[svn-commits] bebuild: tag 1.8.18.0-rc1 r374679 - /tags/1.8.18.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Oct 8 15:41:54 CDT 2012
Author: bebuild
Date: Mon Oct 8 15:41:50 2012
New Revision: 374679
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=374679
Log:
Importing files for 1.8.18.0-rc1 release.
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tags/1.8.18.0-rc1/.version (with props)
tags/1.8.18.0-rc1/ChangeLog (with props)
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--- tags/1.8.18.0-rc1/ChangeLog (added)
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+2012-10-08 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.18.0-rc1 Released.
+
+2012-10-05 20:20 +0000 [r374570-374581] dlee <dlee at localhost>:
+
+ * main/manager.c: I've committed too much. Reverting part of
+ r374570.
+
+ * main/manager.c: Improve AMI long line error handling In AMI's
+ parser, when it receives a long line (> 1024 characters), it
+ discards that line, but continues to process the message
+ normally. Typically, this is not a problem because a) who has
+ lines that long and b) usually a discarded line results in an
+ invalid message. But if that line is specifying an optional
+ field, then the message will be processed, you get a 'Response:
+ Success', but things don't work the way you expected them to.
+ This patch changes the behavior when a line-too-long parse error
+ occurs. * Changes the log message to avoid way-too-long (and
+ truncated anyways) log messages * Adds a 'parsing' status flag to
+ Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if,
+ well, a line is too long * Responds with an appropriate error if
+ parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John
+ Bigelow Review: https://reviewboard.asterisk.org/r/2142/
+
+2012-10-05 18:20 +0000 [r374536] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c,
+ channels/misdn/isdn_lib.h, channels/chan_misdn.c: Merged
+ revisions 374515-374535 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500
+ (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode *
+ Made setup_bc() static. Patches: patch1_unused-code.diff (license
+ #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882
+ ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500
+ (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan
+ states Patches: patch2_unused-states.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012)
+ | 16 lines chan_misdn: Remove unnecessary null pointer checks and
+ checks for stack->nt * cleanup_bc() is always called with valid
+ bc (or it would've crashed before). * Value of stack->nt is known
+ in advance at some places. * Rename handle_event() to
+ handle_event_te(), handle_frm() to handle_frm_te(). Patches:
+ patch3_checks.diff (license #6372) patch uploaded by Guenther
+ Kelleter Modified JIRA ABE-2882 ................ r374518 |
+ rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Fix spelling in log messages Patches:
+ patch4_spelling.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2882 ................ r374519 | rmudgett |
+ 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines
+ chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after
+ calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is
+ emptied, cleaned and set not in use, although
+ misdn_lib_send_event() already did the same. This is bad. When
+ it's not in use we are not allowed to touch it. * Moved log
+ message in front of the resulting actions and fixed it to match
+ the case. Patches: patch5_bccleanup.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012)
+ | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up
+ etc., really bad stuff. * Fix return codes of cb_events() for
+ EVENT_SETUP to use caller's cleanup mechanisms. * Move
+ cl_queue_chan() call after bearer check. Patches:
+ patch6_leaks.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2882 ................ r374521 | rmudgett |
+ 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines
+ chan_misdn: We must initialize cause on sending a DISCONNECT. We
+ must initialize cause on sending a DISCONNECT, so it is later
+ correctly indicated to ast_channel in case the answer
+ (RELEASE/RELEASE_COMPLETE) does not include one. Patches:
+ patch7_hangupcause.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2882 ................ r374522 |
+ rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Remove unused code for upqueue Patches:
+ patch8_unused-upqueue.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2882 ................ r374523 |
+ rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Improve debugging (port number, messages fixed, dups
+ removed) Patches: patch9_debug.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012)
+ | 8 lines chan_misdn: Better debug: we can print_bc_info even if
+ there's no ast leg. Patches: patch10_debug-bc-2.diff (license
+ #6372) patch uploaded by Guenther Kelleter Modified. JIRA
+ ABE-2882 ................ r374534 | rmudgett | 2012-10-05
+ 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn:
+ setup_bc() is called too early for an incoming SETUP on TE. This
+ prevents the B channel from being setup for HDLC mode when
+ requested by the bearer capability and config option hdlc=yes. It
+ violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not
+ connect to the channel until a CONNECT ACKNOWLEDGE message has
+ been received." * Call setup_bc() on receipt of
+ CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for
+ PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by
+ Guenther Kelleter Modified. JIRA ABE-2881 ................
+ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012)
+ | 2 lines chan_misdn: Remove some more deadcode. ................
+
+2012-10-04 20:15 +0000 [r374475-374479] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * CHANGES, main/dsp.c, configs/dsp.conf.sample: dsp.c User
+ Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END Instead of
+ a recompile, allow values to be adjusted in dsp.conf For binary
+ distributions allows easy adjustment for wobbly GSM calls, and
+ other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and
+ DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Tested by:
+ alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2144/
+
+ * main/dsp.c: dsp.c fix incorrect DTMF Digit_Duration. it's always
+ short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if hitstobegin=2
+ (issue ASTERISK-16003) Tested by: alecdavis alecdavis (license
+ 585) Review https://reviewboard.asterisk.org/r/2145/
+
+2012-10-04 17:39 +0000 [r374456] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix a regression from direct media ACLs
+ where the directrtpsetup option no longer works. A check was
+ added for direct media ACLs that immediately forbid remote
+ bridging if there was no bridged channel. This caused
+ directrtpsetup to no longer function as it needs this information
+ before bridging actually occurs. Logic has now been adjusted so
+ if there is no bridged channel a remote bridge will still be
+ attempted. (closes issue ASTERISK-20511) Reported by: kristoff
+ Review: https://reviewboard.asterisk.org/r/2146/
+
+2012-10-04 15:25 +0000 [r374426] dlee <dlee at localhost>:
+
+ * main/db.c, res/res_agi.c: Fix DBDelTree error codes for AMI, CLI
+ and AGI The AMI DBDelTree command will return Success/Key tree
+ deleted successfully even if the given key does not exist. The
+ CLI command 'database deltree' had a similar problem, but was
+ saved because it actually responded with '0 database entries
+ removed'. AGI had a slightly different error, where it would
+ return success if the database was unavailable. This came from
+ confusion about the ast_db_deltree retval, which is -1 in the
+ event of a database error, or number of entries deleted
+ (including 0 for deleting nothing). * Adds a Doxygen comment to
+ process_db_keys explaining its retval * Changed some poorly named
+ res variables to num_deleted * Specified specific errors when
+ calling ast_db_deltree (database unavailable vs. entry not found
+ vs. success) * Fixed similar bug in AGI database deltree, where
+ 'Database unavailable' results in successful result (closes issue
+ AST-967) Reported by: John Bigelow Review:
+ https://reviewboard.asterisk.org/r/2138/
+
+2012-10-04 04:39 +0000 [r374365-374384] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * CHANGES, main/dsp.c, configs/dsp.conf.sample: dsp.c User
+ configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values
+ Asterisk's DTMF Specifications are based on AT&T specs, which may
+ not be compatible in other countries. Various countries have
+ different specifications for the maximum power level differences
+ between the DTMF low group and high group of frequencies. Power
+ level difference between frequencies for different
+ Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to
+ 8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian
+ = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1
+ (2006-03) Now allow 4 variables to be individually configured in
+ dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T
+ specifications Add's the following variables to dsp.conf
+ ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51
+ ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98
+ (closes issue ASTERISK-20442) Reported by: tbsky Tested by:
+ tbsky,alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2141/
+
+ * main/dsp.c: _dsp_init: bring inline with trunk preparation for
+ clean merge of DTMF TWIST patch No functional changes, just
+ style. alecdavis (license 585) Reported by: Alec Davis Tested by:
+ alecdavis related https://reviewboard.asterisk.org/r/2141
+
+2012-10-04 02:09 +0000 [r374177-374335] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_jabber.c: Check for presence of buddy in info/dinfo
+ handlers The res_jabber resource module uses the ASTOBJ library
+ for managing its ref counted objects. After calling
+ ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to
+ the object has to be checked to see if the buddy existed. Prior
+ to this patch, the buddy object was not checked for NULL; with
+ this patch in both aji_client_info_handler and aji_dinfo_handler
+ the pointer is checked before used and, if no buddy object was
+ found, the handlers return an error code. This patch does not
+ take the approach that our JID can be used to log in from another
+ resource. If that approach is desired, an improvement could be
+ made to this patch to create the buddy on the fly. This patch
+ seeks only to prevent Asterisk from crashing. Note that multiple
+ people have proposed patches for this issue; the patch being
+ committed here is based on those. (closes issue ASTERISK-19532)
+ Reported by: Karsten Wemheuer Tested by: Byron Clark patches:
+ fix-jabber uploaded by Karsten Wemheuer (license #5930)
+ xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark
+ (license #6157) (closes issue ASTERISK-19557) Reported by:
+ ulugutz
+
+ * main/ccss.c: Destroy the generic_monitors container after the
+ core_instances in ccss For each item in core_instances disposed
+ of in the shutdown of ccss, any generic monitor instances
+ referenced by the objects will be removed from generic_monitors
+ during their destruction. Hilarity ensues if generic_monitors no
+ longer exists. Thanks to the Asterisk Test Suite's generic_ccss
+ test for complaining loudly when it ran into this.
