[svn-commits] bebuild: tag 13.3.0-rc1 r433312 - in /tags/13.3.0-rc1: ./ contrib/realtime/my...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Mar 23 11:52:58 CDT 2015
Author: bebuild
Date: Mon Mar 23 11:52:56 2015
New Revision: 433312
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=433312
Log:
Importing files for 13.3.0-rc1 release.
Added:
tags/13.3.0-rc1/.lastclean (with props)
tags/13.3.0-rc1/.version (with props)
tags/13.3.0-rc1/ChangeLog (with props)
tags/13.3.0-rc1/contrib/realtime/mysql/mysql_cdr.sql (with props)
tags/13.3.0-rc1/contrib/realtime/mysql/mysql_config.sql (with props)
tags/13.3.0-rc1/contrib/realtime/mysql/mysql_voicemail.sql (with props)
tags/13.3.0-rc1/contrib/realtime/oracle/oracle_cdr.sql (with props)
tags/13.3.0-rc1/contrib/realtime/oracle/oracle_config.sql (with props)
tags/13.3.0-rc1/contrib/realtime/oracle/oracle_voicemail.sql (with props)
tags/13.3.0-rc1/contrib/realtime/postgresql/postgresql_cdr.sql (with props)
tags/13.3.0-rc1/contrib/realtime/postgresql/postgresql_config.sql (with props)
tags/13.3.0-rc1/contrib/realtime/postgresql/postgresql_voicemail.sql (with props)
tags/13.3.0-rc1/contrib/realtime/sqlserver/mssql_cdr.sql (with props)
tags/13.3.0-rc1/contrib/realtime/sqlserver/mssql_config.sql (with props)
tags/13.3.0-rc1/contrib/realtime/sqlserver/mssql_voicemail.sql (with props)
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==============================================================================
--- tags/13.3.0-rc1/ChangeLog (added)
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+2015-03-23 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 13.3.0-rc1 Released.
+
+2015-03-22 23:58 +0000 [r433247-433269] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_queue.c, main/cli.c, main/cdr.c, main/manager.c,
+ main/rtp_engine.c, /, funcs/func_cdr.c: Fix compilations errors
+ on 64-bit OpenBSD systems In versiong 5.5, OpenBSD went to 64-bit
+ time values. This requires a cast to (long) when printing members
+ of certain time structs. Review:
+ https://reviewboard.asterisk.org/r/4507 ASTERISK-24879 #close
+ Reported by: snuffy Tested by: snuffy patches:
+ openbsd-time64.diff uploaded by snuffy (License 5024) ........
+ Merged revisions 433268 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/asterisk.c, main/loader.c, main/xmldoc.c, /: Fix compilation
+ issues for OpenBSD This patch addresses compilation issues for
+ OpenBSD. Specifically, it addresses: * It allows including
+ <sys/vmmeter.h> in asterisk.c * Provides a needed (size_t) cast
+ in xmldoc.c In 13+, it also addresses a conditional inclusion in
+ loader.c. Review: https://reviewboard.asterisk.org/r/4506
+ ASTERISK-24880 #close Reported by: snuffy Tested by: snuffy
+ patches: misc-openbsd.diff uploaded by snuffy (License 5024)
+ ........ Merged revisions 433245 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-03-20 19:52 +0000 [r433199-433222] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_pjsip_messaging.c, res/res_pjsip/pjsip_options.c,
+ res/res_pjsip.c, res/res_pjsip_nat.c: Audit
+ ast_pjsip_rdata_get_endpoint() usage for ref leaks. Valgrind
+ found some memory leaks associated with
+ ast_pjsip_rdata_get_endpoint(). The leaks would manifest when
+ sending responses to OPTIONS requests, processing MESSAGE
+ requests, and res_pjsip supplements implementing the
+ incoming_request callback. * Fix ast_pjsip_rdata_get_endpoint()
+ endpoint ref leaks in res/res_pjsip.c:supplement_on_rx_request(),
+ res/res_pjsip/pjsip_options.c:send_options_response(),
+ res/res_pjsip_messaging.c:rx_data_to_ast_msg(), and
+ res/res_pjsip_messaging.c:send_response(). * Eliminated
+ RAII_VAR() use with ast_pjsip_rdata_get_endpoint() in
+ res/res_pjsip_nat.c:nat_on_rx_message(). * Fixed inconsistent but
+ benign return value in
+ res/res_pjsip/pjsip_options.c:options_on_rx_request(). Review:
+ https://reviewboard.asterisk.org/r/4511/
+
+ * res/res_pjsip_sdp_rtp.c, main/sorcery.c, main/xmldoc.c:
+ res_pjsip_sdp_rtp,sorcery: Fix invalid access and memory leak
+ respectively. Valgrind found a memory leak and invalid access. *
+ Fix invalid access by sscanf() being fed a non-nul terminated
+ string of digits in res/res_pjsip_sdp_rtp.c:get_codecs(). * Fix
+ memory leak in main/sorcery.c:sorcery_object_field_destructor().
