[asterisk-commits] bebuild: tag certified-11.6-cert5 r422744 - /certified/tags/11.6-cert5/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Sep 5 19:34:12 CDT 2014
Author: bebuild
Date: Fri Sep 5 19:34:08 2014
New Revision: 422744
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=422744
Log:
Importing files for 11.6-cert5 release.
Modified:
certified/tags/11.6-cert5/.version
certified/tags/11.6-cert5/ChangeLog
Modified: certified/tags/11.6-cert5/.version
URL: http://svnview.digium.com/svn/asterisk/certified/tags/11.6-cert5/.version?view=diff&rev=422744&r1=422743&r2=422744
==============================================================================
--- certified/tags/11.6-cert5/.version (original)
+++ certified/tags/11.6-cert5/.version Fri Sep 5 19:34:08 2014
@@ -1,1 +1,1 @@
-11.6.0
+11.6-cert5
Modified: certified/tags/11.6-cert5/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/certified/tags/11.6-cert5/ChangeLog?view=diff&rev=422744&r1=422743&r2=422744
==============================================================================
--- certified/tags/11.6-cert5/ChangeLog (original)
+++ certified/tags/11.6-cert5/ChangeLog Fri Sep 5 19:34:08 2014
@@ -1,3 +1,657 @@
+2014-09-05 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 11.6-cert5 Released.
+
+2014-08-17 01:54 +0000 [r421209] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_snmp.c, apps/app_dictate.c, apps/app_test.c,
+ apps/app_ices.c, res/res_http_websocket.c, cdr/cdr_radius.c,
+ build_tools/cflags.xml, funcs/func_pitchshift.c,
+ apps/app_osplookup.c, funcs/func_frame_trace.c,
+ channels/console_gui.c, apps/app_mp3.c, pbx/pbx_ael.c,
+ channels/console_board.c, formats/format_jpeg.c,
+ channels/chan_mgcp.c, res/res_config_pgsql.c, cel/cel_tds.c,
+ apps/app_dahdiras.c, res/res_ael_share.c, apps/app_talkdetect.c,
+ utils/conf2ael.c, channels/chan_jingle.c, channels/chan_misdn.c,
+ formats/format_vox.c, res/res_timing_pthread.c,
+ res/res_corosync.c, cel/cel_sqlite3_custom.c, apps/app_sms.c,
+ apps/app_zapateller.c, res/res_fax_spandsp.c,
+ res/res_timing_kqueue.c, utils/check_expr.c,
+ channels/chan_unistim.c, build_tools/cflags-devmode.xml,
+ utils/muted.c, cdr/cdr_sqlite3_custom.c, res/res_phoneprov.c,
+ channels/console_video.c, apps/app_alarmreceiver.c,
+ apps/app_chanisavail.c, apps/app_image.c, channels/chan_gtalk.c,
+ cdr/cdr_pgsql.c, res/res_config_sqlite.c, res/res_pktccops.c,
+ cdr/cdr_csv.c, utils/stereorize.c, channels/chan_phone.c,
+ channels/chan_skinny.c, build_tools/embed_modules.xml,
+ apps/app_minivm.c, pbx/pbx_realtime.c, apps/app_amd.c,
+ channels/chan_alsa.c, apps/app_url.c, apps/app_externalivr.c,
+ cdr/cdr_odbc.c, res/res_config_ldap.c, apps/app_jack.c,
+ apps/app_adsiprog.c, utils/refcounter.c, apps/app_nbscat.c,
+ apps/app_festival.c, apps/app_waitforsilence.c, utils/astman.c,
+ apps/app_morsecode.c, utils/smsq.c, pbx/pbx_lua.c,
+ channels/chan_console.c, apps/app_getcpeid.c,
+ channels/chan_oss.c, cdr/cdr_tds.c, apps/app_waitforring.c,
+ pbx/pbx_dundi.c, utils/ael_main.c, utils/extconf.c,
+ channels/chan_nbs.c, utils/streamplayer.c, cel/cel_pgsql.c,
+ cel/cel_radius.c: Add missing commit from 11.2-cert This disables
+ building by default for all extended modules for Certified
+ Asterisk 11.6. This commit was missed from 11.2-cert when
+ creating the 11.6-cert branch. ASTERISK-24104 #close Reported by:
+ Rusty Newton
+
+2014-08-08 17:18 +0000 [r420559] Richard Mudgett <rmudgett at 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 ........ Merged
+ revisions 420435 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-25 23:27 +0000 [r419662] Richard Mudgett <rmudgett at 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 ........ Merged
+ revisions 419631 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-23 14:34 +0000 [r419308] Scott Griepentrog <sgriepentrog at 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/ ........
