roman Skvirsky Review:
+ https://reviewboard.asterisk.org/r/3982/ ........ Merged
+ revisions 423150 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-09-15 10:45 +0000 [r423068-423128] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * contrib/ast-db-manage/config/versions/5950038a6ead_fix_pjsip_verifiy_typo.py
+ (added): contrib: Fix verifyi typo in alembic DB script
+ ps_transport table. Reported by: Zogot (on IRC) Patches: tmp.diff
+ uploaded by Zogot, cleaned up by me.
+
+ * configs/sip.conf.sample, /: chan_sip: Clarify that sipdebug=yes
+ cannot be undone by the CLI. Document it in sip.conf.
+ ASTERISK-24249 #close Reported by: Avinash Mohod Review:
+ https://reviewboard.asterisk.org/r/3926/ ........ Merged
+ revisions 423066 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 423067 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-09-12 16:01 +0000 [r422984] Jonathan Rose <jrose at digium.com>
+
+ * main/config.c: Realtime: Fix a bug that caused realtime destroy
+ command to crash Also has could affect with anything that goes
+ through ast_destroy_realtime. If a CLI user used the command
+ 'realtime destroy <family>' with only a single column/value pair,
+ Asterisk would crash when trying to create a variable list from a
+ NULL value. ASTERISK-24231 #close Reported by: Niklas Larsson
+ Review: https://reviewboard.asterisk.org/r/3985/
+
+2014-09-11 22:16 +0000 [r422964] Mark Michelson <mmichelson at digium.com>
+
+ * main/app.c: Remove undocumented default behavior of
+ ast_play_and_record_full acceptdtmf. ast_play_and_record_full()
+ has a parameter called "acceptdtmf" that is a string of
+ acceptable DTMF digits that may be pressed by a caller to end and
+ accept the recording. ARI uses this function in order to perform
+ recording, and it provides options for what is passed as
+ acceptdtmf to ast_play_and_record_full(). By default, ARI passes
+ an empty string, with the intention that no DTMF can be used to
+ end the recording. The problem is that ast_play_and_record_full()
+ attempts to be "helpful" by setting "#" as the acceptdtmf if an
+ empty string or NULL pointer has been passed in. With ARI, this
+ results in unexpected behavior occurring if you have attempted to
+ intercept "#" yourself in order to perform some other
+ manipulation of the live recording. This change removes the
+ "helpful" behavior by no longer accepting "#" as a default
+ acceptdtmf if none is specified by the caller of
+ ast_play_and_record_full(). This makes the ARI scenario work as
+ expected. The other callers of ast_play_and_record_full() are
+ app_voicemail and app_minivm, and in both cases, they pass an
+ explicit "#" to ast_play_and_record_full() as acceptdtmf, so they
+ are unaffected by this change.
+
+2014-09-10 16:02 +0000 [r422904] George Joseph <george.joseph at fairview5.com>
+
+ * main/config.c, /: config: bug: fix truncation of included config
+ files on permissions error ast_config_text_file_save() currently
+ truncates include files as they are processed. If a subsequent
+ include file or the main config file has a permissions error that
+ prevents writing, earlier include files are left truncated
+ resulting in a frantic search for backups. This patch causes
+ ast_config_text_file_save to check for write access on all files
+ before it truncates any of them. Will be applied 1.8 > trunk.
+ Tested by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3986/ ........ Merged
+ revisions 422900 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 422903 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-09-10 15:58 +0000 [r422899] Sean Bright <sean at malleable.com>
+
+ * res/res_pjsip/config_auth.c: pjsip/config_auth.c: Add missing
+ whitespace to log messages. The errors generated when validating
+ 'auth' settings are missing a space which makes the messages a
+ little confusing.
+
+2014-09-07 00:09 +0000 [r422791] Rusty Newton <rnewton at digium.com>
+
+ * /, sounds/sounds.xml, sounds/Makefile: Sounds/BuildSystem:
+ Modifications to include new releases and Japanese language.