+ * Fix potential NULL pointer dereference in
+ main/xmldoc.c:xmldoc_get_syntax_config_option(). Review:
+ https://reviewboard.asterisk.org/r/4513/
+
+2015-03-19 19:19 +0000 [r433174] Matthew Jordan <mjordan at digium.com>
+
+ * funcs/func_env.c, tests/test_func_file.c, /: funcs/func_env: Fix
+ regression caused in FILE read operation When r432935 was merged,
+ it did correctly fix a situation where a FILE read operation on
+ the middle of a file buffer would not read the requested length
+ in the parameters passed to the FILE function. Unfortunately, it
+ would also allow the FILE function to append more bytes than what
+ was available in the buffer if the length exceeded the end of the
+ buffer length. This patch takes the minimum of the remaining
+ bytes in the buffer along with the calculated length to append
+ provided by the original patch, and uses that as the length to
+ append in the return result. This patch also updates the unit
+ tests with the scenarios that were originally pointed out in
+ ASTERISK-21765 that the original implementation treated
+ incorrectly. ASTERISK-21765 ........ Merged revisions 433173 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-03-19 10:20 +0000 [r433113-433126] Corey Farrell <git at cfware.com>
+
+ * main/logger.c, /: logger: Apply default console logging when
+ configuration cannot be loaded. When logger.conf is missing or
+ invalid enable console logging and display an error message.
+ ASTERISK-24817 #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4497/ ........ Merged
+ revisions 433122 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/sip/include/dialog.h, channels/chan_sip.c,
+ channels/sip/include/sip.h: chan_sip: Simplify dialog/peer
+ references, improve REF_DEBUG output. * Replace functions for
+ ref/undef of dialogs and peers with macro's to call
+ ao2_t_bump/ao2_t_cleanup. * Enable passthough of REF_DEBUG caller
+ information to sip_alloc and find_call. ASTERISK-24882 #close
+ Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4189/
+
+ * /, channels/chan_sip.c: chan_sip: Fix dialog reference leaked to
+ scheduler for reinvite_timeout. Release the scheduler reference
+ to the dialog for reinvite timeout during dialog_unlink_all.
+ ASTERISK-24876 #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4491/ ........ Merged
+ revisions 433112 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-03-18 02:34 +0000 [r433088] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_pjsip_session.c: res_pjsip_session: Fix off-nominal extra
+ unref of session.
+
+2015-03-17 22:15 +0000 [r433060-433064] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/asterisk.c, main/config.c, main/xmldoc.c, main/manager.c,
+ include/asterisk/config.h, main/utils.c, main/codec_builtin.c,
+ main/endpoints.c: Various: bugfixes found via chaos Using
+ DEBUG_CHAOS several instances of a null pointer crash, and one
+ uninitialized variable were uncovered and fixed. Also added
+ details on why Asterisk failed to initialize. Review:
+ https://reviewboard.asterisk.org/r/4468/
+
+ * build_tools/cflags.xml, include/asterisk/utils.h: core: Introduce
+ chaos into memory allocations Locate potential crashes by
+ exercising seldom used code paths. This patch introduces a new
+ define DEBUG_CHAOS, and mechanism to randomly return an error
+ condition from functions that will seldom do so. Functions that
+ handle the allocation of memory get the first treatment. Review:
+ https://reviewboard.asterisk.org/r/4463/
+
+2015-03-17 21:49 +0000 [r433057] Richard Mudgett <rmudgett at digium.com>
+
+ * main/netsock2.c, /, res/res_pjsip_sdp_rtp.c, res/res_pjsip_t38.c,
+ apps/app_externalivr.c, res/res_pjsip_acl.c: Audit
+ ast_sockaddr_resolve() usage for memory leaks. Valgrind found
+ some memory leaks associated with ast_sockaddr_resolve(). Most of
+ the leaks had already been fixed by earlier memory leak hunt
+ patches. This patch performs an audit of ast_sockaddr_resolve()
+ and found one more. * Fix ast_sockaddr_resolve() memory leak in
+ apps/app_externalivr.c:app_exec(). * Made
+ main/netsock2.c:ast_sockaddr_resolve() always set the addrs
+ parameter for safety so the pointer will never be uninitialized
+ on return. The same goes for
+ res/res_pjsip_acl.c:extract_contact_addr(). * Made functions that
+ call ast_sockaddr_resolve() with RAII_VAR() controlling the addrs
+ variable use ast_free instead of ast_free_ptr to provide better
+ MALLOC_DEBUG information. Review:
+ https://reviewboard.asterisk.org/r/4509/ ........ Merged