+ Merged revisions 419284 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-11 16:39 +0000 [r418368] Scott Griepentrog <sgriepentrog at 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/
+ ........ Merged revisions 418366 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-01 15:37 +0000 [r417724] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, /,
+ channels/chan_sip.c, UPGRADE.txt, configs/sip.conf.sample,
+ include/asterisk/rtp_engine.h, channels/sip/include/sip.h:
+ Multiple revisions
+ 402345,405234,409129-409130,409565,413008,417141,417677 ........
+ r402345 | kmoore | 2013-11-01 05:31:49 -0700 (Fri, 01 Nov 2013) |
+ 11 lines chan_sip: Fix RTCP port for SRFLX ICE candidates This
+ corrects one-way audio between Asterisk and Chrome/jssip as a
+ result of Asterisk inserting the incorrect RTCP port into RTCP
+ SRFLX ICE candidates. This also exposes an ICE component
+ enumeration to extract further details from candidates. (closes
+ issue ASTERISK-21383) Reported by: Shaun Clark Review:
+ https://reviewboard.asterisk.org/r/2967/ ........ r405234 |
+ kharwell | 2014-01-09 08:49:55 -0800 (Thu, 09 Jan 2014) | 19
+ lines res_rtp_asterisk: Fails to resume WebRTC call from hold In
+ ast_rtp_ice_start if the ice session create check list failed,
+ start check was never initiated and ice_started was never set to
+ true. Upon re-entering the function (for instance, [un]hold) it
+ would try to create the check list again with duplicate remote
+ candidates. Fixed so that if the create check list fails the
+ necessary data structures are properly re-initialized for any
+ subsequent retries. Note, it was decided to not stop ice support
+ (by calling ast_rtp_ice_stop) on a check list failure because it
+ possible things might still work. However, a debug message was
+ added to help with any future troubleshooting. (closes issue
+ ASTERISK-22911) Reported by: Vytis ValentinaviÄius Patches:
+ works_on_my_machine.patch uploaded by xytis (license 6558)
+ ........ r409129 | jrose | 2014-02-27 11:19:02 -0800 (Thu, 27 Feb
+ 2014) | 15 lines res_rtp_asterisk: Fix checklist creating
+ problems in ICE sessions Prior to this patch, local candidate
+ lists including SRFLX would fail to start properly when building
+ ICE candidate check lists. This patch fixes that problem by
+ making sure that each SRFLX candidate is associated with the
+ proper base address so that the check list can create matches
+ properly. This patch was written by jcolp. The issue will be left
+ open to await testing by the issue participants. (issue
+ ASTERISK-23213) Reported by: Andrea Suisani Review:
+ https://reviewboard.asterisk.org/r/3256/ ........ r409130 | jrose
+ | 2014-02-27 11:38:10 -0800 (Thu, 27 Feb 2014) | 8 lines
+ res_rtp_asterisk: correct build error from r409129 Accidentally
+ placed a declaration below functional code (issue ASTERISK-23213)
+ Reported by: Andrea Suisani Review:
+ https://reviewboard.asterisk.org/r/3256/ ........ r409565 | jrose
+ | 2014-03-04 08:40:39 -0800 (Tue, 04 Mar 2014) | 9 lines
+ res_rtp_asterisk: Fix one way audio problems with hold/unhold
+ when using ICE ICE sessions will now be restarted if sessions are
+ changed to use new sets of remote candidates. (closes issue
+ ASTERISK-22911) Reported by: Vytis ValentinaviÄius Review:
+ https://reviewboard.asterisk.org/r/3275/ ........ r413008 |
+ mjordan | 2014-04-25 10:47:21 -0700 (Fri, 25 Apr 2014) | 14 lines
+ res_rtp_asterisk: Add support for DTLS handshake retransmissions
+ On congested networks, it is possible for the DTLS handshake
+ messages to get lost. This patch adds a timer to res_rtp_asterisk
+ that will periodically check to see if the handshake has
+ succeeded. If not, it will retransmit the DTLS handshake. Review:
+ https://reviewboard.asterisk.org/r/3337 ASTERISK-23649 #close
+ Reported by: Nitesh Bansal patches: dtls_retransmission.patch
+ uploaded by Nitesh Bansal (License 6418) ........ r417141 | file
+ | 2014-06-23 11:49:14 -0700 (Mon, 23 Jun 2014) | 5 lines
+ res_rtp_asterisk: Return the length of data written when sending
+ via ICE instead of 0. ASTERISK-23834 #close Reported by: Richard
+ Kenner ........ r417677 | file | 2014-06-30 12:42:18 -0700 (Mon,
+ 30 Jun 2014) | 12 lines 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/ ........ Merged
+ revisions 402345,405234,409129-409130,409565,413008,417141,417677