+ Modifying Makefile and sounds.xml to include new core 1.4.26 and
+ extra 1.4.15 sound prompt releases, plus the new Japanese core
+ sound prompts contributed by QLOOG. ASTERISK-23324 Reported by:
+ Kevin McCoy Tested by: Rusty Newton ........ Merged revisions
+ 422789 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 422790 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-09-06 22:48 +0000 [r422766-422769] Matthew Jordan <mjordan at digium.com>
+
+ * main/cdr.c: main/cdr: Copy over location information during a
+ fork When a CDR is forked, a new CDR is created and appended to
+ the CDR chain for the Party A. The forked CDR starts life off as
+ a clone of the last non-finalized for the particular Party A. In
+ the past, merely copying over the snapshots for Party A/Party B
+ would be sufficient. However, as the CDRs now contain cached
+ information from Party A - specifically application/data,
+ context, and extension - we need to copy that over during a fork
+ as well. Huzzah for unit tests catching this when the
+ context/extension were derived from a cached value on the CDR
+ instead of on Party A.
+
+ * main/rtp_engine.c: main/rtp_engine: Format NTP timestamps as
+ unsigned ints On some systems, a timeval's tv_sec/tv_usec will be
+ unsigned lont ints, as opposed to long ints. When the RTP engine
+ formats these as strings, it was previously formatting them as
+ signed integers, which can result in some odd negative timestamp
+ values (particularly on 32-bit systems). This patch formats the
+ values as unsigned long integers.
+
+2014-09-06 19:11 +0000 [r422746] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Fix retrieval of
+ "ice-pwd" attribute if in session and not media stream.
+
+2014-09-05 22:02 +0000 [r422715-422718] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_stack.c, main/cdr.c, apps/app_macro.c,
+ include/asterisk/channel.h: main/cdrs: Preserve context/extension
+ when executing a Macro or GoSub The context/extension in a CDR is
+ generally considered the destination of a call. When looking at a
+ 2-party call CDR, users will typically be presented with the
+ following: context exten channel dest_channel app data default
+ 1000 SIP/8675309 SIP/1000 Dial SIP/1000,,20 However, if the Dial
+ actually takes place in a Macro, the current behaviour in 12 will
+ result in the following CDR: context exten channel dest_channel
+ app data macro-dial s SIP/8675309 SIP/1000 Dial SIP/1000,,20 The
+ same is true of a GoSub: context exten channel dest_channel app
+ data subs dial_stuff SIP/8675309 SIP/1000 Dial SIP/1000,,20 This
+ generally makes the context/exten fields less than useful. It
+ isn't hard to preserve these values in the CDR state machine;
+ however, we need to have something that informs us when a channel
+ is executing a subroutine. Prior to this patch, there isn't
+ anything that does this. This patch solves this problem by adding
+ a new channel flag, AST_FLAG_SUBROUTINE_EXEC. This flag is set on
+ a channel when it executes a Macro or a GoSub. The CDR engine
+ looks for this value when updating a Party A snapshot; if the
+ flag is present, we don't override the context/exten on the main
+ CDR object. In a funny quirk, executing a hangup handler must
+ *not* abide by this logic, as the endbeforehexten logic assumes
+ that the user wants to see data that occurs in hangup logic,
+ which includes those subroutines. Since those execute outside of
+ a typical Dial operation (and will typically have their own
+ dedicated CDR anyway), this is unlikely to cause any heartburn.
+ Review: https://reviewboard.asterisk.org/r/3962/ ASTERISK-24254
+ #close Reported by: tm1000, Tony Lewis Tested by: Tony Lewis
+
+ * main/cdr.c: main/cdr: Fix crash/memory consumption in CDRs in
+ multi-party bridge scenarios This patch fixes an issue where CDRs
+ would get stuck generating an infinite number of CDRs, eventually
+ crashing Asterisk (and consuming a lot of memory along the way).
