Author: bebuild Date: Mon Aug 11 13:50:34 2014 New Revision: 420807 URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=420807 Log: Importing files for 11.12.0-rc1 release.
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If the space left in a + stringfield is between 0 and + (alignof(ast_string_field_allocation)-1) adding new data would + cause memory corruption, because we would assume enough space + (unsigned underrun). Thanks Arnd Schmitter for reporting and + finding out the cause! ASTERISK-23508 #close Reported by: Arnd + Schmitter Tested by: Arnd Schmitter, JoshE Review: + https://reviewboard.asterisk.org/r/3898/ ........ Merged + revisions 420680 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * main/tcptls.c, /: tcptls: Avoid compiler warning on non-dev-mode. + ........ Merged revisions 420654 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-08-07 21:37 +0000 [r420435] Richard Mudgett <rmudg...@digium.com> + + * /, channels/chan_sip.c: chan_sip: Replace sip_tls_read() and + resolve the large SDP poll issue. Replace sip_tls_read() and + sip_tcp_read() with a single function and resolve the poll/wait + issue with large SDP payloads. ASTERISK-18345 #close Reported by: + Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835) + patch uploaded by Elazar Broad Review: + https://reviewboard.asterisk.org/r/3882/ ........ Merged + revisions 420434 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-08-06 16:08 +0000 [r420147] George Joseph <george.jos...@fairview5.com> + + * pbx/pbx_lua.c, main/pbx.c, /: pbx_lua: fix regression with global + sym export and context clash by pbx_config. ASTERISK-23818 (lua + contexts being overwritten by contexts of the same name in + pbx_config) surfaced because pbx_lua, having the + AST_MODFLAG_GLOBAL_SYMBOLS set, was always force loaded before + pbx_config. Since I couldn't find any reason for pbx_lua to + export it's symbols to the rest of Asterisk, I simply changed the + flag to AST_MODFLAG_DEFAULT. Problem solved. What I didn't + realize was that the symbols need to be exported not because + Asterisk needs them but because any external Lua modules like + luasql.mysql need the base Lua language APIs exported + (ASTERISK-17279). Back to ASTERISK-23818... It looks like there's + an issue in pbx.c where context_merge was only merging includes, + switches and ignore patterns if the context was already existing + AND has extensions, or if the context was brand new. If pbx_lua + is loaded before pbx_config, the context will exist BUT pbx_lua, + being implemented as a switch, will never place extensions in it, + just the switch statement. The result is that when pbx_config + loads, it never merges the switch statement created by pbx_lua + into the final context. This patch sets pbx_lua's modflag back to + AST_MODFLAG_GLOBAL_SYMBOLS and adds an "else if" in context_merge + that catches the case where an existing context has includes, + switchs or ingore patterns but no actual extensions. + ASTERISK-23818 #close Reported by: Dennis Guse Reported by: Timo + Teräs Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3891/ ........ Merged + revisions 420146 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-08-05 18:23 +0000 [r420054] Richard Mudgett <rmudg...@digium.com> + + * main/format.c: format.c: Add reason comments for the format_list + ordering. + +2014-08-04 19:44 +0000 [r419943] Rusty Newton <rnew...@digium.com> + + * main/manager.c, /: Manager - Improve documentation for manager + commands Getvar and Setvar. The documentation for these commands + did not make it clear that they could accept expressions and + functions. Modified to make this clear, but tried not to be + overly explicit. ASTERISK-21178 #close Reported by: Rusty Newton + Tested by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/3854 ........ Merged revisions + 419942 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-07-28 18:34 +0000 [r419685] Richard Mudgett <rmudg...@digium.com> + + * funcs/func_jitterbuffer.c, apps/app_queue.c, + apps/app_speech_utils.c, /, funcs/func_frame_trace.c: datastores: + Audit ast_channel_datastore_remove usage. Audit of v1.8 usage of + ast_channel_datastore_remove() for datastore memory leaks. * + Fixed leaks in app_speech_utils and func_frame_trace. * Fixed + app_speech_utils not locking the channel when accessing the + channel datastore list. Review: + https://reviewboard.asterisk.org/r/3859/ Audit of v11 usage of + ast_channel_datastore_remove() for datastore memory leaks. * + Fixed leak in func_jitterbuffer. Review: + https://reviewboard.asterisk.org/r/3860/ ........ Merged + revisions 419684 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-07-25 23:13 +0000 [r419631] Richard Mudgett <rmudg...@digium.com> + + * main/features.c, /: features.c: Allow appliationmap to use Gosub. + Using DYNAMIC_FEATURES with a Gosub application as the mapped + application does not work. It does not work because Gosub just + pushes the current dialplan context, exten, and priority onto a + stack and sets the specified Gosub location. Gosub does not have + a dialplan execution loop to run dialplan like Macro. * Made the + DYNAMIC_FEATURES application mapping feature call + ast_app_exec_macro() and ast_app_exec_sub() for the Macro and + Gosub applications respectively. * Backported + ast_app_exec_macro() and ast_app_exec_sub() from v11 to execute + dialplan routines from the DYNAMIC_FEATURES application mapping + feature. NOTE: This issue does not affect v12+ because it already + does what this patch implements. AST-1391 #close Reported by: + Guenther Kelleter Review: + https://reviewboard.asterisk.org/r/3844/ ........ Merged + revisions 419630 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-07-24 17:56 +0000 [r419441] Corey Farrell <g...@cfware.com> + + * /, channels/chan_sip.c: chan_sip: sip_subscribe_mwi_destroy + should not call sip_destroy sip_subscribe_mwi_destroy calls + sip_destroy on the reference counted mwi->call. This results in + the fields of mwi->call being freed, but mwi->call itself it + leaked. If other code is still using mwi->call it can cause + problems. This change uses dialog_unref instead, to balance the + ref provided by sip_alloc(). ASTERISK-24087 #close Reported by: + Corey Farrell Review: https://reviewboard.asterisk.org/r/3834/ + ........ Merged revisions 419440 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-07-24 16:49 +0000 [r419375] Jason Parker <jpar...@digium.com> + + * addons/chan_ooh323.c, /: Don't cause Asterisk to exit if + ooh323.conf not found. (closes issue ASTERISK-23814) ........ + Merged revisions 419374 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-07-23 13:21 +0000 [r419284] Scott Griepentrog <sgriepent...@digium.com> + + * apps/app_voicemail.c: app_voicemail: use a consistent generator + string When updating voicemail.conf when a user changes their + pin, change the generator string to be the same as the module + name when reading so that the same config_hook will be called. + Review: https://reviewboard.asterisk.org/r/3837/ + +2014-07-22 14:00 +0000 [r419162] Kinsey Moore <kmo...@digium.com> + + * tests/test_voicemail_api.c, tests/test_aoc.c, + tests/test_astobj2.c, tests/test_config.c, + addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c, + addons/chan_ooh323.c, tests/test_astobj2_thrash.c, /, + apps/app_meetme.c, tests/test_abstract_jb.c, tests/test_logger.c, + tests/test_event.c, tests/test_format_api.c, + tests/test_hashtab_thrash.c, res/res_jabber.c: Fix more dev-mode + build issues ........ Merged revisions 419129 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-07-15 22:05 +0000 [r418713] Matthew Jordan <mjor...@digium.com> + + * main/manager.c: manager: Return ActionID on nominal responses to + PresenceState action When the PresenceState action is executed, + the nominal path fails to include the ActionID in the successful + response. This patch adds a call to astman_start_ack, which + guarantees that an ActionID (if provided) will be sent back to + the AMI client. Review: https://reviewboard.asterisk.org/r/3776/ + ASTERISK-23985 #close + +2014-07-15 17:32 +0000 [r418649] Jonathan Rose <jr...@digium.com> + + * funcs/func_uri.c, /: func_uri: URIENCODE/URIDECODE - allow empty + strings as argument Previously these two dialplan functions would + issue warnings and return failure when an empty string is used as + the argument. Now they will not issue a warning and will + successfully return an empty string. ASTERISK-23911 #close + Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3745/ ........ Merged + revisions 418641 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-07-13 21:51 +0000 [r418465-418505] Corey Farrell <g...@cfware.com> + + * /, main/astobj2.c, contrib/scripts/refcounter.py: astobj2: work + around REF_DEBUG race which causes out of order log entries * + Update refcounter.py to use delta's to track the current + reference count. * Use result from internal_ao2_ref to write + old_refcount to refs_log. Review: + https://reviewboard.asterisk.org/r/3756/ ........ Merged + revisions 418504 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * apps/app_skel.c: Fix minor reference leaks in app_skel and + TEST_FRAMEWORK * Cleanup games object in app_skel. * Cleanup + stasis subscription to TEST_FRAMEWORK in manager.c (12+). Review: + https://reviewboard.asterisk.org/r/3757/ + +2014-07-11 14:23 +0000 [r418366] Scott Griepentrog <sgriepent...@digium.com> + + * main/config.c: config: inform config hook of change when writing + file When updated configuration is written back to the conf file + - for example when a user changes their voicemail pin, make sure + that any config hook that wants to know of changes is informed. + Review: https://reviewboard.asterisk.org/r/3708/ + +2014-07-10 15:35 +0000 [r418323] Matthew Jordan <mjor...@digium.com> + + * include/asterisk/xmpp.h: include/asterisk/xmpp.h: Convert + indentation to tabs This is a whitespace only change. + +2014-07-10 01:42 +0000 [r418262] Richard Mudgett <rmudg...@digium.com> + + * /, channels/sig_pri.c: chan_dahdi/sig_pri: Fix type mismatch in + the idledial feature's channel creation. Square pegs in round + holes don't work very well. ........ Merged revisions 418261 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-07-10 Asterisk Development Team <asteriskt...@digium.com> + + * Asterisk 11.11.0 Released. + +2014-07-08 Asterisk Development Team <asteriskt...@digium.com> + + * Asterisk 11.11.0-rc1 Released. + +2014-07-03 21:48 +0000 [r417957] Richard Mudgett <rmudg...@digium.com> + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, /, UPGRADE.txt, + channels/sig_pri.c: chan_dahdi: Add inband_on_setup_ack + compatibility option. The new inband_on_setup_ack option causes + Asterisk to assume inband audio may be present when a + SETUP_ACKNOWLEDGE message is received. Q.931 Section 5.1.3 says + that in scenarios with overlap dialing, when a dialtone is sent + from the network side, progress indicator 8 "Inband info now + available" MAY be sent to the CPE if no digits were received with + the SETUP. It is thus implied that the ie is mandatory if digits + came with the SETUP and dialtone is needed. This option should be + enabled, when the network sends dialtone and you want to hear it, + but the network doesn't send the progress indicator when needed. + NOTE: For Q.SIG setups this option should be enabled when + outgoing overlap dialing is also enabled because Q.SIG does not + send the progress indicator with the SETUP ACK. The commit + -r413714 (AST-1338) which causes this issue was dealing with a + SIP-to-ISDN interoperability issue. This commit is a merge of the + two patches indicated below. ASTERISK-23897 #close Reported by: + Pavel Troller Patches: pri-4.diff (license #6302) patch uploaded + by Pavel Troller jira_asterisk_23897_v11.patch (license #5621) + patch uploaded by rmudgett Review: + https://reviewboard.asterisk.org/r/3633/ ........ Merged + revisions 417956 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-07-03 11:24 +0000 [r417798] Matthew Jordan <mjor...@digium.com> + + * /, main/utils.c: main/untils: Prevent potential infinite loop in + ast_careful_fwrite A loop in ast_careful_fwrite exists that will + continually attempt to write to a file stream, even in the + presence of EAGAIN/EINTR errors. However, if a connection that + uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's + call to fflush may return EAGAIN/EINTER along with EOF. A + subsequent call to fflush will return EOF but not clear errno, + resulting in an infinite loop. This patch clears errno after it + is detected and handled the loop, such that any subsequent call + to fflush will not get erroneously stuck. Review: + https://reviewboard.asterisk.org/r/3704 #ASTERISK-23984 #close + Reported by: Steve Davies patches: fflush_loop_fix uploaded by + one47 (License 5012) ........ Merged revisions 417797 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-30 19:42 +0000 [r417677] Joshua Colp <jc...@digium.com> + + * channels/sip/include/sip.h, res/res_rtp_asterisk.c, + main/rtp_engine.c, channels/chan_sip.c, UPGRADE.txt, + configs/sip.conf.sample, include/asterisk/rtp_engine.h: + res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS + negotiation on RTCP. This change fixes up DTLS support in + res_rtp_asterisk so it can accept and provide a SHA-256 + fingerprint, so it occurs on RTCP, and so it occurs after ICE + negotiation completes. Configuration options to chan_sip have + also been added to allow behavior to be tweaked (such as forcing + the AVP type media transports in SDP). ASTERISK-22961 #close + Reported by: Jay Jideliov Review: + https://reviewboard.asterisk.org/r/3679/ + +2014-06-30 03:23 +0000 [r417588] Matthew Jordan <mjor...