+
+ * main/asterisk.c: Ensure Shutdown AMI event is still fired during
+ Asterisk shutdown Richard pointed out that having the manager
+ dispose of itself gracefully during shutdown meant that the
+ Shutdown event will no longer get fired. This patch moves the AMI
+ event just prior to running the atexit callbacks.
+
+ * main/event.c, main/taskprocessor.c, res/res_musiconhold.c,
+ main/cel.c, main/indications.c, main/channel.c, main/data.c,
+ main/pbx.c, main/manager.c, main/ccss.c, main/features.c: Fix a
+ variety of ref counting issues This patch resolves a number of
+ ref leaks that occur primarily on Asterisk shutdown. It adds a
+ variety of shutdown routines to core portions of Asterisk such
+ that they can reclaim resources allocate duringd initialization.
+ Review: https://reviewboard.asterisk.org/r/2137
+
+2012-10-01 16:45 +0000 [r374108] Sean Bright <sean at malleable.com>
+
+ * tests/test_db.c, apps/app_queue.c, main/db.c,
+ include/asterisk/astdb.h: app_queue: Support persisting and
+ loading of long member lists. Greenlight in #asterisk brought up
+ that he was receiving an error message "Could not create
+ persistent member string, out of space" when running app_queue in
+ Asterisk 10. dump_queue_members() made an assumption that 8K
+ would be enough to store the generated string, but with queues
+ that have large member lists this is not always the case. This
+ patch removes the limitation and uses ast_str instead of a fixed
+ sized buffer. The complicating factor comes from the fact that
+ ast_db_get requires a buffer and buffer size argument, which
+ doesn't let us pull back more than what we pass in, so I
+ introduced a new ast_db_get_allocated() which returns an
+ ast_strdup()'d copy of the value from astdb. As an aside, I did
+ some testing on the maximum size of data that we can store in the
+ BDB library we distribute and was able to store a 10MB string and
+ retrieve it with no problems, so I feel this is a safe patch.
+ Review: https://reviewboard.asterisk.org/r/2136/
+
+2012-09-28 19:03 +0000 [r374032] Jonathan Rose <jrose at digium.com>
+
+ * res/res_jabber.c: res_jabber: Remove CLI command 'jabber test'
+ The opinion of development was that it is both improper to have
+ Matt's personal email address used in the source and that the
+ command wouldn't be useful without it. (closes issue AST-467)
+ Reported by: Malcolm Davenport
+
+2012-09-28 12:14 +0000 [r373989] Joshua Colp <jcolp at digium.com>
+
+ * res/res_agi.c: Update documentation to make it explicit that
+ "stream file" will not restart musiconhold. (issue
+ ASTERISK-17367) Reported by: oej
+
+2012-09-27 22:08 +0000 [r373945] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_senddtmf.c: Fix SendDTMF crash and channel reference
+ leak using channel name parameter. The SendDTMF channel name
+ parameter has two issues. 1) Crashes if the channel name does not
+ exist. 2) Leaks a channel reference if the channel is the current
+ channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF
+ documentation. * Renamed app to senddtmf_name and tweaked the
+ type.
