[asterisk-commits] bebuild: tag 13.0.0-beta2 r423615 - in /tags/13.0.0-beta2: ./ contrib/realtim...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Sep 19 16:02:16 CDT 2014
Author: bebuild
Date: Fri Sep 19 16:02:13 2014
New Revision: 423615
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=423615
Log:
Importing files for 13.0.0-beta2 release.
Added:
tags/13.0.0-beta2/.lastclean (with props)
tags/13.0.0-beta2/.version (with props)
tags/13.0.0-beta2/ChangeLog (with props)
tags/13.0.0-beta2/contrib/realtime/mysql/mysql_cdr.sql (with props)
tags/13.0.0-beta2/contrib/realtime/mysql/mysql_config.sql (with props)
tags/13.0.0-beta2/contrib/realtime/mysql/mysql_voicemail.sql (with props)
tags/13.0.0-beta2/contrib/realtime/oracle/oracle_cdr.sql (with props)
tags/13.0.0-beta2/contrib/realtime/oracle/oracle_config.sql (with props)
tags/13.0.0-beta2/contrib/realtime/oracle/oracle_voicemail.sql (with props)
tags/13.0.0-beta2/contrib/realtime/postgresql/postgresql_cdr.sql (with props)
tags/13.0.0-beta2/contrib/realtime/postgresql/postgresql_config.sql (with props)
tags/13.0.0-beta2/contrib/realtime/postgresql/postgresql_voicemail.sql (with props)
tags/13.0.0-beta2/contrib/realtime/sqlserver/mssql_cdr.sql (with props)
tags/13.0.0-beta2/contrib/realtime/sqlserver/mssql_config.sql (with props)
tags/13.0.0-beta2/contrib/realtime/sqlserver/mssql_voicemail.sql (with props)
Added: tags/13.0.0-beta2/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/13.0.0-beta2/.lastclean?view=auto&rev=423615
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==============================================================================
--- tags/13.0.0-beta2/ChangeLog (added)
+++ tags/13.0.0-beta2/ChangeLog Fri Sep 19 16:02:13 2014
@@ -1,0 +1,17974 @@
+2014-09-19 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 13.0.0-beta2 Released.
+
+2014-09-19 19:51 +0000 [r423580] 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 ........ Merged revisions 423579 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-19 17:08 +0000 [r423561] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_pjsip.c, res/res_pjsip_sdp_rtp.c:
+ res_pjsip_sdp_rtp.c: Fix native formats containing formats that
+ were not negotiated. Outgoing PJSIP calls can result in
+ non-negotiated formats listed in the channel's native formats if
+ video formats are listed in the endpoint's configuration. The
+ resulting call could then use a non-negotiated format resulting
+ in one way audio. * Simplified the update of session->req_caps in
+ set_caps(). Why do something in five steps when only one is
+ needed? AFS-162 #close Review:
+ https://reviewboard.asterisk.org/r/4000/
+
+2014-09-19 15:18 +0000 [r423524-423530] 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/ ........
+ Merged revisions 423525 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * channels/chan_iax2.c: chan_iax2: Fix a crash when using chan_iax2
+ jitterbuffer settings Caused by format changes in Asterisk 13
+ ASTERISK-24265 #close Reported by: Dafi Ni Review:
+ https://reviewboard.asterisk.org/r/3999/
+
+2014-09-19 12:45 +0000 [r423504] Kinsey Moore <kmoore at digium.com>
+
+ * /, main/framehook.c, res/res_pjsip_t38.c,
+ include/asterisk/framehook.h: 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/ ........ Merged
+ revisions 423503 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 19:30 +0000 [r423482] 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 ........ Merged revisions
+ 423481 from http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 19:22 +0000 [r423478] 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/ ........ Merged
+ revisions 423476 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 18:31 +0000 [r423462] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_pjsip_pubsub.c: Add subscription state test events. These
+ are needed for a set of batched notification RLS tests that are
+ about to be committed to the testsuite. Review:
+ https://reviewboard.asterisk.org/r/3967
+
+2014-09-18 17:11 +0000 [r423425] 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/ ........ Merged
+ revisions 423417 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 17:09 +0000 [r423418-423423] Richard Mudgett <rmudgett at digium.com>
+
+ * bridges/bridge_softmix.c: bridge_softmix.c: Made use
+ ao2_replace() instead of the inline equivalent. * Clarified some
+ read/write format comments. * Fixed a doxygen tag typo.