+ revisions 433056 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-03-17 18:34 +0000 [r433028-433031] Kevin Harwell <kharwell at digium.com>
+
+ * include/asterisk/res_pjsip.h,
+ res/res_pjsip_endpoint_identifier_anonymous.c,
+ res/res_pjsip_endpoint_identifier_ip.c, res/res_pjsip.c,
+ res/res_pjsip_endpoint_identifier_user.c: res_pjsip: Allow
+ configuration of endpoint identifier query order Updated some
+ documentation stating that endpoint identifiers registered
+ without a name are place at the front of the lookup list. Also
+ renamed register method
+ 'ast_sip_register_endpoint_identifier_by_name' to
+ 'ast_sip_register_endpoint_identifier_with_name' ASTERISK-24840
+ Reported by: Mark Michelson
+
+ * configs/samples/pjsip.conf.sample, CHANGES, res/res_pjsip.c,
+ res/res_pjsip_endpoint_identifier_user.c,
+ include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
+ res/res_pjsip_endpoint_identifier_anonymous.c,
+ res/res_pjsip_endpoint_identifier_ip.c,
+ contrib/ast-db-manage/config/versions/45e3f47c6c44_add_pjsip_endpoint_identifier_order.py:
+ res_pjsip: Allow configuration of endpoint identifier query order
+ This patch fixes previously reverted code that caused binary
+ incompatibility problems with some modules. And like the original
+ patch it makes sure that no matter what order the endpoint
+ identifier modules were loaded, priority is given based on the
+ ones specified in the new global 'endpoint_identifier_order'
+ option. ASTERISK-24840 Reported by: Mark Michelson Review:
+ https://reviewboard.asterisk.org/r/4489/
+
+2015-03-17 16:10 +0000 [r433005] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_pjsip.c: res_pjsip: Add reason comment.
+
+2015-03-14 02:28 +0000 [r432971] Matthew Jordan <mjordan at digium.com>
+
+ * /, main/format_cap.c: main/frame: Don't report empty disallow
+ values as an error In realtime, it is normal to have a database
+ with both 'allow' and 'disallow' columns in the schema. It is
+ perfectly valid to have an 'allow' value of '!all,g722,ulaw,alaw'
+ and no 'disallow' value. Unlike in static conf files, you can't
+ *not* provide the disallow value. Thus, the empty disallow value
+ causes a spurious WARNING message, which is kind of annoying.
+ This patch makes it so that a 'disallow' value with no ... value
+ ... is ignored. Granted, you can still screw this up as well, as
+ technically specifying 'disallow=all,!ulaw' allows only ulaw, and
+ then you would have no 'allow' value in your database. But
+ really, why would you do that? WHY? ASTERISK-16779 #close
+ Reported by: Atis Lezdins ........ Merged revisions 432970 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-03-14 02:00 +0000 [r432945-432949] Joshua Colp <jcolp at digium.com>
+
+ * funcs/func_curl.c, /: func_curl: Don't hold exclusive lock when
+ performing HTTP request. This code originally kept a lock held
+ when performing the HTTP request to ensure that the options
+ provided to curl remain valid. This doesn't seem to be necessary
+ these days and holding the lock caused requests to happen
+ sequentially instead of in parallel. ASTERISK-18708 #close
+ Reported by: Dave Cabot ........ Merged revisions 432948 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/cli.c, /: core: Fix tab completion of "core set debug
+ channel" CLI command. The "core set debug channel" CLI command
+ mistakenly had source filenames added to its tab completion. This
+ occurred because the CLI generator fell back to the "core set
+ debug" command which permits setting debug at a source filename
+ level. ASTERISK-21038 #close Reported by: Richard Kenner ........
+ Merged revisions 432944 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-03-14 01:21 +0000 [r432920-432938] Matthew Jordan <mjordan at digium.com>
+
+ * /, funcs/func_env.c: FILE: fix retrieval of file contents when
+ offset is specified The loop that reads in a file was not
+ correctly using the offset when determining what bytes to append
+ to the output. This patch corrects the logic such that the
+ correct portion of the file is extracted when an offset is
+ specified. ASTERISK-21765 Reported by: John Zhong Tested by: Matt
+ Jordan, Di-Shi Sun patches: file_read_390821.patch uploaded by
+ Di-Shi Sun (License 5076) ........ Merged revisions 432935 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/app_amd.c, /, configs/samples/amd.conf.sample: apps/app_amd:
+ Document maximum_word_length option; fix AMDCAUSE documentation
+ This patch corrects the documentation for the AMD application.