+
+2012-09-27 16:49 +0000 [r373878-373909] Joshua Colp <jcolp at digium.com>
+
+ * main/loader.c: loader: Ensure dependent modules are properly
+ initialized. If an Asterisk module specifies a dependency in
+ ast_module_info.nonoptreq, it is possible for Asterisk to skip
+ calling the modules's .load function. Asterisk was loading and
+ linking the module via load_dynamic_module() but was not adding
+ the module to the resource_heap. Therefore the module was not
+ initialized based on it's priority along with the other modules
+ in the heap. Now use load_resource() instead of
+ load_dynamic_module() for non-optional requirement. This will add
+ the module to the resource_heap so the module can be properly
+ initialized in the correct order. This is required if there are
+ any module global data structures initialized in the .load()
+ callback for the module on platforms which do not support weak
+ references. (issue ASTERISK-20439) Reported by: sruffell Patches:
+ 0001-loader-Ensure-dependent-modules-are-properly-initial.patch
+ uploaded by sruffell (license 5417)
+
+ * channels/chan_local.c: Fix an issue where Local channels dialed
+ by app_queue are considered in use immediately. The chan_local
+ channel driver returns a device state of in use even if a created
+ Local channel has not yet been dialed. This fix changes the logic
+ to return a state of not in use until the channel itself has been
+ dialed. (closes issue ASTERISK-20390) Reported by: tim_ringenbach
+ Review: https://reviewboard.asterisk.org/r/2116/
+
+2012-09-26 21:11 +0000 [r373848] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Move handling of 408 response so there is no
+ misleading warning message. (closes issue ASTERISK-20060)
+ Reported by: Walter Doekes
+
+2012-09-26 18:04 +0000 [r373815] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_meetme.c: Fixed meetme tab completion and command
+ documentation. * Removed unnecessary case sensitivity in meetme
+ list, lock, unlock, mute, unmute, and kick commands. * Separated
+ meetme lock/unlock, mute/unmute, and kick commands into their own
+ registered commands to simplify tab completion and parameter
+ checking. meetme_lock_cmd(), meetme_mute_cmd(), and
+ meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue
+ AST-1006) Reported by: John Bigelow Tested by: rmudgett
+
+2012-09-25 23:07 +0000 [r373735-373773] Mark Michelson <mmichelson at digium.com>
+
+ * main/say.c: Fix saying of date in Dutch. The Dutch say the date
+ before the month. (closes issue ASTERISK-20353) Reported by: Teun
+ Ouwehand
+
+ * configs/agents.conf.sample, channels/chan_agent.c: Remove dead
+ code and documentation for nonexistent feature. multiplelogin was
+ removed from chan_agent back in 1.6.0 when AgentCallbackLogin()
+ was removed. (closes issue AST-948) reported by Steve Pitts
+
+ * apps/app_voicemail.c: Fix error where improper IMAP greetings
+ would be deleted. (closes issue ASTERISK-20435) Reported by:
+ fhackenberger Patches: asterisk-20435-imap-del-greeting.diff
+ uploaded by Michael L. Young (License #5026) (with suggested
+ modification made by me)
+
+2012-09-25 20:10 +0000 [r373705] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_local.c: Fix T.38 support when used with chan_local
+ in between. Users of the T.38 API can indicate
+ AST_T38_REQUEST_PARMS on a channel to request that the channel
+ indicate a T.38 negotiation with the parameters present on the
+ channel. The return value of this indication is expected to be
+ AST_T38_REQUEST_PARMS upon success but with chan_local involved
+ this could never occur. This fix changes chan_local to always
+ return AST_T38_REQUEST_PARMS for this situation. If the
+ underlying channel technology on the other side does not support
+ T.38 this would have been determined ahead of time using
+ ast_channel_get_t38_state and an indication would not occur.