+ When a channel enters into a multi-party bridge, the CDR engine
+ creates mappings of each participant to each other participant,
+ picking the 'A' party as it goes. So, if we have four channels in
+ a multi-party bridge (Alice, Bob, Charlie, Denise), we would have
+ something like: Alice => Bob Alice => Charlie Alice => Denise Bob
+ => Charlie Bob => Denise Charlie => Denise This works fine when
+ participants enter the bridge a single time. When a participant
+ leaves a bridge, the CDRs for that channel are transitioned to a
+ finalized state. The bug occurs if Bob rejoins. When the CDR
+ engine creates mappings between the channels, it walks through
+ all the participants currently in the bridge, and realizes that
+ no one in the bridge can create a CDR with the channel (Bob). As
+ such it creates a new CDR for the candidate and appends it to
+ that candidate's chain. Unfortunately, on this particular code
+ path, it doesn't stop traversing the candidate's chain. Since we
+ just added ourselves to the chain, this causes the loop to keep
+ going, constantly adding new CDRs. This patch makes it so the
+ engine bails when it creates a CDR match in this case. Review:
+ https://reviewboard.asterisk.org/r/3964/ ASTERISK-24241 #close
+ Reported by: Deepak Singh Rawat Tested by: Deepak Singh Rawat
+ ASTERISK-24208 Reported by: Frankie Chin
+
+2014-09-05 17:46 +0000 [r422626-422664] Jonathan Rose <jrose at digium.com>
+
+ * main/cli.c: Call IDs: Fix appearance of call ID in core show
+ channels when NULL NULL call IDs were meant to appear as '(none)'
+ but instead were showing the contents of an uninitialized
+ character buffer. ASTERISK-24223 Review:
+ https://reviewboard.asterisk.org/r/3979/
+
+ * /, main/manager.c: Manager: Require read permission for SYSTEM in
+ order to send FullyBooted Review:
+ https://reviewboard.asterisk.org/r/3969/ ........ Merged
+ revisions 422584 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 422625 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-09-03 14:03 +0000 [r422557] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_transport_websocket.c:
+ res_pjsip_transport_websocket: Fix crash when the Contact header
+ is not a URI. The code for changing the Contact header wrongly
+ assumed that the Contact would always contain a URI. This is
+ incorrect. ASTERISK-24271 Reported by: Dafi Ni
+
+2014-09-02 18:16 +0000 [r422536] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_pjsip.c, res/res_pjsip_diversion.c,
+ res/res_pjsip_session.c, include/asterisk/res_pjsip_session.h:
+ Resolve race condition where channels enter dialplan application
+ before media has been negotiated. Testsuite tests will
+ occasionally fail because on reception of a 200 OK SIP response,
+ an AST_CONTROL_ANSWER frame is queued prior to when media has
+ finished being negotiated. This is because session supplements
+ are called into before PJSIP's inv_session code has told us that
+ media has been updated. Sometimes the queued answer frame is
+ handled by the PBX thread before the ensuing media negotiations
+ occur, causing a test failure. As it turns out, there is another
+ place that session supplements could be called into, which is
+ after media has finished getting negotiated. What this commit
+ introduces is a means for session supplements to indicate when
+ they wish to be called into when handling an incoming SIP
+ response. By default, all session supplements will be run at the
+ same point that they were prior to this commit. However, session
+ supplements may indicate that they wish to be handled earlier
+ than normal on redirects, or they may indicate they wish to be
+ handled after media has been negotiated. In this changeset, two
+ session supplements have been updated to indicate a preference
+ for when they should be run: res_pjsip_diversion executes before
+ handling redirection in order to get information from the
+ Diversion header, and chan_pjsip now handles responses to INVITEs
+ after media negotiation to fix the race condition mentioned
+ previously. ASTERISK-24212 #close Reported by Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/3930
+
+2014-09-01 14:16 +0000 [r422503-422506] Matthew Jordan <mjordan at digium.com>
+
+ * main/cli.c: main/cli: Do not attempt to show CDR data for
+ internal channels Internal channels don't have CDRs. Querying the
+ CDR engine for their variables will make it cranky.