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-13 05:29 +0000 [r415977-416106] Richard Mudgett <rmudgett at 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 ........ Merged
+ revisions 416067 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/http.c, UPGRADE.txt, main/utils.c,
+ include/asterisk/tcptls.h, res/res_http_websocket.c,
+ configs/http.conf.sample, include/asterisk/utils.h,
+ main/tcptls.c, main/manager.c, /, channels/chan_sip.c:
+ 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 ........ Merged
+ revisions 415854 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-12 16:27 +0000 [r415867] Scott Griepentrog <sgriepentrog at 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 16:06 +0000 [r415842] Jonathan Rose <jrose at 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 ........ Merged revisions 415837 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-13 00:48 +0000 [r413773] Richard Mudgett <rmudgett at digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sig_pri.c: chan_dahdi/sig_pri: Prevent unnecessary
+ PROGRESS events when overlap dialing is enabled. When overlap
+ dialing is enabled, the lack of inband audio available
+ information in the SETUP_ACKNOWLEDGE events causes an
+ interoperability problem with SIP. sig_pri doesn't know if there
+ is dialtone present when a SETUP_ACKNOWLEDGE is received so it
+ assumes it is there and posts an AST_CONTROL_PROGRESS frame. The
+ SIP channel driver then sends out a 183 Session Progress and
+ blocks the desired 180 Ringing message when the ALERTING message
+ comes in. * Made the configure script detect if the installed
+ version of libpri supports the SETUP_ACKNOWLEDGE enhancements. *
+ Using the new API, made generate an AST_CONTROL_PROGRESS frame on
+ an incoming SETUP_ACKNOWLEDGE message when the message indicates
+ inband audio is present instead of assuming that dialtone is
+ present. * Using the new API, made SETUP_ACKNOWLEDGE send out an
+ inband audio available indication only if dialtone is expected.
+ The change also makes the fallback behaviour of sending the
+ PROGRESS message better by sending it only if dialtone is
+ expected. * Changed receiving a PROCEEDING message to not
+ generate an AST_CONTROL_PROGRESS frame if the progress indication
+ ie indicates non-end-to-end-ISDN. This helps interoperability
+ with SIP. * Changed sending a PROCEEDING message in response to
+ an AST_CONTROL_PROCEEDING frame to not indicate inband audio
+ available. It was silly to do so anyway because the channel
+ driver doesn't know if inband audio is even available. This helps
+ interoperability with SIP. This patch and a corresponding change
+ in libpri work together to allow Asterisk to control the inband
+ audio available progress indication ie on the SETUP_ACKNOWLEDGE
+ message when dialtone is present. AST-1338 #close Reported by:
+ Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/
+ ........ Merged revisions 413714 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 413765 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-04-11 17:27 +0000 [r412212] Kevin Harwell <kharwell at digium.com>
+
+ * main/asterisk.c, /: asterisk.c: suppress live_dangerously warning
+ on rasterisk Even since the fixes of AST-2013-007, Asterisk
+ prints the following warning on startup if the user decided to
+ live dangerously: Privilege escalation protection disabled! See
+ https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This
+ message is intended for the logs and interactive startup. No need
+ for it to appear on a remote console. This commit removes it from
+ there. (closes issue ASTERISK-23084) Review:
+ https://reviewboard.asterisk.org/r/3101/ ........ Merged
+ revisions 404861 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 404888 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-10 17:34 +0000 [r410429] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/http.c: AST-2014-001: Stack overflow in HTTP processing
+ of Cookie headers. Sending a HTTP request that is handled by
+ Asterisk with a large number of Cookie headers could overflow the
+ stack. Another vulnerability along similar lines is any HTTP
+ request with a ridiculous number of headers in the request could
+ exhaust system memory. (closes issue ASTERISK-23340) Reported by:
+ Lucas Molas, researcher at Programa STIC, Fundacion; and Dr.