+ (closes issue ASTERISK-20229) Reported by: wdoekes Patches:
+ ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review:
+ https://reviewboard.asterisk.org/r/2070/
+
+2012-09-25 19:32 +0000 [r373666-373702] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_rtp_asterisk.c: Fix an issue where media would not flow
+ for situations where the legacy STUN code is in use. The STUN
+ packets should *not* be blocked by strict RTP. (closes issue
+ ASTERISK-20415) Reported-by: Michele Cicciotti Patch-by: Josh
+ Colp (trunk r369817)
+
+ * apps/app_queue.c: "show" completion option for "queue" shouldn't
+ appear twice When tab-completing CLI commands starting with
+ "queue", "show" appeared twice in the list due to the way that
+ Asterisk's tab completion functions and the order in which the
+ commands were registered. The registration order has been altered
+ to resolve this issue. (closes issue AST-940) Reported-by: Steve
+ Pitts
+
+2012-09-25 17:21 +0000 [r373652] Terry Wilson <twilson at digium.com>
+
+ * configs/sip.conf.sample, channels/sip/include/sip.h,
+ channels/chan_sip.c: Properly handle UAC/UAS roles for SIP
+ session timers The SIP session timer mechanism contains a
+ mandatory 'refresher' parameter (included in the Session-Expires
+ header) which is used in the session timer offer/answer signaling
+ within a SIP Invite dialog. It looks like asterisk is
+ interpreting the uac resp. uas role only as the initial role of
+ client and server (caller is uac, callee is uas). The standard
+ rfc 4028 however assigns the client role to the ((RE)-Invite)
+ requester, the server role to the ((RE)-Invite) responder. This
+ patch has Asterisk track the actual refresher as "us" or "them"
+ as opposed to relying on just the configured "uas" or "uac"
+ properties. (closes issue AST-922) Reported by: Thomas Airmont
+ Review: https://reviewboard.asterisk.org/r/2118/
+
+2012-09-25 17:18 +0000 [r373618-373640] Richard Mudgett <rmudgett at digium.com>
+
+ * codecs/ilbc/iLBC_decode.c, codecs/ilbc/iLBC_encode.c: Fix
+ valgrind found memcpy issues in codec_ilbc. Valgrind found
+ codec_ilbc using memcpy instead of memmove for overlapping memory
+ blocks. (issue ASTERISK-19890) (closes issue ASTERISK-20231)
+ Reported by: Walter Doekes Patches: ASTERISK-20231.patch (license
+ #5674) patch uploaded by Walter Doekes
+
+ * codecs/Makefile: Make rebuild GSM, ilbc, or lpc10 codecs if the
+ respective sources change.
+
+2012-09-25 16:15 +0000 [r373617] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Set Quality of Service for video
+ rtp instance (closes issue ASTERISK-20201) Reported by: ddkprog
+ Patches: chan_sip.c.diff uploaded by ddkprog (license 6008)
+
+2012-09-25 13:27 +0000 [r373578] Kinsey Moore <kmoore at digium.com>
+
+ * configs/res_odbc.conf.sample: Fix documentation for default
+ username in res_odbc This was previously stated to be "root", but
+ is actually the name of the context if unspecified. (closes issue
+ ASTERISK-20258) Reported by: Stefan x
+
+2012-09-25 11:58 +0000 [r373532-373550] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_multicast.c: Fix an issue where a caller to ast_write
+ on a MulticastRTP channel would determine it failed when in
+ reality it did not. When sending RTP packets via multicast the
+ amount of data sent is stored in a variable and returned from the
+ write function. This is incorrect as any non-zero value returned
+ is considered a failure while a return value of 0 is success. For
+ callers (such as ast_streamfile) that checked the return value
+ they would have considered it a failure when in reality nothing
+ went wrong and it was actually a success. The write function for
+ the multicast RTP engine now returns -1 on failure and 0 on
+ success, as it should. (closes issue ASTERISK-17254) Reported by:
+ wybecom
+
+ * channels/chan_sip.c: Add missing checks that I neglected. The SIP
+ technology and SIP info technology should be considered equal.