+ Specifically: * It documents the maximum_word_length option,
+ which limits the maximum allowed length of a single utterance. *
+ It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH.
+ MAXWORDLENGTH was documented as MAXWORDS, while MAXWORDS was
+ undocumented. Thanks to the issue reporter, Frank DiGennaro, for
+ pointing out the issues. ASTERISK-19470 #close Reported by: Frank
+ DiGennaro ........ Merged revisions 432918 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-03-13 17:04 +0000 [r432892-432894] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_pjsip/pjsip_configuration.c: chan_pjsip: AMI action
+ PJSIPShowEndpoint closes AMI connection on error. Also fixed
+ similar problem with AMI action PJSIPShowEndpoints.
+ ASTERISK-24872 #close Reported by: Dmitriy Serov Review:
+ https://reviewboard.asterisk.org/r/4487/
+
+ * channels/chan_pjsip.c, res/res_pjsip_caller_id.c:
+ chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and
+ consistent. The res_pjsip modules were manually checking both
+ name and number presentation values when there is a function that
+ determines the combined presentation for a party ID struct. The
+ function takes into account if the name or number components are
+ valid while the manual code rarely checked if the data was even
+ valid. * Made use ast_party_id_presentation() rather than
+ manually checking party ID presentation values. * Ensure that
+ set_id_from_pai() and set_id_from_rpid() will not return
+ presentation values other than what is pulled out of the SIP
+ headers. It is best if the code doesn't assume that
+ AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. *
+ Fixed copy paste error in add_privacy_params() dealing with RPID
+ privacy. * Pulled the id->number.valid test from
+ add_privacy_header() and add_privacy_params() up into the parent
+ function add_id_headers() to skip adding PAI/RPID headers
+ earlier. * Made update_connected_line_information() not send out
+ connected line updates if the connected line number is invalid.
+ Lower level code would not add the party ID information and thus
+ the sent message would be unnecessary. * Eliminated RAII_VAR
+ usage in send_direct_media_request(). Review:
+ https://reviewboard.asterisk.org/r/4472/
+
+2015-03-13 14:48 +0000 [r432868] Kevin Harwell <kharwell at digium.com>
+
+ * include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
+ res/res_pjsip_endpoint_identifier_anonymous.c,
+ res/res_pjsip_endpoint_identifier_ip.c,
+ contrib/ast-db-manage/config/versions/45e3f47c6c44_add_pjsip_endpoint_identifier_order.py,
+ configs/samples/pjsip.conf.sample, CHANGES, res/res_pjsip.c,
+ res/res_pjsip_endpoint_identifier_user.c: Revert - res_pjsip:
+ Allow configuration of endpoint identifier query order Due to a
+ break in binary compatibility with some other modules these
+ changes are being reverted until the issue can be resolved.
+ ASTERISK-24840 Reported by: Mark Michelson
+
+2015-03-12 12:58 +0000 [r432808-432811] Matthew Jordan <mjordan at digium.com>
+
+ * /, main/audiohook.c: main/audiohook: Update internal sample rate
+ on reads When an audiohook is created (which is used by the
+ various Spy applications and Snoop channel in Asterisk 13+), it
+ initially is given a sample rate of 8kHz. It is expected,
+ however, that this rate may change based on the media that passes
+ through the audiohook. However, the read/write operations on the
+ audiohook behave very differently. When a frame is written to the
+ audiohook, the format of the frame is checked against the
+ internal sample rate. If the rate of the format does not match
+ the internal sample rate, the internal sample rate is updated and
+ a new SLIN format is chosen based on that sample rate. This works
+ just fine. When a frame is read, however, we do something quite
+ different. If the format rate matches the internal sample rate,
+ all is fine. However, if the rates don't match, the audiohook
+ attempts to "fix up" the number of samples that were requested.
+ This can result in some seriously large number of samples being
+ requested from the read/write factories. Consider the worst case
+ - 192kHz SLIN. If we attempt to read 20ms worth of audio produced
+ at that rate, we'd request 3840 samples (192000 / (1000 / 20)).