@digium.com> + + * /, channels/chan_sip.c: chan_sip: be more tolerant of whitespace + between attributes in SDP fmtp line This patch is essentially a + backport of a small portion of r397526 from ASTERISK-21981. In + that patch, pass through support and format attribute negotiation + was added for Opus. Part of that included being more tolerant to + whitespace in the fmtp line of an SDP; that part of the patch is + being applied here. As the author of the backport pointed out, in + SDP, the fmtp line is allowed to include whitespace between + attributes. RFC 3267 chapter 8.3 (from 2001) includes an example + for this. This was not removed in the updated RFC 4867 in 2007. + Review: https://reviewboard.asterisk.org/r/3658 ASTERISK-23916 + #close Reported by: Alexander Traud patches: + sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud + (License 6520) ........ Merged revisions 417587 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-27 19:26 +0000 [r417481-417505] Corey Farrell <g...@cfware.com> + + * /, main/astobj2.c: Ensure REF_DEBUG records entrys for attempts + to ao2_ref an invalid object This change ensures that + __ao2_ref_debug writes to ref_log when given a non-NULL pointer + to an invalid ao2 object. This is to ensure that we record any + attempt manipulate references of already freed objects. + ASTERISK-23948 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3677/ ........ Merged + revisions 417500 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * /, contrib/scripts/refcounter.py: refcounter.py: prevent use of + excessive RAM with large refs logs When processing a 212MB refs + file, refcounter.py used over 3GB of RAM. This change greatly + reduces memory usage in two ways: * Saving object history in + whole lines instead of separated values. * Not saving + normal/skewed/leaked object lists unless they are requested. + ASTERISK-23921 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3668/ ........ Merged + revisions 417480 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-26 18:25 +0000 [r417310-417419] Matthew Jordan <mjor...@digium.com> + + * res/res_http_websocket.exports.in: res_http_websocket: Export + symbol for ast_websocket_set_timeout Thanks to Sean Bright for + pointing out that this was missed in #asterisk-dev. + + * main/udptl.c, /: udptl: Correct FEC to not consider negative + sequence numbers as missing When using FEC, with span=3 and + entries=4 Asterisk will attempt to repair the packet with + sequence number 5, as it will see that packet -4 is missing. The + result is Asterisk sending garbage packets that can kill a fax. + This patch adds a check to see if the sequence number is valid + before checking if the packet is missing. Review: + https://reviewboard.asterisk.org/r/3657/ #ASTERISK-23908 #close + Reported by: Torrey Searle patches: udptl_fec.patch uploaded by + Torrey Searle (License 5334) ........ Merged revisions 417318 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * UPGRADE.txt, configs/sip.conf.sample, res/res_http_websocket.c, + channels/sip/include/sip.h, channels/chan_sip.c, + include/asterisk/http_websocket.h: res_http_websocket: Close + websocket correctly and use careful fwrite When a client takes a + long time to process information received from Asterisk, a write + operation using fwrite may fail to write all information. This + causes the underlying file stream to be in an unknown state, such + that the socket must be disconnected. Unfortunately, there are + two problems with this in Asterisk's existing websocket code: 1. + Periodically, during the read loop, Asterisk must write to the + connected websocket to respond to pings. As such, Asterisk + maintains a reference to the session during the loop. When + ast_http_websocket_write fails, it may cause the session to + decrement its ref count, but this in and of itself does not break + the read loop. The read loop's write, on the other hand, does not + break the loop if it fails. This causes the socket to get in a + 'stuck' state, preventing the client from reconnecting to the + server. 