+ (closes issue ASTERISK-20409) Reported by: michele cicciotti
+ privatewave
+
+2012-09-24 22:15 +0000 [r373504] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_rtp_asterisk.c: Revert change to res_rtp_asterisk
+ committed in r373236 (1.8) The change committed in r373236
+ attempted to account for endpoints that increased their RTP
+ timestamp in DTMF end of event re-transmissions. This change
+ attempted to make Asterisk continue to work with endpoints that
+ failed to follow the RFC while maintaining the fix that allowed
+ for out of order DTMF to be handled. Unfortunately, there is no
+ free lunch, and this patch broke any system that sent DTMF
+ immediately after an RTP session was established or when an SSRC
+ is updated. As such, that patch is being reverted for the
+ previous behavior. Endpoints that erroneously increase the RTP
+ timestamp in DTMF end of event packets will not work properly
+ with Asterisk. (issue ASTERISK-20424)
+
+2012-09-24 22:09 +0000 [r373500] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: Be consistent, send From: "Anonymous"
+ <sip:anonymous at anonymous.invalid> When setting
+ CALLERID(pres)=unavailable in the dialplan, the From header in
+ the SIP message contains "Anonymous"
+ <sip:Anonymous at anonymous.invalid>. For consistency, Asterisk
+ should use a lowercase a in the userpart of the URI. * Make the
+ From header use a lowercase A in the userpart of the anonymous
+ URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola
+ Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383)
+ patch uploaded by Antti Yrjola
+
+2012-09-24 20:57 +0000 [r373467] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_mixmonitor.c, funcs/func_audiohookinherit.c:
+ func_audiohookinherit: Document some missed sources. This patch
+ also mentions that AUDIOHOOK_INHERIT can be used to transfer
+ MixMonitor audiohooks. There is also wiki that addresses
+ audiohooks and the use of AUDIOHOOK_INHERIT at the following
+ link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks
+ (closes issue ASTERISK-18220) Reported by: Ishfaq Malik
+
+2012-09-24 19:15 +0000 [r373438] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix a deadlock caused by a race condition
+ between removing a hint and reloading the dialplan and
+ subscribing to the removed hint. If conditions were right it was
+ possible for both the PBX core and chan_sip to deadlock by both
+ having a lock that the other wants. In the case of the PBX core
+ it had the contexts lock and wanted a SIP dialog lock, while in
+ the case of chan_sip it had the SIP dialog lock and wanted the
+ contexts lock. This fix unlocks the SIP dialog before getting the
+ extension state so that the other thread will not block on trying
+ to lock it. Once the extension state is retrieved the SIP dialog
+ is locked again and life carries on. As the SIP dialog is
+ reference counted it is not possible for it to go away after
+ unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins
+
+2012-09-24 15:40 +0000 [r373424] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: Fix potential reentrancy problems in
+ chan_sip. Asterisk v1.8 and later was not as vulnerable to this
+ issue. * Made find_call() lock each private as it processes the
+ found dialogs. (Primary cause of ABE-2876) * Made the other
+ functions that traverse the dialogs container lock each private
+ as it examines them. * Fix race condition in sip_call() if the
+ thread that sent the INVITE is held up long enough for a response
+ to be processed. The p->initid for the INVITE retransmission
+ could be added after it was canceled by the response processing.
+ * Made __sip_destroy() clean up resource pointers after freeing.
+ This is primarily defensive in case someone has a stale private
+ pointer. * Removed redundant memset() in reqprep(). The call to
+ init_req() already does the memset() and is the first reference
+ to req in reqprep(). * Removed useless set of req.method in
+ transmit_invite(). The calls to initreqprep() and reqprep() have
+ to do this because they memset() the req. JIRA ABE-2876
+ .......... Merged -r373423 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+
+2012-09-21 19:00 +0000 [r373298-373342] Jonathan Rose <jrose at digium.com>
+
+ * channels/iax2-provision.c: iax2-provision: Fix improper return on
+ failed cache retrieval (closes issue ASTERISK-20337) reported by:
+ John Covert Patches: iax2-provision.c.patch uploaded by John
+ Covert (license 5512)
+
+ * apps/app_queue.c: app_queue: Make queue reload members and
+ variants of that work Prior to this patch, 'queue reload members'
+ cli command did not work at all. This also affects the manager
+ function 'QueueReload' when supplied with the 'members: yes'
+ field. (closes issue AST-956) Reported by: John Bigelow
+
+2012-09-20 19:12 +0000 [r373242] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_meetme.c: Fix incorrect MeetME conference bridge
+ reference count decrementing and sometimes premature destruction.