+ Manuel Sadosky, Buenos Aires, Argentina ........ Merged revisions
+ 410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 410381 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-10 14:04 +0000 [r410359] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: AST-2014-002: chan_sip: Exit early on bad
+ session timers request This change allows chan_sip to avoid
+ creation of the channel and consumption of associated file
+ descriptors altogether if the inbound request is going to be
+ rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey
+ Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey
+ Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by
+ Corey Farrell (license 5909) ........ Merged revisions 410308
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 410311 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-02-19 19:17 +0000 [r408392] Richard Mudgett <rmudgett at digium.com>
+
+ * main/config.c, /: config: Add file size and nanosecond resolution
+ fields to the cached modified config file information. Repeatedly
+ modifying config files and reloading too fast sometimes fails to
+ reload the configuration because the cached modification
+ timestamp has one second resolution. * Added file size and
+ nanosecond resolution fields to the cached config file
+ modification timestamp information. Now if the file size changes
+ or the file system supports nanosecond resolution the modified
+ file has a better chance of being detected for reload. * Added a
+ missing unlock in an off-nominal code path. (closes issue
+ AST-1303) Review: https://reviewboard.asterisk.org/r/3235/
+ ........ Merged revisions 408387 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 408388 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-02-07 19:30 +0000 [r407746] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/frame.h,
+ configs/iax.conf.sample, /: chan_iax2: Block unnecessary control
+ frames to/from the wire. Establishing an IAX2 call between
+ Asterisk v1.4 and v1.8 (or later) results in an unexpected call
+ disconnect. The problem happens because newer values in the enum
+ ast_control_frame_type are not consistent between the branch
+ versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later)
+ using IAX2 2) v1.8 answers and sends a connected line update
+ control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4
+ receives the control frame as an end-of-q (on v1.4
+ AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the
+ receive queue becomes empty. Several things are done by this
+ patch to fix the problem and attempt to prevent it from happening
+ again in the future: * Added a warning at the definition of enum
+ ast_control_frame_type about how to add new control frame values.
+ * Made block sending and receiving control frames that have no
+ reason to go over the wire. * Extended the connectedline iax.conf
+ parameter to also include the redirecting information updates. *
+ Updated the connectedline iax.conf parameter documentation to
+ include a notice that the parameter must be "no" when the peer is
+ an Asterisk v1.4 instance. (closes issue AST-1302) Review:
+ https://reviewboard.asterisk.org/r/3174/ ........ Merged
+ revisions 407678 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 407727 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-25 00:13 +0000 [r406358-406469] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/cel.c: CEL: Protect data structures during reload and
+ shutdown. The CEL data structures need to be protected during a
+ configuration reload and shutdown. Asterisk crashed during a
+ shutdown because CEL events were still in flight and the CEL data
+ structures were already destroyed. * Protected the appset and
+ linkedids ao2 containers using the reload_lock. As a result
+ appset, linkedids, and held objects don't need a lock. * Added
+ NULL checks before use of the appset and linkedids ao2 containers
+ in case the CEL module is already shutdown. * Fixed overloading
+ of the linkedids held objects reference count. During shutdown
+ any held objects would be leaked. * Fixed memory leak of
+ linkedids held objects if the LINKEDID_END is not being tracked.