+ However, if the audiohook is still expecting an internal sample
+ rate of 8000, we'll attempt to "fix up" the requested samples to:
+ samples_converted = samples * (ast_format_get_sample_rate(format)
+ / (float) audiohook->hook_internal_samp_rate); which is: 92160 =
+ 3840 * (192000 / 8000) This results in us attempting to read
+ 92160 samples from our factories, as opposed to the 3840 that we
+ actually wanted. On a 64-bit machine, this miraculously survives
+ - despite allocating up to two buffers of length 92160 on the
+ stack. The 32-bit machines aren't quite so lucky. Even in the
+ case where this works, we will either (a) get way more samples
+ than we wanted; or (b) get about 3840 samples, assuming the
+ timing is pretty good on the machine. Either way, the calculation
+ being performed is wrong, based on the API users expectations. My
+ first inclination was to allocate the buffers on the heap. As it
+ is, however, there's at least two drawbacks with doing this: (1)
+ It's a bit complicated, as the size of the buffers may change
+ during the lifetime of the audiohook (ew). (2) The stack is
+ faster (yay); the heap is slower (boo). Since our calculation is
+ flat out wrong in the first place, this patch fixes this issue by
+ instead updating the internal sample rate based on the format
+ passed into the read operation. This causes us to read the
+ correct number of samples, and has the added benefit of setting
+ the audihook with the right SLIN format. Note that this issue was
+ caught by the Asterisk Test Suite as a result of r432195 in the
+ 13 branch. Because this issue is also theoretically possible in
+ Asterisk 11, the change is being made here as well. Review:
+ https://reviewboard.asterisk.org/r/4475/ ........ Merged
+ revisions 432810 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * Makefile, include/asterisk/utils.h, /, configure, main/Makefile,
+ configure.ac, include/asterisk/inline_api.h, makeopts.in: Add
+ support for the clang compiler; update RAII_VAR to use
+ BlocksRuntime RAII_VAR, which is used extensively in Asterisk to
+ manage reference counted resources, uses a GCC extension to
+ automatically invoke a cleanup function when a variable loses
+ scope. While this functionality is incredibly useful and has
+ prevented a large number of memory leaks, it also prevents
+ Asterisk from being compiled with clang. This patch updates the
+ RAII_VAR macro such that it can be compiled with clang. It makes
+ use of the BlocksRuntime, which allows for a closure to be
+ created that performs the actual cleanup. Note that this does not
+ attempt to address the numerous warnings that the clang compiler
+ catches in Asterisk. Much thanks for this patch goes to: * The
+ folks on StackOverflow who asked this question and Leushenko for
+ providing the answer that formed the basis of this code:
+ http://stackoverflow.com/questions/24959440/rewrite-gcc-cleanup-macro-with-nested-function-for-clang
+ * Diederik de Groot, who has been extremely patient in working on
+ getting this patch into Asterisk. Review:
+ https://reviewboard.asterisk.org/r/4370/ ASTERISK-24133
+ ASTERISK-23666 ASTERISK-20399 ASTERISK-20850 #close Reported by:
+ Diederik de Groot patches: RAII_CLANG.patch uploaded by Diederik
+ de Groot (License 6600) ........ Merged revisions 432807 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-03-11 16:38 +0000 [r432764-432787] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_pjsip/config_domain_aliases.c,
+ res/res_pjsip/include/res_pjsip_private.h,
+ include/asterisk/res_pjsip.h: res_pjsip: Move internal
+ init/destroy prototypes to private header file. Done as a
+ separate commit from a finding in
+ https://reviewboard.asterisk.org/r/4467/
+
+ * include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
+ res/res_pjsip/pjsip_configuration.c: res_pjsip: Fix pjsip.conf
+ type=global object default value handling. When a type=global
+ section is not defined in pjsip.conf the global defaults are not
+ applied. As a result the mandatory Max-Forwards header is not
+ added to SIP messages for res_pjsip/chan_pjsip. The handling of
+ pjsip.conf type=global objects has several problems: 1) If the
+ global object is missing the defaults are not applied. 2) If the
+ global object is missing the default_outbound_endpoint's default
+ value is not returned by
+ ast_sip_global_default_outbound_endpoint(). 3) Defines are needed
+ so default values only need to be changed in one place. * Added a
+ sorcery instance observer callback to check if there were any
+ type=global sections loaded. If there were more than one then
+ issue an error message. If there were none then apply the global
+ defaults. * Fixed ast_sip_global_default_outbound_endpoint() to
+ return the documented default when no type=global object is
+ defined. * Made defines for the global default values. *
+ Increased the default_useragent[] size because SVN version
+ strings can get lengthy and 128 characters may not be enough. *
+ Fixed an off-nominal code path ref leak in global_alloc() if the
+ string fields fail to initialize. * Eliminated RAII_VAR in
+ get_global_cfg() and ast_sip_global_default_outbound_endpoint().