+
+ * res/res_stasis.c, res/stasis/stasis_bridge.c: res_stasis: Don't
+ play MoH to channels by default when added to holding bridges
+ When ARI manipulates a bridge, it generally doesn't care what the
+ mixing technology is. Operations on a bridge initiated through
+ ARI should perform their action in generally the same way,
+ regardless of the bridge's mixing technology. While the mixing
+ technology may determine how media flows to channels, the actual
+ operations on a bridge themselves should be the same. Currently,
+ this isn't the case with holding bridges. When a channel joins
+ without a role, MoH is started on that channel automatically.
+ Subsequent bridge operations that would stop MoH would fail (as
+ there is no Announcer channel playing MoH to the bridge).
+ Starting MoH on the bridge will also create two MoH streams: one
+ from the MoH being played on the participant channel, and one
+ from the announcer channel. From the perspective of ARI users,
+ this is counter-intuitive - I would not expect MoH to be started
+ for me. The mixing technology determines how media is shared
+ between participants, not the application experience. This patch
+ does the following: * The Stasis bridge class now inspects
+ channels as they are going into a bridge. If the bridge has a
+ holding capability, and the channel has no roles, we give it a
+ participant role and mark the default behaviour to have no
+ entertainment. This allows addChannel operations to continue to
+ set a participant role with an entertainment option if it felt
+ like it (or could do it). * The music on hold channel is now
+ Stasis approved (tm) Review:
+ https://reviewboard.asterisk.org/r/3929/ ASTERISK-24264 #close
+ Reported by: Samuel Galarneau Tested by: Samuel Galarneau
+
+2014-08-30 17:28 +0000 [r422441-422444] George Joseph <george.joseph at fairview5.com>
+
+ * apps/app_confbridge.c: confbridge: Add Duration to ConfbridgeList
+ event The ConfbridgeList event doesn't include how long the user
+ has been a member of the conference. This patch adds Duration
+ (seconds) which is based on user->chan->answertime. Tested by:
+ George Joseph Review: https://reviewboard.asterisk.org/r/3955/
+
+ * main/manager.c, /: manager: Make WaitEvent action respect
+ eventfilters A WaitEvent issued via an http session isn't
+ respecting eventfilters defined for the user. I just added a
+ match_filter to the predicate that controls astman_append. Tested
+ by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3958/ ........ Merged
+ revisions 422439 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 422440 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-08-29 19:39 +0000 [r422295-422378] Matthew Jordan <mjordan at digium.com>
+
+ * doc/smsq.8 (added), /: doc: Add a manpage for the smsq utility
+ This patch adds a manpage for the smsq utility. Note that this is
+ one of the patches the Debian distro applies for the Asterisk
+ project, as per ASTERISK-24191. Review:
+ https://reviewboard.asterisk.org/r/3895/ ASTERISK-24171 #close
+ Reported by: Jeremy Laine patches: smsq.8 uploaded by Jeremy
+ Laine (License 6561) ........ Merged revisions 422376 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 422377 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, doc/aelparse.8 (added): doc: Add a manpage for the aelparse
+ utility This patch adds a manpage for the aelparse utility. Note
+ that this is one of the patches the Debian distro applies for the
+ Asterisk project, as per ASTERISK-24191. Review:
+ https://reviewboard.asterisk.org/r/3896/ ASTERISK-24171 #close
+ Reported by: Jeremy Laine patches: aelparse.8 uploaded by Jeremy
+ Laine (License 6561) ........ Merged revisions 422371 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 422372 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, LICENSE: LICENSE: Clarify language in Asterisk's LICENSE to
+ allow for linking to UniMRCP The UniMRCP project distributes
+ Asterisk modules that integrate Asterisk with UniMRCP, and other
+ Asterisk users use the UniMRCP library as well. Unfortunately,
+ the UniMRCP license is Apache 2.0, which per the Free Software
+ Foundation, is not a compatible license with the GPLv2. "Please
+ note that this license is not compatible with GPL version 2,
+ because it has some requirements that are not in that GPL
+ version. These include certain patent termination and
+ indemnification provisions. The patent termination provision is a