+
+ * 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 ........ Merged
+ revisions 423416 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 16:37 +0000 [r423348-423414] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/format_cap.h, main/channel.c, main/format_cap.c,
+ main/translate.c: Add API call to determine if format capability
+ structure is "empty". Empty here means that there are no formats
+ in the format_cap structure or the only format in it is the
+ "none" format. I've added calls to check the emptiness of a
+ format_cap in a few places in order to short-circuit operations
+ that would otherwise be pointless as well as to prevent some
+ assertions from being triggered in cases where channels with no
+ formats are used.
+
+ * /, 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 ........ Merged
+ revisions 423365 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, 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/res_pjsip_exten_state.c, include/asterisk/res_pjsip_pubsub.h,
+ res/res_pjsip_pidf_body_generator.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 ........ Merged revisions
+ 423344 from http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 15:13 +0000 [r423284] George Joseph <george.joseph at fairview5.com>
+
+ * res/res_pjsip_endpoint_identifier_ip.c,
+ res/res_pjsip/pjsip_configuration.c,
+ res/res_pjsip/pjsip_options.c, res/res_pjsip/config_transport.c,
+ include/asterisk/res_pjsip.h, res/res_pjsip/config_auth.c, /,
+ res/res_pjsip/location.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/
+ ........ Merged revisions 423282 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 15:00 +0000 [r423281] David M. Lee <dlee at digium.com>
+
+ * Makefile, makeopts.in: Only install dahdi_span_config_hook if
+ DAHDI is enabled This patch changes the install to only install
+ the hook script if DAHDI is enabled. It also adds the script to
+ the uninstall task, and moves the DAHDI_UDEV_HOOK_DIR variable so
+ that it's not between the _MAKEOPTS variables and their comment.
+ This allows installs which specify a --prefix to work normally,
+ as long as they don't enable DAHDI. Review:
+ https://reviewboard.asterisk.org/r/3972/
+
+2014-09-18 14:45 +0000 [r423279] George Joseph <george.joseph at fairview5.com>
+
+ * include/asterisk/config.h, main/config.c, main/manager.c, /:
+ 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 ........ Merged
+ revisions 423278 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-17 18:05 +0000 [r423209-423255] 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 ........ Merged
+ revisions 423254 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * 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 ........ Merged
+ revisions 423211 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * 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 ........ Merged
+ revisions 423208 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-16 16:32 +0000 [r423192] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/file.c, apps/app_voicemail.c, include/asterisk/file.h:
+ Voicemail: get correct duration when copying file to vm Changes
+ made during format improvements resulted in the recording to
+ voicemail option 'm' of the MixMonitor app writing a zero length
+ duration in the msgXXXX.txt file. This change introduces a new
+ function ast_ratestream(), which provides the sample rate of the
+ format associated with the stream, and updates the app_voicemail
+ function for ast_app_copy_recording_to_vm to calculate the right
+ duration. Review: https://reviewboard.asterisk.org/r/3996/
+ ASTERISK-24328 #close
+
+2014-09-16 12:12 +0000 [r423152-423173] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_session.c, /: res_pjsip_session: Fix usage of wrong
+ memory pool when creating local SDP. ........ Merged revisions
+ 423172 from http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * include/asterisk/rtp_engine.h, 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 ........ Merged
+ revisions 423151 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-15 10:49 +0000 [r423069-423129] 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. ........ Merged revisions
+ 423128 from http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, configs/samples/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 ........ Merged
+ revisions 423068 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-12 16:09 +0000 [r422985] 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/ ........ Merged
+ revisions 422984 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-11 22:16 +0000 [r422965] 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. ........ Merged revisions 422964
+ from http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-10 16:04 +0000 [r422905] 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 ........ Merged
+ revisions 422904 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-10 15:59 +0000 [r422901] 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. ........ Merged revisions 422899 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-09 20:01 +0000 [r422883] 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 ........ Merged
+ revisions 422791 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-08 18:03 +0000 [r422851-422855] Mark Michelson <mmichelson at digium.com>
+
+ * configs/samples/pjsip.conf.sample: Add note about configuring
+ list_items on a single line.
+
+ * configs/samples/pjsip.conf.sample: Add sample configuration for
+ resource lists. On review /r/3977, it was recommended to note in
+ the sample configuration about the size limitation for resource
+ lists. However, since there was no section in the sample
+ configuration at all for resource list subscriptions, I decided
+ to make a separate commit where I have added the necessary sample
+ configuration as well as the size limitation warning.