2. More importantly, however, is that the fwrite in + ast_http_websocket_write fails with a large volume of data when + the client takes awhile to process the information. When it does + fail, it fails writing only a portion of the bytes. With some + debugging, it was shown that this was failing in a similar + fashion to ASTERISK-12767. Switching this over to + ast_careful_fwrite with a long enough timeout solved the problem. + ASTERISK-23917 #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3624/ + +2014-06-26 10:04 +0000 [r417249] Corey Farrell <g...@cfware.com> + + * /, channels/chan_sip.c: chan_sip: Fix handling of "From" headers + longer than 256 characters From headers were processed using a + 256 character buffer on the stack. This change replaces that with + a heap allocation by ast_strdup. ASTERISK-23790 #close Reported + by: uniken1 Tested by: uniken1 Review: + https://reviewboard.asterisk.org/r/3669/ Patches: + chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes + (license 5674) ........ Merged revisions 417248 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-23 18:49 +0000 [r417141] Joshua Colp <jc...@digium.com> + + * res/res_rtp_asterisk.c: res_rtp_asterisk: Return the length of + data written when sending via ICE instead of 0. ASTERISK-23834 + #close Reported by: Richard Kenner + +2014-06-23 14:35 +0000 [r417077] Rusty Newton <rnew...@digium.com> + + * configs/features.conf.sample: main/features - documentation - + reformat examples and options in features.conf.sample to show + clearly which options apply in which section The features.conf + sample can be a bit confusing about what parking options can be + set only in the general context, or both in the general context + (for the default parking lot) and in other parking lot contexts. + A bug was filed due to confusion and a little googling will show + lots of other confused users. Despite some comments on the + individual options, it still reads in a confusing way. In this + patch I separate out those options with some headings in to + attempt a better layout. I went ahead and modified other headings + in the file, or added them to facilitate better visual scanning. + ASTERISK-23667 Review: https://reviewboard.asterisk.org/r/3622/ + +2014-06-22 20:52 +0000 [r417017] George Joseph <george.jos...@fairview5.com> + + * Makefile.rules, Makefile, /: build: Turn FORTIFY_SOURCE off if + DONT_OPTIMIZE is set. AST_FORTIFY_SOURCE is automatically set in + ./Makefile even if DONT_OPTIMIZE is set in menuselect. This + causes gcc to complain that _FORTIFY_SOURCE requires optimization + and the build will fail. You can specify "make + AST_FORTIFY_SOURCE=''" but I always forget. This patch moves the + set of AST_FORTIFY_SOURCE to Makefile.rules and only sets it if + DONT_OPTIMIZE is "no". The move is necessary because the + top-level Makefile doesn't include menuselect.makeopts. This + doesn't solve the entire problem however because res_config_mysql + seems to force _FORTIFY_SOURCE so res_config_mysql has to be + disabled for now if DONT_OPTIMIZE is set. Tested by: George + Joseph Review: https://reviewboard.asterisk.org/r/3664/ ........ + Merged revisions 417016 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-20 23:14 +0000 [r416870-416930] George Joseph <george.jos...@fairview5.com> + + * /, configure, include/asterisk/autoconfig.h.in: build: Allow + autoconf/ast_ext_tool_check to handle cross-compiling better. + ast_ext_tool_check.m4 isn't handling cases where a path to a + package is provided (E.G. --with-mysqlclient=/some/sysroot) and + the package has a config tool (E.G. mysql_config) and the package + has its own subdirectories in include or lib. For example, + mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but + ast_ext_tool_check sets MYSQLCLIENT_LIB to + ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its + includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not + directly in ${LIBXML2_DIR}/usr/include. Both cause configure to + fail and there are others in the same boat. The problem is caused + by logic in ast_ext_tool_check that overrides the result of the + config tool's --cflags and --libs options if package_DIR is set. + This patch prepends package_DIR (if specified) to the -L and -I + results from the package's config tool instead of overriding + them. A regenerated ./configure and + include/asterisk/autoconfig.h.in are included but can be + regenerated by running ./bootstrap.sh at any time. Tested by: + George Joseph Tested by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3550/ ........ Merged + revisions 416929 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * autoconf/ast_ext_tool_check.m4: build: Allow + autoconf/ast_ext_tool_check to handle cross-compiling better. + ast_ext_tool_check.m4 isn't handling cases where a path to a + package is provided (E.G. --with-mysqlclient=/some/sysroot) and + the package has a config tool (E.G. mysql_config) and the package + has its own subdirectories in include or lib. For example, + mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but + ast_ext_tool_check sets MYSQLCLIENT_LIB to + ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its + includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not + directly in ${LIBXML2_DIR}/usr/include. Both cause configure to + fail and there are others in the same boat. The problem is caused + by logic in ast_ext_tool_check that overrides the result of the + config tool's --cflags and --libs options if package_DIR is set. + This patch prepends package_DIR (if specified) to the -L and -I + results from the package's config tool instead of overriding + them. Tested by: George Joseph Tested by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3550/ + +2014-06-19 19:34 +0000 [r416733] Kinsey Moore <kmo...@digium.com> + + * main/bridging.c, /, channels/sip/reqresp_parser.c, main/logger.c, + main/test.c: Fix build warnings with TEST_FRAMEWORK enabled + ........ Merged revisions 416732 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-19 16:02 +0000 [r416581-416668] George Joseph <george.jos...@fairview5.com> + + * pbx/pbx_lua.c: Remove the problematic and unneeded + AST_MODFLAG_GLOBAL_SYMBOLS from pbx_lua.c + AST_MODFLAG_GLOBAL_SYMBOLS was causing the module to be + incorrectly loaded before pbx_config. pbx_config was therefore + blowing away contexts that were created by pbx_lua. With + AST_MODFLAG_DEFAULT the load order is now correct and contexs are + being properly merged. AST_MODFLAG_GLOBAL_SYMBOLS was not needed + anyway since no other modules needed its global symbols that + early. ASTERISK-23818 #close Reported by: Dennis Guse Tested by: + Dennis Guse Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3629/ + + * configs/extensions.lua.sample: Update extensions.lua.sample with + naming conflict guidance. The sample extensions.lua was causing + pbx_lua to fail to load when parsing 'app.goto("default", "s", + 1)' because in Lua 5.2, 'goto' is now a reserved word. This patch + adds guidance to extensions.lua.sample and changed + 'app.goto("default", "s", 1)' to 'app.['goto']("default", "s", + 1)'. ASTERISK-23844 #close Reported by: rnewton Tested by: + gtjoseph Review: https://reviewboard.asterisk.org/r/3627/ + +2014-06-17 18:40 +0000 [r416501] Mark Michelson <mmichel...@digium.com> + + * /, funcs/func_strings.c: Allow the PUSH and UNSHIFT functions to + set inheritable channel variables. ........ Merged revisions + 416500 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-17 16:21 +0000 [r416440] Kinsey Moore <kmo...@digium.com> + + * res/res_musiconhold.c, /: MoH: Don't restart stream on repeated + start calls Currently, music on hold will stop and then start + again from the beginning if ast_moh_start() is called multiple + times. This can happen if a call is put on hold repeatedly (the + channel receives multiple HOLD control frames) and can be + triggered from ARI by starting MoH on a channel multiple times. + This is fairly jarring/annoying to users. This change prevents + MoH from being restarted if the requested music class is the same + as the one currently playing. This includes an extra check to + prevent the errors previously experienced in the testsuite and + has 100+ test runs behind it. Review: + https://reviewboard.asterisk.org/r/3615/ ........ Merged + revisions 416439 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-16 09:00 +0000 [r416337] Igor Goncharovskiy <igor.goncharov...@gmail.com> + + * cel/cel_sqlite3_custom.c, main/db.c, res/res_config_sqlite3.c, + cdr/cdr_sqlite3_custom.c, /: We have faced situation when using + CDR and CEL by sqlite3 modules. With system having high load + (~100 concurrent calls created by sipp) we found many cdr and cel + records missed. There is special finction in sqlite3, that make + able to fix this situation - sqlite3_wait_timeout, that also can + replace awful code cdr_sqlite3 ad cel_sqlite3 modules. Also this + function can be used for aastdb and res_config_sqlite3 to avoid + missed writes to sqlite db. #ASTERISK-23766 #close Reported by: + Igor Goncharovsky Review: + https://reviewboard.asterisk.org/r/3559/ ........ Merged + revisions 416336 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-15 21:17 +0000 [r416252] Matthew Jordan <mjor...