+ ASTERISK-24807 #close Reported by: Anatoli Review:
+ https://reviewboard.asterisk.org/r/4467/
+
+ * res/res_pjsip/pjsip_global_headers.c: res_pjsip: Fixed invalid
+ empty Server and User-Agent SIP headers. Setting pjsip.conf
+ useragent to an empty string results in an empty SIP header being
+ sent. * Made not add an empty SIP header item to the global SIP
+ headers list. Review: https://reviewboard.asterisk.org/r/4467/
+
+2015-03-10 23:09 +0000 [r432742] Joshua Colp <jcolp at digium.com>
+
+ * main/stasis_channels.c, main/endpoints.c, main/stasis_bridges.c:
+ core: Don't create snapshots with locks. Snapshots are immutable
+ and are never changed. Allocating them with a lock is wasteful.
+ Review: https://reviewboard.asterisk.org/r/4469/
+
+2015-03-10 21:33 +0000 [r432693-432721] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_config_odbc.c, /: res/res_config_odbc: Fix improper
+ escaping of backslashes with MySQL When escaping backslashes with
+ MySQL, the proper way to escape the characters in a LIKE clause
+ is to escape the '\' four times, i.e., '\\\\'. To quote the MySQL
+ manual: "Because MySQL uses C escape syntax in strings (for
+ example, “\n†to represent a newline character), you must double
+ any “\†that you use in LIKE strings. For example, to search for
+ “\nâ€, specify it as “\\nâ€. To search for “\â€, specify it as
+ “\\\\â€; this is because the backslashes are stripped once by the
+ parser and again when the pattern match is made, leaving a single
+ backslash to be matched against." ASTERISK-24808 #close Reported
+ by: Javier Acosta patches: res_config_odbc.diff uploaded by
+ Javier Acosta (License 6690) ........ Merged revisions 432720
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/app_voicemail.c, /: app_voicemail: Fix crash with IMAP
+ backends when greetings aren't present When an IMAP backend is in
+ use and greetings are set to be used, but aren't present for a
+ user in their IMAP folder, Asterisk will crash. This occurs due
+ to the mailstream being set to the 'greetings' folder and being
+ left in that particular state, regardless of the success/failure
+ of the attempt to access the folder the mailstream points to.
+ Later access of the mailstream assumes that it points to the
+ 'INBOX' (or some other folder), resulting in either a crash (if
+ the greetings folder didn't exist and the mailstream is invalid)
+ or an inability to read messages from the 'INBOX' folder. This
+ patch restores the mailstream to its correct state after
+ accessing the greetings. This fixes the crash, and sets the
+ mailstream to the state that VoiceMailMain expects. Note that
+ while ASTERISK-23390 also contained a patch for this issue, the
+ patch on ASTERISK-24786 is the one being merged here. Review:
+ https://reviewboard.asterisk.org/r/4459/ ASTERISK-23390 #close
+ Reported by: Ben Smithurst ASTERISK-24786 #close Reported by:
+ Graham Barnett Tested by: Graham Barnett patches:
+ app_voicemail.c.patch.SIGSEGV3rev2 uploaded by Graham Barnett
+ (License 6685) ........ Merged revisions 432695 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/stdtime/localtime.c: localtime: Fix file descriptor leak
+ on kqueue(2) systems The localtime management in the Asterisk
+ core contains a thread that watches for changes in the local
+ timezone. On systems where the directory containing
+ /etc/localtime is modified frequently, the thread monitoring the
+ changes will be woken up to determine if any changes in timezone
+ have occurred. When using kqueue(2), this can cause a leak of
+ file descriptors due to some improper management of resources.