+
+ * res/res_pjsip_pubsub.c: Pre-allocate transmission data buffer for
+ RLS NOTIFY requests. PJSIP, unless a constant is modified at
+ compilation time, limits SIP requests to 4000 bytes. Full-state
+ RLS notifications can easily exceed this limit with moderately
+ small lists. This changeset allows for Asterisk to work around
+ this size limit by performing its own allocation of the
+ transmission data buffer. This way, Asterisk can allocate a
+ buffer that exceeds the built-in maximum. We still impose our own
+ limit of 64000 bytes, mainly because making allocations larger
+ than that is a bit absurd. ASTERISK-24181 #close Reported by Mark
+ Michelson Review: https://reviewboard.asterisk.org/r/3977
+
+2014-09-08 15:41 +0000 [r422836] Jonathan Rose <jrose at digium.com>
+
+ * res/res_pjsip_pubsub.c: res_pjsip_pubsub: Check supported headers
+ for eventlist when subscribing to resource list
+ https://wiki.asterisk.org/wiki/display/AST/Resource+List+Subscription+Test+Plan
+ According to the off-nominal plan, if evenlist support is not
+ specified in a SUBSCRIBE's supported header(s), that subscription
+ should be rejected with an error. ASTERISK-23871 Reported by:
+ Mark Michelson Review:
+ https://reviewboard.asterisk.org/r/3960/diff/#index_header
+
+2014-09-06 22:49 +0000 [r422767-422770] 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. ........ Merged revisions 422769 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * 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. ........ Merged revisions
+ 422766 from http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-06 19:12 +0000 [r422747] 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. ........
+ Merged revisions 422746 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-05 22:03 +0000 [r422716-422719] Matthew Jordan <mjordan at digium.com>
+
+ * /, apps/app_macro.c, include/asterisk/channel.h,
+ apps/app_stack.c, main/cdr.c: 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
+ ........ Merged revisions 422718 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * 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 ........ Merged
+ revisions 422715 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-05 20:35 +0000 [r422700] Richard Mudgett <rmudgett at digium.com>
+
+ * funcs/func_channel.c: func_channel.c: Add missing locking to some
+ CHANNEL() requests. * The CHANNEL() audionativeformat,
+ videonativeformat, audioreadformat, and audiowriteformat now need
+ locking since the media format rework when accessing the
+ channel's format pointers. * Increased the buffer size for
+ CHANNEL() audionativeformat and videonativeformat output strings
+ since the allow=all can be a lengthy list. * Tweaked the
+ CHANNEL() XML documentation for secure_bridge_signaling,
+ secure_bridge_media, and state. * Ensured the output buffer is
+ initialized for secure_bridge_signaling and secure_bridge_media.
+ * Made use the locked_copy_string() macro instead of inlining it
+ for trace and checkhangup.
+
+2014-09-05 20:11 +0000 [r422665-422684] Jonathan Rose <jrose at digium.com>
+
+ * include/asterisk/dial.h, main/dial.c: Dial API: Add a dial option
+ to indicate the dialed channel will replace dialer Adds an option
+ to the dial API that marks an outgoing dial as replacing the
+ dialing channel for the purpose of propagating accountcode. When
+ it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of
+ AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on
+ the involved channels with ast_channel_req_accountcodes. Review:
+ https://reviewboard.asterisk.org/r/3968/
+
+ * 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/ ........ Merged
+ revisions 422664 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-05 17:36 +0000 [r422661] Richard Mudgett <rmudgett at digium.com>
+
+ * main/devicestate.c, channels/chan_iax2.c: devicestate.c: Minor
+ tweaks * In ast_state_chan2dev() use ARRAY_LEN() instead of a
+ sentinel value in chan2dev[]. * Fix some comments in chan_iax2.c.
+
+2014-09-05 13:28 +0000 [r422646] Kinsey Moore <kmoore at digium.com>
+
+ * menuselect/menuselect.c: Menuselect: Fix incorrect enabling on
+ failed deps This corrects a situation where menuselect can
+ incorrectly enable a module by default that has defaultenabled
+ set to "no" and has failed/non-selected dependencies. The bug is
+ due to an inverted test when checking for whether the given
+ module should be set to enabled by default on load. Review:
+ https://reviewboard.asterisk.org/r/3975/ Reported by: John
+ Bigelow
+
+2014-09-04 21:23 +0000 [r422631] Jonathan Rose <jrose at digium.com>
+
+ * 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 ........ Merged
+ revisions 422626 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-03 14:05 +0000 [r422558] 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 ........ Merged
+ revisions 422557 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-02 20:29 +0000 [r422542] 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 ........ Merged revisions
+ 422536 from http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-01 14:16 +0000 [r422504-422507] 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. ........
+ Merged revisions 422506 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * res/stasis/stasis_bridge.c, res/res_stasis.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
+ ........ Merged revisions 422503 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-08-30 17:32 +0000 [r422442-422445] 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/ ........ Merged
+ revisions 422444 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * main/manager.c, /: manager: Make WaitEvent action respect
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