+ good thing, which is why we recommend the Apache 2.0 license for
+ substantial programs over other lax permissive licenses." On the
+ other hand, UniMRCP is a great project and we'd like to let
+ people use it with Asterisk. This patch updates the LICENSE text
+ to allow users to link Asterisk with UniMRCP and distribute the
+ resulting binaries. ........ Merged revisions 422293 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 422294 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-08-28 20:29 +0000 [r422275] Michael L. Young <elgueromexicano at gmail.com>
+
+ * channels/chan_iax2.c, /: chan_iax2: Fix Dynamic IAX2
+ Registrations After Temporary DNS Failure The reporter on the
+ issue found some issues when upgrading from version 10 to 11 on
+ 55 hosts. Two situations that can occur with dynamic
+ registrations. 1. With dnsmgr disabled, if the host is not
+ resolvable we are not trying to resolve the host again when it is
+ time to attempt to register again. This results in never
+ registering to the host. 2. With dnsmgr enabled, when the host is
+ temporarily not resolvable the address is set to 0.0.0.0:0 and
+ then when the host is resolvable the port is not being restored
+ and stays set to 0. This patch resolves these two issues by: *
+ Storing the hostname so that it can be used for resolving with
+ DNS. * Resolve the hostname on the next scheduled attempt to
+ register. * Storing the port used to reach the host so that when
+ the hostname is resolvable again, we can set the port again if
+ the port is still unset after looking up the host. ASTERISK-23767
+ #close Reported by: David Herselman Tested by: David Herselman,
+ Michael L. Young Patches:
+ asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/3856/ ........ Merged
+ revisions 422274 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-08-28 17:19 +0000 [r422214-422255] Richard Mudgett <rmudgett at digium.com>
+
+ * UPGRADE.txt: Added ConfBridge AMI event note to UPGRADE.txt.
+
+ * res/res_pjsip/pjsip_options.c: res/res_pjsip/pjsip_options.c:
+ Eliminate excessive RAII_VAR usage. * Fix off nominal ref leak in
+ find_or_create_contact_status(). * Add missing NULL check of
+ status in update_contact_status() and init_start_time().
+
+2014-08-27 17:21 +0000 [r422176] George Joseph <george.joseph at fairview5.com>
+
+ * apps/confbridge/confbridge_manager.c, apps/app_confbridge.c:
+ confbridge: Add 'Admin' param to join, leave, mute, unmute and
+ talking events Currently there's no way to tell if a user is an
+ admin or not when receiving the join, leave, mute, unmute and
+ talking events. This patch adds that capability. Tested by:
+ George Joseph Review: https://reviewboard.asterisk.org/r/3950/
+
+2014-08-27 15:14 +0000 [r422114] Kinsey Moore <kmoore at digium.com>
+
+ * main/callerid.c, main/utils.c, res/res_pjsip_caller_id.c,
+ include/asterisk/utils.h, /, channels/chan_sip.c,
+ tests/test_callerid.c (added), tests/test_utils.c: CallerID: Fix
+ parsing of malformed callerid This allows the callerid parsing
+ function to handle malformed input strings and strings containing
+ escaped and unescaped double quotes. This also adds a unittest to
+ cover many of the cases where the parsing algorithm previously
+ failed. Review: https://reviewboard.asterisk.org/r/3923/ ........
+ Merged revisions 422112 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 422113 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-08-26 23:18 +0000 [r422090] George Joseph <george.joseph at fairview5.com>
+
+ * apps/app_confbridge.c: confbridge: Make kick, mute and unmute
+ handle channel targets consistently. Kick, mute and unmute were a
+ little inconsistent in their handling of channel targets. This
+ patch cleans that up by insuring they all handle the 'all' target
+ consistently and adds the 'participants' target which acts on
+ non-admins. Documentation for kick was also cleaned up as it
+ never supported partial channel names. Tested by: George Joseph
+ Review: https://reviewboard.asterisk.org/r/3944/
+
+2014-08-26 22:08 +0000 [r422070] Mark Michelson <mmichelson at digium.com>
+
+ * main/sched.c: Fix race condition in the scheduler when deleting a
+ running entry. When scheduled tasks run, they are removed from
+ the heap (or hashtab). When a scheduled task is deleted, if the
+ task can't be found in the heap (or hashtab), an assertion is
+ triggered. If DO_CRASH is enabled, this assertion causes a crash.