@digium.com> + + * /, res/res_musiconhold.c: MoH: Undo commit r416150 (1.8) This + patch reverts r416150. When the comparison between mohclass->name + and state->class->name is made, you are not guaranteed that (a) + state->class is non-NULL or that state or state->class are in a + safe state. Crashes caught by the bridges/transfer_capabilities + test. ........ Merged revisions 416251 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-13 13:08 +0000 [r416151] Kinsey Moore <kmo...@digium.com> + + * /, res/res_musiconhold.c: MoH: Don't restart stream on repeated + start calls Currently, music on hold will stop and then start + again from the beginning if ast_moh_start() is called multiple + times. This can happen if a call is put on hold repeatedly (the + channel receives multiple HOLD control frames) and can be + triggered from ARI by starting MoH on a channel multiple times. + This is fairly jarring/annoying to users. This change prevents + MoH from being restarted if the requested music class is the same + as the one currently playing. Review: + https://reviewboard.asterisk.org/r/3615/ ........ Merged + revisions 416150 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-13 05:06 +0000 [r416067] Richard Mudgett <rmudg...@digium.com> + + * main/tcptls.c, main/manager.c, /, channels/chan_sip.c, + main/http.c, include/asterisk/tcptls.h: AST-2014-007: Fix of fix + to allow AMI and SIP TCP to send messages. ASTERISK-23673 #close + Reported by: Richard Mudgett Review: + https://reviewboard.asterisk.org/r/3617/ ........ Merged + revisions 416066 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-12 21:16 +0000 [r415999] Rusty Newton <rnew...@digium.com> + + * main/pbx.c, /: main/pbx - documentation - enhance 'core show + hints' and 'core show hint' help text Adds descriptive help text + to 'core show hints' and 'core show hint'. The text describes the + various columns for the sake of clarity. ASTERISK-23764 Review: + https://reviewboard.asterisk.org/r/3610/ ........ Merged + revisions 415998 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-12 17:20 +0000 [r415915] Corey Farrell <g...@cfware.com> + + * channels/sip/sdp_crypto.c, /: chan_sip: DEBUG messages in + sdp_crypto.c display despite a DEBUG level of zero Change debug + level for messages in sdp_crypto.c from zero to one. This ensures + the messages are not displayed when debugging is disabled. Change + does not apply to 12+ as it was already fixed in those versions. + ASTERISK-23246 #close Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/3605/ ........ Merged + revisions 415908 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-12 16:22 +0000 [r415854] Richard Mudgett <rmudg...@digium.com> + + * res/res_http_websocket.c, configs/http.conf.sample, + include/asterisk/utils.h, main/tcptls.c, main/manager.c, /, + channels/chan_sip.c, main/http.c, UPGRADE.txt, main/utils.c, + include/asterisk/tcptls.h: AST-2014-007: Fix DOS by consuming the + number of allowed HTTP connections. Simply establishing a TCP + connection and never sending anything to the configured HTTP port + in http.conf will tie up a HTTP connection. Since there is a + maximum number of open HTTP sessions allowed at a time you can + block legitimate connections. A similar problem exists if a HTTP + request is started but never finished. * Added http.conf + session_inactivity timer option to close HTTP connections that + aren't doing anything. Defaults to 30000 ms. * Removed the + undocumented manager.conf block-sockets option. It interferes + with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections + now have better authentication timeout protection. Though I + didn't remove the bizzare TLS timeout polling code from chan_sip. + * chan_sip can now handle SSL certificate renegotiations in the + middle of a session. It couldn't do that before because the + socket was non-blocking and the SSL calls were not restarted as + documented by the OpenSSL documentation. * Fixed an off nominal + leak of the ssl struct in handle_tcptls_connection() if the FILE + stream failed to open and the SSL certificate negotiations + failed. The patch creates a custom FILE stream handler to give + the created FILE streams inactivity timeout and timeout after a + specific moment in time capability. This approach eliminates the + need for code using the FILE stream to be redesigned to deal with + the timeouts. This patch indirectly fixes most of ASTERISK-18345 + by fixing the usage of the SSL_read/SSL_write operations. + ASTERISK-23673 #close Reported by: Richard Mudgett ........ + Merged revisions 415841 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-12 15:42 +0000 [r415837] Jonathan Rose <jr...@digium.com> + + * UPGRADE.txt: Correct UPGRADE.txt notes in r415825 The change was + marked against the wrong version of Asterisk. My apologies. + +2014-06-12 15:40 +0000 [r415835] Scott Griepentrog <sgriepent...@digium.com> + + * /, apps/app_queue.c: app_queue: delayed state can cause early + leavewhenempty ringing In app_queue, device state changes arrive + in event messages and update the queue member status value. That + value is checked in get_member_status() to decide that the caller + should leave when there are no available members. Although event + messages can be delayed by other activity, there is no adverse + affect by lagged status except in one specific case: there is + only one available member, it was just rung, and leavewhenempty + is enabled set for ringing members. This change adds a direct + check of the device state only under this condition where the + caller may be dropped incorrectly, resolving this issue without + affecting performance of app_queue normally. AST-1248 #close + Review: https://reviewboard.asterisk.org/r/3595/ Reported by: + Thomas Arimont ........ Merged revisions 415833 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-12 15:22 +0000 [r415825] Jonathan Rose <jr...@digium.com> + + * UPGRADE.txt, apps/app_mixmonitor.c: MixMonitor: Add class + authorization requirements to MixMonitor AMI commands MixMonitor + AMI commands StartMixMonitor and StopMixMonitor lacked class + authorization. StopMixMonitor now requires that the manager user + either have the call or system class authorization. + StartMixMonitor is a slightly larger issue since it can execute + shell commands if the right arguments are passed into it, and we + consider this a permission escalation. A security release will be + issued for problem this shortly. ASTERISK-23609 #close Reported + by: Corey Farrell + +2014-06-11 22:44 +0000 [r415728] Richard Mudgett <rmudg...@digium.com> + + * main/format.c: format.c: Fix misuse of hash container function. + The supplied hash function to a container must be idempotent + given the object's key value to figure out which container bucket + the object belongs in. Returning a random number or the current + container count is not idempotent. The "computed hash" value + doesn't help find the object later in those cases. * Fixed the + format_list container to actually be a list since that is how the + container is used. Conceptually, if more than 283 formats were + added to the format_list then odd things may have happened before + the fix. + +2014-06-10 09:13 +0000 [r415599] Alexandr Anikin <m...@telecom-service.ru> + + * addons/chan_ooh323.c: chan_ooh323: fix loading module failure if + there no accessible h323_log or ooh323 config file change return + 1 to return AST_MODULE_LOAD_FAILURE on module load routine few + cosmetic changes ASTERISK-23814 #close (closes issue + ASTERISK-23814) Reported by: Igor Goncharovsky Patches: + ASTERISK-23814-ast11.patch + +2014-06-09 11:57 +0000 [r415522] Walter Doekes <walter+aster...@wjd.nu> + + * contrib/scripts/safe_asterisk, /: safe_asterisk: Cleanup + additions to r415132. Replaced a stray echo that should've been a + message call in safe_asterisk. I'm using the contents of the old + message inside the if $NOTIFY so peoples log parsing scripts + won't get confused by new messages. I'll clean that up in trunk. + (Note that a 'make install' still won't overwrite your old + safe_asterisk if it exists. See ASTERISK-21965.) ASTERISK-23492 + #close ........ Merged revisions 415521 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-09 03:47 +0000 [r415464] Corey Farrell <g...@cfware.com> + + * main/autoservice.c, /: autoservice: stop thread on graceful + shutdown This change adds thread shutdown to autoservice for + graceful shutdowns only. ast_register_cleanup is backported to + 1.8 to allow this. The logger callid is also released on shutdown + in 11+. ASTERISK-23827 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3594/ ........ Merged + revisions 415463 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2014-06-06 21:27 +0000 [r415390] Jonathan Rose <jr...@digium.com> + [... 30385 lines stripped ...] -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- svn-commits mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/svn-commits