+ When using the 'e' or 'E' option to MeetMe the configured
+ conference bridges are loaded and examined to see if any are
+ empty. If no conference bridges are empty the caller is prompted
+ to enter the number of one. This operation left around a pointer
+ to the last created conference bridge still containing
+ participants. When the caller that was not able to find any empty
+ conference bridge hung up this pointer was disposed of and the
+ reference count of the conference bridge decremented. If there
+ was only a single participant in the conference bridge it was
+ ultimately destroyed prematurely. (closes issue AST-994) Reported
+ by: John Bigelow
+
+2012-09-20 18:41 +0000 [r373236] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_rtp_asterisk.c: When processing RFC 2833 DTMF, accomodate
+ increasing timestamps in End events While endpoints should not be
+ changing the source timestamp between DTMF event packets, the
+ fact is there exists those endpoints that do exactly that. To
+ work around this, we absorb timestamps within the expected
+ re-transmit period. Note that this period only affects End of
+ Event packets, so it should not prevent the detection of new DTMF
+ digits that happen to arrive right on top of each other. (closes
+ issue ASTERISK-20424) Reported by: Vladimir Mikhelson Tested by:
+ mjordan, Vladimir Mikhelson Review:
+ https://reviewboard.asterisk.org/r/2124
+
+2012-09-19 16:02 +0000 [r373165] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix a regression where direct media was not
+ permitted for calls using SIP INFO DTMF. A change was committed
+ to fix direct media ACL support. This change wrongly assumed that
+ only a single channel technology structure exists for chan_sip.
+ This is in fact false as a second exists for calls using SIP INFO
+ DTMF. The code which performs direct media ACL checking now
+ checks for both the non-INFO DTMF and INFO DTMF channel
+ technology structures. (closes issue ASTERISK-20409) Reported by:
+ michele cicciotti privatewave
+
+2012-09-18 20:12 +0000 [r373131] Sean Bright <sean at malleable.com>
+
+ * main/manager.c: Don't crash when passing a NULL message to
+ __astman_get_header. Before this commit, __astman_get_header
+ would blindly dereference the passed in 'struct message *' to
+ traverse the header list. There are cases, however, such as
+ '*CLI> sip qualify peer foo' where the message pointer is NULL,
+ so we need to check for that.
+
+2012-09-15 00:13 +0000 [r373090] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_ss7.c: Made companding law for SS7 calls only
+ determined by SS7 signaling type. For SS7, the companding law for
+ a call was chosen inconsistently depending upon ss7type (ITU vs
+ ANSI) and the DAHDI companding default (T1 vs E1). For incoming
+ calls, the companding law was determined by ss7type. For outgoing
+ calls, the companding law was determined by the DAHDI default.
+ With the wrong combination you would get A-law/u-law conflicts.
+ An A-law/u-law conflict sounds like bad static on the line. SS7
+ ITU signaling with E1 line: ok SS7 ITU signaling with T1 line:
+ noise SS7 ANSI signaling with E1 line: noise SS7 ANSI signaling
+ with T1 line: ok * Fix the companding law used to be determined
+ by the SS7 signaling type only.
+
+2012-09-14 19:07 +0000 [r373061] Matthew Jordan <mjordan at digium.com>
+
+ * main/ssl.c, main/tcptls.c, channels/chan_sip.c: Resolve memory
+ leaks in TLS initialization and TLS client connections This patch
+ resolves two sources of memory leaks when using TLS in Asterisk:
+ 1) It removes improper initialization (and multiple
+ re-initializations) of portions of the SSL library. Asterisk
+ calls SSL_library_init and SSL_load_error_strings during SSL
+ initialization; collectively this obviates the need for calling
+ any of the following during initialization or client connection
+ handling: * ERR_load_crypto_strings (handled by
+ SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for
+ SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for
+ SSL_library_init) 2) Failure to completely clean up all memory
+ allocated by Asterisk and by the SSL library for TLS clients.