+ The objects in the linkedids container were not removed if the
+ LINKEDID_END event is not used. * Added access protection to the
+ appset container during the CLI "cel show status" command. * Made
+ CEL config reload not set defaults if the cel.conf file is
+ invalid. (closes issue AST-1253) Reported by: Guenther Kelleter
+ Review: https://reviewboard.asterisk.org/r/3127/ ........ Merged
+ revisions 406417 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 406418 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/manager.c, /: manager: Protect data structures during
+ shutdown. Occasionally, the manager module would get an
+ "INTERNAL_OBJ: bad magic number" error on a "core restart
+ gracefully" command if an AMI connection is established. * Added
+ ao2_global_obj protection to the sessions global container. *
+ Fixed the order of unreferencing a session object in
+ session_destroy(). * Removed unnecessary container traversals of
+ the white/black filters during session_destructor(). (closes
+ issue AST-1242) Reported by: Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/3144/ ........ Merged
+ revisions 406341 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-15 15:27 +0000 [r405536-405578] Matthew Jordan <mjordan at digium.com>
+
+ * main/pbx.c, /: pbx.c: put copy of ast_exten.data on stack to
+ prevent memory corruption During dialplan execution in
+ pbx_extension_helper(), the contexts global read lock prevents
+ link list corruption, but was released with a pointer to the
+ ast_exten and data later used in variable substitution. Instead,
+ this patch removes pbx_substitute_variables() and locates a copy
+ of the ast_exten data on the stack before releasing the lock,
+ where ast_exten could get free'd by another thread performing a
+ module reload. (issue AST-1179) Reported by: Thomas Arimont
+ (issue AST-1246) Reported by: Alexander Hömig Review:
+ https://reviewboard.asterisk.org/r/3055/ ........ Merged
+ revisions 403862 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 403863 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: chan_sip: Hangup transferer/transferee
+ when transfer to Parking fails When performing a SIP transfer to
+ a Park extension, if the Park fails, chan_sip will currently not
+ hang up either the transferer or the transfer target. This
+ results in the channels being orphaned with no thread to service
+ frames, resulting in stuck channels. This patch immediately hangs
+ up the two channels if a Park fails. (closes issue
+ ASTERISK-22834) Reported by: rsw686 Tested by: rsw686 (closes
+ issue ASTERISK-23047) Reported by: Tommy Thompson Tested by:
+ Tommy Thomspon Review: https://reviewboard.asterisk.org/r/3107
+ ........ Merged revisions 405380 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-14 18:50 +0000 [r405488] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_verbose.c, main/asterisk.c, configs/logger.conf.sample,
+ main/cli.c, include/asterisk/logger.h, main/pbx.c,
+ main/manager.c, /, funcs/func_timeout.c, apps/app_dumpchan.c,
+ main/logger.c, UPGRADE.txt: verbosity: Fix performance of console
+ verbose messages. The per console verbose level feature as
+ previously implemented caused a large performance penalty. The
+ fix required some minor incompatibilities if the new rasterisk is
+ used to connect to an earlier version. If the new rasterisk
+ connects to an older Asterisk version then the root console
+ verbose level is always affected by the "core set verbose"
+ command of the remote console even though it may appear to only
+ affect the current console. If an older version of rasterisk
+ connects to the new version then the "core set verbose" command
+ will have no effect. * Fixed the verbose performance by not
+ generating a verbose message if nothing is going to use it and
+ then filtered any generated verbose messages before actually
+ sending them to the remote consoles. * Split the "core set debug"
+ and "core set verbose" CLI commands to remove the per module
+ verbose support that cannot work with the per console verbose
+ level. * Added a silent option to the "core set verbose" command.
+ * Fixed "core set debug off" tab completion. * Made "core show
+ settings" list the current console verbosity in addition to the
+ root console verbosity. * Changed the default verbose level of
+ the 'verbose' setting in the logger.conf [logfiles] section. The
+ default is now to once again follow the current root console
+ level. As a result, using the AMI Command action with "core set
+ verbose" could again set the root console verbose level and
+ affect the verbose level logged. (closes issue AST-1252) Reported
+ by: Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/3114/ ........ Merged
+ revisions 405431 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-09 16:34 +0000 [r405233] Matthew Jordan <mjordan at digium.com>
+
+ * /, apps/app_confbridge.c,
+ apps/confbridge/conf_state_multi_marked.c: app_confbridge: Fix
+ crash caused when waitmarked/marked users leave together When
+ waitmarked users join a ConfBridge, the conference state is
+ transitioned from EMPTY -> INACTIVE. In this state, the users are
+ maintined in a waiting users list. When a marked user joins, the
+ ConfBridge conference transitions from INACTIVE -> MULTI_MARKED,
+ and all users are put onto the active list of users. This process
+ works correctly. When the marked user leaves, if they are the
+ last marked user, the MULTI_MARKED state does the following: (1)
+ It plays back a message to the bridge stating that the leader has
+ left the conference. This requires an unlocking of the bridge.