+ The problem is, sometimes it just so happens that someone
+ attempts to delete a scheduled task at the time that it is
+ running, leading to a crash. This change corrects the issue by
+ tracking which task is currently running. If that task is
+ attempted to be deleted, then we mark the task, and then wait for
+ the task to complete. This way, we can be sure to coordinate task
+ deletion and memory freeing. ASTERISK-24212 Reported by Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/3927
+
+2014-08-25 16:11 +0000 [r421978] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_musiconhold.c, /: res_musiconhold: Fix MOH restarting
+ where it left off from the last hold. Restore code removed by
+ https://reviewboard.asterisk.org/r/3536/ that introduced a
+ regression that prevents MOH from restarting were it left off the
+ last time. ASTERISK-24019 #close Reported by: Jason Richards
+ Patches: jira_asterisk_24019_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Review:
+ https://reviewboard.asterisk.org/r/3928/ ........ Merged
+ revisions 421976 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 421977 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-08-24 19:34 +0000 [r421910-421955] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_transport_websocket.c:
+ res_pjsip_transport_websocket: Attach the Websocket module on
+ outgoing INVITEs. In order to alter the Contact header on
+ in-dialog requests and responses the Websocket module must be
+ attached on outgoing INVITEs. The Contact header is modified so
+ that the PJSIP transport layer can find and use the existing
+ Websocket connection based on the source IP address, port, and
+ transport. ASTERISK-24143 #close Reported by: Aleksei Kulakov
+
+ * res/res_pjsip_transport_websocket.c:
+ res_pjsip_transport_websocket: Fix a progressive memory growth.
+ The packet structure used to receive messages was using the
+ transport pool. This meant that for each parsing the pool would
+ grow accordingly. Since memory can not be reclaimed without
+ resetting it this would cause the memory pool to grow and grow.
+ This change uses a specific memory pool for the packet structure
+ and resets it to a fresh state after the message has been
+ received and handled.
+
+ * res/res_pjsip_transport_websocket.c:
+ res_pjsip_transport_websocket: Ensure secure Websocket clients
+ can be called. This change enforces the transport in the Contact
+ header for Websocket clients. Previously a client may provide a
+ transport of 'ws' when it is actually using a transport of 'wss'.
+ This would cause outgoing calls to fail as the existing
+ connection could not be found.
+
+ * /, channels/chan_sip.c: chan_sip: Use the server reflexive ICE
+ candidate RTCP port as provided. This code originally worked
+ around an issue within res_rtp_asterisk itself. The wrong socket
+ was being used for the STUN check for RTCP, causing the port to
+ be the same as RTP. This was subsequently fixed and the RTCP port
+ provided for the ICE candidate is correct and does not need to be
+ incremented. ASTERISK-23997 #close Reported by: Badalian
+ Vyacheslav Patches: plus1.diff submitted by Badalian Vyacheslav
+ (license 5249) ........ Merged revisions 421909 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-08-22 16:27 +0000 [r421879] Jonathan Rose <jrose at digium.com>
+
+ * res/res_stasis.c, res/stasis/command.c,
+ res/res_stasis_playback.c, res/stasis/control.c,
+ res/stasis/stasis_bridge.c, res/stasis/command.h,
+ include/asterisk/stasis_app_impl.h, res/res_stasis_recording.c,
+ res/res_stasis_answer.c: ARI: Fix a crash caused by hanging
+ during playback to a channel in a bridge ASTERISK-24147 #close
+ Reported by: Edvin Vidmar Review:
+ https://reviewboard.asterisk.org/r/3908/
+
+2014-08-22 13:50 +0000 [r421859] Matthew Jordan <mjordan at digium.com>
+
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