+ This included not freeing the SSL_CTX object in the SIP channel
+ driver, as well as not clearing the error stack when the TLS
+ client exited. Note that these memory leaks were found by Thomas
+ Arimont, and this patch was essentially written by him with some
+ minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont
+ Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas
+ Arimont (license 5525) Review:
+ https://reviewboard.asterisk.org/r/2105
+
+2012-09-13 18:39 +0000 [r373024] dlee <dlee at localhost>:
+
+ * include/asterisk/channel.h, main/channel.c: Fix timeouts for
+ ast_waitfordigit[_full]. ast_waitfordigit_full would simply pass
+ its timeout to ast_waitfor_nandfds, expecting it to decrement the
+ timeout by however many milliseconds were waited. This is a
+ problem if it consistently waits less than 1ms. The timeout will
+ never be decremented, and we wait... FOREVER! This patch makes
+ ast_waitfordigit_full manage the timeout itself. It maintains the
+ previously undocumented behavior that negative timeouts wait
+ forever. (closes issue ASTERISK-20375) Reported by: Mark
+ Michelson Tested by: Mark Michelson Review:
+ https://reviewboard.asterisk.org/r/2109/
+
+2012-09-13 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.17.0-rc1 Released.
+
+2012-09-12 15:42 +0000 [r372959] Matthew Jordan <mjordan at digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h: Constify
+ __ao2_ref_debug in astobj2 When REF_DEBUG is enabled in certain
+ files - most notably ccss.c - the 'tag' parameter passed to
+ __ao2_ref_debug will be a const char *. The function currently
+ expects that parameter to not be const. This causes a warning
+ when compiling, as the const qualifier is being discarded. With
+ dev-mode enabled, this prevents compiling Asterisk. This patch
+ makes __ao2_ref_debug's tag and file parameters const. (closes
+ issue ASTERISK-20408) Reported by: mjordan
+
+2012-09-12 14:51 +0000 [r372932] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Add channel name to a warning to make
+ debugging easier. The "autodestruct with owner in place" message
+ is typically indicative of a channel reference leak. Printing out
+ the name of the channel in the message may be helpful when trying
+ to debug the issue.
+
+2012-09-11 22:11 +0000 [r372902] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_local.c: chan_local: Switch from using a random 4
+ digit hex identifier to unique id Changes chan_local channels to
+ use an 8 digit hex identifier generated atomically and
+ sequentially in order to eliminate the chance of having multiple
+ channels with the same name during high call volume situations.
+ (issue ASTERISK-20318) Reported by: Dan Cropp Review:
+ https://reviewboard.asterisk.org/r/2104/
+
+2012-09-11 15:26 +0000 [r372840] Mark Michelson <mmichelson at digium.com>
+
+ * main/features.c: Fix bad channel application data reference. When
+ channels get bridged due to an AMI bridge action or a DTMF
+ attended transfer, the two channels that get bridged have their
+ application data pointing to the other channel's name. This means
+ that if one channel is hung up but the other moves on, it means
+ that the channel that moves on will have its application data
+ pointing at freed memory. (issue ASTERISK-20335) Reported by:
+ aragon
+
+2012-09-10 20:53 +0000 [r372804] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_iax2.c: Ensure iax2 debug output is displayed when
+ expected When IAX2 debug was changed from iax_showframe to
+ iax_outputframe, some instances were missed (or added afterward).
+ This was causing debug output to not be displayed when expected.
+ (closes issue ASTERISK-20338) Reported-by: John Covert Patch-by:
+ John Covert
+
+2012-09-10 18:35 +0000 [r372765] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_meetme.c: app_meetme: Document that 'p' option will
+ continue in dialplan. (closes issue AST-991) Reported by John
+ Bigelow
+
+2012-09-10 18:31 +0000 [r372763] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: Warn on CLI when UDPTL init fails This adds
+ a CLI warning when a SDP offer is rejected due to UDPTL
+ initialization failure. Previously, there was no indication of
+ the reason for offer rejection in this case. (closes issue
+ ASTERISK-20357) Reported-by: Francesco Usseglio Gaudi
+
+2012-09-10 17:07 +0000 [r372736] Jonathan Rose <jrose at digium.com>
+
+ * main/channel.c: Masquerade: Retain parkinglot settings made by
+ CHANNEL function. Prior to this patch, the user would have a
+ parkinglot set on a channel that was parked and when the channel
+ was retrieved, any attempt by that channel to park would simply
+ use the default. This patch makes parkinglot values set in this
+ way be retained through the masquerade. (closes issue AST-990)
+ Reported by: Nick Huskinson Patches:
+ masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose
+ (license 6182)
+
+2012-09-09 01:19 +0000 [r372709] Matthew Jordan <mjordan at digium.com>
+
[... 41775 lines stripped ...]
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