+ (2) It moves waitmarked users back to the waiting list (3) It
+ transitions to the appropriate state: in this case, INACTIVE
+ However, because it plays the prompt back to the bridge before
+ moving the users and before finishing the state transition, this
+ creates a race condition: with the bridge unlocked, waitmarked
+ users who leave the conference (or are kicked from it) can cause
+ a state transition of the bridge to another state before the
+ conference is transitioned to the INACTIVE state. This causes the
+ state machine to get a bit wonky, often leading to a crash when
+ the MULTI_MARKED state attempts to conclude its processing. This
+ patch fixes this problem: (1) It prevents kicked users from being
+ kicked again. That's just a nicety. (2) More importantly, it
+ fixes the race condition by only playing the prompt once the
+ state has transitioned correctly to INACTIVE. If waitmarked users
+ sneak out during the prompt being played, no harm no foul.
+ Review: https://reviewboard.asterisk.org/r/3108/ (closes issue
+ AST-1258) Reported by: Steve Pitts ........ Merged revisions
+ 405215 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-12-19 16:38 +0000 [r404349] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/db.c, /: astdb: crash in sqlite3 during shutdown When
+ Asterisk is shut down, the astdb_atexit() function releases
+ (finalize) the previously initiated (prepared) SQL statements in
+ sqlite3. Another thread making a subsequent request can cause a
+ crash in sqlite3. This patch eliminates that issue by resetting
+ the statement pointer after it is released/cleared. The sqlite3
+ code detects the null pointer, and aborts the operation cleanly.
+ (closes issue AST-1265) Reported by: Alexander Hömig (closes
+ issue ASTERISK-22350) Reported by: Birger "WIMPy" Harzenetter
+ Review: https://reviewboard.asterisk.org/r/3078/ ........ Merged
+ revisions 404344 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-12-16 17:29 +0000 [r403956] David M. Lee <dlee at digium.com>
+
+ * funcs/func_realtime.c, main/pbx.c, main/tcptls.c,
+ funcs/func_db.c, /, README-SERIOUSLY.bestpractices.txt,
+ configs/asterisk.conf.sample, funcs/func_shell.c,
+ funcs/func_env.c, funcs/func_lock.c, UPGRADE.txt,
+ include/asterisk/pbx.h, main/asterisk.c: security: Inhibit
+ execution of privilege escalating functions This patch allows
+ individual dialplan functions to be marked as 'dangerous', to
+ inhibit their execution from external sources. A 'dangerous'
+ function is one which results in a privilege escalation. For
+ example, if one were to read the channel variable SHELL(rm -rf /)
+ Bad Things(TM) could happen; even if the external source has only
+ read permissions. Execution from external sources may be enabled
+ by setting 'live_dangerously' to 'yes' in the [options] section
+ of asterisk.conf. Although doing so is not recommended. (closes
+ issue ASTERISK-22905) Review:
+ http://reviewboard.digium.internal/r/432/ ........ Merged
+ revisions 403913 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 403917 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-12-16 15:38 +0000 [r403860] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * apps/app_sms.c: app_sms: BufferOverflow when receiving odd length
+ 16 bit message This patch prevents an infinite loop overwriting
+ memory when a message is received into the unpacksms16()
+ function, where the length of the message is an odd number of
+ bytes. (closes issue ASTERISK-22590) Reported by: Jan Juergens
+ Tested by: Jan Juergens
+
+2013-11-04 21:20 +0000 [r402463] Kevin Harwell <kharwell at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: notify dialog info ignores
+ presentation indicator in callerid The presentation indicator in
+ a callerid (e.g. set by dialplan function
+ Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog
+ Info Notifies are generated during extension monitoring. Added a
+ check to make sure the name and/or number presentations on the
+ callee (remote identity) are set to allow. If they are restricted
+ then "anonymous" is used instead. (closes issue AST-1175)
+ Reported by: Thomas Arimont Review:
+ https://reviewboard.asterisk.org/r/2976/ ........ Merged