+ This patch updates the kqueue(2) handling in localtime, such that
+ is no longer leaks resources. Review:
+ https://reviewboard.asterisk.org/r/4450/ ASTERISK-24739 #close
+ Reported by: Ed Hynan patches: 11.15.0-u.diff uploaded by Ed
+ Hynan (Licnese 6680) 11.7.0-u.diff uploaded by Ed Hynan (License
+ 6680) svn-trunk-Jan-26-2015-u.diff uploaded by Ed Hynan (License
+ 6680) ........ Merged revisions 432691 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-03-10 16:04 +0000 [r432668] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_pjsip_refer.c, res/res_pjsip_session.c,
+ include/asterisk/res_pjsip_session.h,
+ res/res_pjsip_session.exports.in: res_pjsip_refer: Fix occasional
+ unexpected BYE sent after receiving a REFER. A race condition
+ happened between initiating a transfer and requesting that a
+ dialog termination be delayed. Occasionally, the transferrer
+ channels would exit the bridge and hangup before the dialog
+ termination delay was requested. * Made request dialog
+ termination delay before initiating the transfer action. If the
+ transfer fails then cancel the delayed dialog termination
+ request. ASTERISK-24755 #close Reported by: John Bigelow Review:
+ https://reviewboard.asterisk.org/r/4460/
+
+2015-03-09 16:12 +0000 [r432638] Kevin Harwell <kharwell at digium.com>
+
+ * include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
+ res/res_pjsip_endpoint_identifier_anonymous.c,
+ res/res_pjsip_endpoint_identifier_ip.c,
+ contrib/ast-db-manage/config/versions/45e3f47c6c44_add_pjsip_endpoint_identifier_order.py
+ (added), configs/samples/pjsip.conf.sample, CHANGES,
+ res/res_pjsip.c, res/res_pjsip_endpoint_identifier_user.c:
+ res_pjsip: Allow configuration of endpoint identifier query order
+ It's possible to have a scenario that will create a conflict
+ between endpoint identifiers. For instance an incoming call could
+ be identified by two different endpoint identifiers and the one
+ chosen depended upon which identifier module loaded first. This
+ of course causes problems when, for example, the incoming call is
+ expected to be identified by username, but instead is identified
+ by ip. This patch adds a new 'global' option to res_pjsip called
+ 'endpoint_identifier_order'. It is a comma separated list of
+ endpoint identifier names that specifies the order by which
+ identifiers are processed and checked. ASTERISK-24840 #close
+ Reported by: Mark Michelson Review:
+ https://reviewboard.asterisk.org/r/4455/
+
+2015-03-08 01:46 +0000 [r432614] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Fix wrongful use of
+ USE_PJPROJECT define. As pjproject is now used as a shared
+ library a different define, HAVE_PJPROJECT, is used to specify if
+ pjproject is present. ASTERISK-24830 #close Reported by: Stefan
+ Engström
+
+2015-03-06 22:50 +0000 [r432574-432594] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_pjsip_refer.c: res_pjsip_refer: Make safely get the
+ context for a blind transfer. Made safely get the
+ TRANSFER_CONTEXT channel value while the channel is locked in
+ refer_incoming_attended_request() and
+ refer_incoming_blind_request(). The pointer returned by
+ pbx_builtin_getvar_helper() is only valid while the channel is
+ locked.
+
+ * res/res_pjsip_refer.c: res_pjsip_refer: Made
+ refer_attended_alloc() not create the ao2 object with a lock. The
+ lock is unused.
+
+2015-03-06 21:11 +0000 [r432556] Jonathan Rose <jrose at digium.com>
+
+ * include/asterisk/app.h, main/app.c: app: Add functions to swap
+ voicemail function table for testing purposes
+
+2015-03-06 20:18 +0000 [r432528-432534] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/chan_dahdi.h, channels/sig_analog.h, UPGRADE.txt:
+ chan_dahdi/sig_analog: Fix distinctive ring detection to suck
+ less. The distinctive ring feature interferes with detecting
+ Caller ID and appears to have been broken for years. What happens
+ is if you have a ring-ring cadence as used in the UK you get too
+ many DAHDI events for the distinctive ring pattern array and
+ Caller ID detection is aborted. I think when Zapata/DAHDI added
+ the ring begin event it broke distinctive ring. More events
+ happen than before and the code does no filtering of which event
+ times are recorded in the pattern array. * Made distinctive ring
+ only record the ringt count when the ring ends instead of on just
+ any DAHDI event. Distinctive ring can be ring, ring-ring,
+ ring-ring-ring, or different ring durations for the up to three
+ rings. * Fixed the distinctive ring detection enable
+ (chan_dahdi.conf option usedistinctiveringdetection) to be per
+ port instead of somewhat per port and somewhat global. This has
+ been broken since v1.8. * Fixed using the default distinctive
+ ring context when the detected pattern does not match any
+ configured dringX patterns. The default context did not get set
+ when the previous call was a matched distinctive ring pattern and
+ the current call is not matched. This has been broken since v1.8.
+ * Made distinctive ring have no effect on Caller ID detection
+ when it is disabled. Caller ID detection just monitors for 10
+ seconds before giving up. * Fixed leak of struct callerid_state
+ memory when a polarity reversal during Caller ID detection causes
+ the incoming call to be aborted. DAHDI-1143 AST-1545
+ ASTERISK-24825 #close Reported by: Richard Mudgett ASTERISK-17588
+ Reported by: Daniel Flounders Review:
+ https://reviewboard.asterisk.org/r/4444/ ........ Merged
+ revisions 432530 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: chan_sip: Fix realtime locking inversion
+ when poking a just built peer. When a realtime peer is built it
+ can cause a locking inversion when the just built peer is poked.