+ revisions 402450 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-11-01 20:39 +0000 [r402377-402383] Matthew Jordan <mjordan at digium.com>
+
+ * asterisk-11.6.0-summary.html (removed),
+ asterisk-11.6.0-summary.txt (removed): Remove old summaries
+
+ * include/asterisk/pbx.h, res/res_rtp_asterisk.c, main/pbx.c, /,
+ configure, configure.ac: Multiple revisions
+ 396884,400075,400093,401446,401960 ........ r396884 | jbigelow |
+ 2013-08-16 17:45:10 -0500 (Fri, 16 Aug 2013) | 8 lines Add test
+ suite events to indicate when a feature is detected or not These
+ are needed by the bridge test suite tests for them to be able to
+ run against Asterisk 11. Review:
+ https://reviewboard.asterisk.org/r/2751/ ........ r400075 |
+ mjordan | 2013-09-28 16:59:12 -0500 (Sat, 28 Sep 2013) | 16 lines
+ Add check for openSUSE when detecting bfd library In
+ ASTERISK-17842, some additional library checks were added to the
+ configure script so that the bfd library could be found on CentOS
+ and Fedora systems. As it turns out, openSUSE requires an
+ additional library. This patch adds another check to the
+ configure script for openSUSE that will add that library. Review:
+ https://reviewboard.asterisk.org/r/2885/ (closes issue AST-1169)
+ Reported by: Guenther Kelleter ........ Merged revisions 400073
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r400093 | mjordan | 2013-09-28 17:21:37 -0500 (Sat, 28 Sep 2013)
+ | 23 lines res_rtp_asterisk: Correct erroneous lost packet
+ information in RTCP reports RTCP's calculation of the number of
+ lost packets in an RTP stream is based on that stream's sequence
+ number count, the number of received packets, and how many
+ packets we expect to receive. When the SSRC for an RTP stream
+ changes, there can - and almost always will be - a large jump in
+ the next packet's timestamp and sequence number. If we don't
+ reset the number of received packets, sequence number count, and
+ other metrics used by RTCP, the next RR/SR report will use the
+ previous SSRC's values to calculate the lost packet count for the
+ new SSRC - resulting in a very large number of lost packets. This
+ patch modifies res_rtp_asterisk such that, if it detects a SSRC
+ change, it will reset the various values used by the RTCP
+ calculations. From the perspective of RTCP, this appears as a new
+ media stream - which is what it is. Review:
+ https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174)
+ Reported by: Thomas Arimont ........ Merged revisions 400089 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r401446 | mjordan | 2013-10-22 17:42:24 -0500 (Tue, 22 Oct 2013)
+ | 15 lines res_rtp_asterisk: Fix crash when RTCP is not available
+ during SSRC change In r400089, a patch was put in to correct
+ erroneous RTCP statistic resets. Unfortunately, ast_rtp_read can
+ be called on an RTP instance that does not have RTCP information.
+ This patch prevents that crash by only resetting the statistics
+ if we do actually have an RTCP instance. (issue AST-1174) (closes
+ issue ASTERISK-22667) Reported by: John Bigelow ........ Merged
+ revisions 401445 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r401960 | sgriepentrog | 2013-10-25 15:44:40 -0500 (Fri, 25 Oct
+ 2013) | 15 lines pbx.c: fix confused match caller id that deleted
+ exten still in hash This fixes a bug where a zero length callerid
+ match adjacent to a no match callerid extension entry would be
+ deleted together, which then resulted in hashtable references to
+ free'd memory. A third state of the matchcid value has been added
+ to indicate match to any extension which allows enforcing
+ comparison of matchcid on/off without errors. (closes issue
+ AST-1235) Reported by: Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/2930/ ........ Merged
+ revisions 401959 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 396884,400075,400093,401446,401960 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /: SVN properties: Add svnmerge properties for 11
+
+2013-10-22 16:10 +0000 [r401416] bebuild <bebuild at localhost>:
+
+ * / (added): Create branch for Certified Asterisk 11.6.
+
2013-10-21 Asterisk Development Team <asteriskteam at digium.com>
* Asterisk 11.6.0 Released.
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