+ If the CLI command "sip show channels" is periodically executed
+ then a deadlock can happen because of the locking inversion. *
+ Push the peer poke off onto the scheduler thread to avoid the
+ locking inversion of the just built realtime peer. AST-1540
+ ASTERISK-24838 #close Reported by: Richard Mudgett Review:
+ https://reviewboard.asterisk.org/r/4454/ ........ Merged
+ revisions 432526 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-03-05 16:38 +0000 [r432485] George Joseph <george.joseph at fairview5.com>
+
+ * apps/app_voicemail.c, /: app_voicemail: Fix compile breaking in
+ app_voicemail with IMAP_STORAGE. There is a leftover "assert" in
+ app_voicemail/__messagecount that references variables that don't
+ exist. This causes the compile to fail when --enable-dev-mode and
+ IMAP_STORAGE are selected. This patch removes the assert.
+ Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/4461/ ........ Merged
+ revisions 432484 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-03-04 18:52 +0000 [r432453] Matthew Jordan <mjordan at digium.com>
+
+ * main/translate.c: translate: Prevent invalid memory accesses on
+ fast shutdown When a 'core restart now' or 'core stop now' is
+ executed and a channel is currently in a media operation, the
+ translator matrix can be destroyed while a channel is currently
+ blocked on getting the best translation choice (see
+ ast_translator_best_choice). When the channel gets the mutex, the
+ translation matrix now has invalid memory, and Asterisk crashes.
+ This patch does two things: (1) We now only clean up the
+ translation matrix on a graceful shutdown. In that case, there
+ are no channels, and so there is no risk of this occurring. (2)
+ We also now set the __matrix and __indextable to NULL. In some
+ initial backtraces when this occurred, it looked as if there was
+ a memory corruption occurring, and it wasn't until we determined
+ that something had restarted Asterisk that the issue became
+ clear. By setting these to NULL on shutdown, it becomes a bit
+ easier to determine why a crash is occurring. Note that we could
+ litter the code with NULL checks on the __matrix, but the act of
+ making the translation matrix cleaned up on shutdown should
+ preclude this issue from occurring in the first place, and this
+ part of the code needs to be as fast as possible. Review:
+ https://reviewboard.asterisk.org/r/4457/
+
+2015-03-02 19:14 +0000 [r432423] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Revert portion of
+ r432195 Unfortunately, while initial testing with ConfBridge did
+ not reproduce the audio problem alluded to in the comment in
+ res_pjsip_sdp_rtp, further testing did show that bridge_softmix
+ and/or ConfBridge has a severe problem bridging two or more
+ participants at different sampling rates. Sometimes, it even
+ picks odd sampling rates that cause hideous audio problems. This
+ patch backs out the offending portion of the code until the
+ issues in the affected bridging modules can be more properly
+ analyzed. ASTERISK-24841
+
+2015-02-27 18:23 +0000 [r432404] Richard Mudgett <rmudgett at digium.com>
+
+ * main/json.c, rest-api/api-docs/endpoints.json,
+ res/ari/resource_endpoints.c, res/res_ari_endpoints.c,
+ include/asterisk/json.h, res/ari/resource_channels.c: ARI: Fix
+ crash if integer values used in JSON payload 'variables' object.
+ Sending the following ARI commands caused Asterisk to crash if
+ the JSON body 'variables' object passes values of types other
+ than strings. POST /ari/channels POST /ari/channels/{channelid}
+ PUT /ari/endpoints/sendMessage PUT
+ /ari/endpoints/{tech}/{resource}/sendMessage * Eliminated
+ RAII_VAR usage in ast_ari_channels_originate_with_id(),
+ ast_ari_channels_originate(), ast_ari_endpoints_send_message(),
+ and ast_ari_endpoints_send_message_to_endpoint(). ASTERISK-24751
+ #close Reported by: jeffrey putnam Review:
+ https://reviewboard.asterisk.org/r/4447/
+
+2015-02-26 18:52 +0000 [r432385] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * include/asterisk/dial.h, main/dial.c: Dial API: add self destruct
+ option when complete This patch adds a self-destruction option to
+ the dial api. The usefulness of this is mostly when using async
+ mode to spawn a separate thread used to handle the new call,
+ while the calling thread is allowed to go on about other
+ business. The only alternative to this option would be the
+ calling thread spawning a new thread, or hanging around itself
+ waiting to destroy the dial struct after completion. Example of
+ use (minus error checking): struct ast_dial *dial =
+ ast_dial_create(); ast_dial_append(dial, "PJSIP", "200", NULL);
+ ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC,
[... 27634 lines stripped ...]
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