[svn-commits] bebuild: tag 11.5.0-rc1 r391251 - /tags/11.5.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Jun 10 09:44:36 CDT 2013
Author: bebuild
Date: Mon Jun 10 09:44:34 2013
New Revision: 391251
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=391251
Log:
Importing files for 11.5.0-rc1 release.
Added:
tags/11.5.0-rc1/.lastclean (with props)
tags/11.5.0-rc1/.version (with props)
tags/11.5.0-rc1/ChangeLog (with props)
Added: tags/11.5.0-rc1/.lastclean
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--- tags/11.5.0-rc1/ChangeLog (added)
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+2013-06-10 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.5.0-rc1 Released.
+
+2013-06-10 14:25 +0000 [r391241] Matthew Jordan <mjordan at digium.com>
+
+ * /, configs/queues.conf.sample, UPGRADE.txt, apps/app_queue.c: Add
+ announce-to-first-user option for app_queue In r386792, the
+ ability to play prompts to the first caller in a call queue was
+ added. While this is arguably a bug fix for those who expect the
+ first caller to continue receiving prompts while the agent is
+ dialed, it has the side effect of preventing the first caller
+ from hearing the agent immediately upon bridging. This may not be
+ a problem for those who really want this option, but for those
+ who didn't care whether or not the first caller in queue heard
+ their position, it was an issue. This patch disables the ability
+ for the first caller in the queue to hear prompts and adds a new
+ option, announce-to-first-user, to queues.conf. Those who the
+ behavior can enable it by setting this value to True. Note that
+ if we ever implement the ability to have the prompts be stopped
+ upon bridging, this option can be removed. (closes issue
+ ASTERISK-21782) Reported by: Remi Quezada ........ Merged
+ revisions 391215 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-06-10 09:32 +0000 [r391063-391148] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * /, channels/chan_iax2.c: chan_iax2: nativebridge refactor, missed
+ unlock bridgecallno ........ Merged revisions 391143 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: fix bad edit after conflict resolution
+ ........ Merged revisions 391107 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: IAX2: refactor nativebridge transfer
+ remove triple checking of iaxs[fr->callno]->transferring reduce
+ indentation. Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2602/ ........ Merged
+ revisions 391065 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: IAX2: fix race condition with
+ nativebridge transfers. 1). When touching the bridgecallno, we
+ need to lock it. 2). stop_stuff() which calls
+ iax2_destroy_helper() Assumes the lock on the pvt is already
+ held, when iax2_destroy_helper() is called. Thus we need to lock
+ the bridgecallno pvt before we call
+ stop_stuff(iaxs[fr->callno]->bridgecallno); 3). When evaluating
+ the state of 'callno->transferring' of the current leg, we can't
+ change it to READY unless the bridgecallno is locked. Why, if we
+ are interrupted by the other call leg before 'transferring =
+ TRANSFER_RELEASED', the interrupt will find that it is READY and
+ that the bridgecallno is also READY so Releases the legs. (closes
+ issue ASTERISK-21409) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2594/ ........ Merged
+ revisions 391062 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-31 10:34 +0000 [r390228-390229] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c: remove unnecessary declarations (issue
+ ASTERISK-21800)
+
+ * addons/chan_ooh323.c, /: reject call attempts when gatekeeper is
+ configured but not registered (closes issue ASTERISK-21800)
+ Reported by: Dmitry Melekhov Patches: ASTERISK-21800-1.patch
+ Tested by: Dmitry Melekhov ........ Merged revisions 390181 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 390223 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2013-05-29 20:18 +0000 [r390047] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, /: Fix segfault when dealing with chan_agent
+ channels. Check the returned bridged pointer for NULL to avoid a
+ crash. It looks like chan_agent is returning a NULL pointer when
+ it probably should be returning a pointer to the channel the
+ Agent channel is pretending to be. (closes issue ASTERISK-21793)
+ Reported by: Rodrigo P. Telles Patches:
+ jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Rodrigo P. Telles ........ Merged revisions
+ 390044 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-28 17:43 +0000 [r389896] Jonathan Rose <jrose at digium.com>
+
+ * /, main/slinfactory.c: Fix a memory copying bug in slinfactory
+ which was causing mixmonitor issues. Reported by: Michael Walton
+ Tested by: Jonathan Rose Patches:
+ slinfactory.c.ASTERISK-21799.patch uploaded by Michael Walton
+ (license 6502) (closes issue ASTERISK-21799) ........ Merged
+ revisions 389895 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-24 11:49 +0000 [r389677] Matthew Jordan <mjordan at digium.com>
+
+ * /, main/logger.c: Print all logger messages on shutdown When
+ Asterisk shuts down and shuts down the loggin gsubsystem, any
+ messages currently in flight will not get logged. This patch
+ prevents the loop writing messages from breaking out prematurely,
+ such that all of the messages are logged. (closes issue
+ ASTERISK-21716) Reported by: Corey Farrell patches:
+ logger-process-all-messages.patch uploaded by Corey Farrell
+ (license 5909) ........ Merged revisions 389676 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-24 10:12 +0000 [r389661] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c: Fix several problems caused by multiple
+ line usage with i2004 phones. Reported by: Daniel Bohling,
+ MihaiMircea (closes issue ASTERISK-21061) (closes issue
+ ASTERISK-21120)
+
+2013-05-20 17:43 +0000 [r389245] Jason Parker <jparker at digium.com>
+
+ * /: Add doxygen.log to svn:ignore property. ........ Merged
+ revisions 389244 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-15 15:57 +0000 [r388839] kharwell <kharwell at localhost>:
+
+ * main/lock.c, /: Fix for segfault in __ast_rwlock_destroy with
+ DEBUG_THREADS If DEBUG_THREADS is enabled __ast_rwlock_destroy
+ causes a segfault while trying to access a possible NULL t->track
+ object. A NULL check has been added before trying to access the
+ memory. (closes issue ASTERISK-21724) Reported by: Corey Farrell
+ Fixed by: Corey Farrell Patches: ast_rwlock_destroy-segv.patch
+ uploaded by Corey Farrell (license 5909) ........ Merged
+ revisions 388838 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-15 14:25 +0000 [r388816] Jason Parker <jparker at digium.com>
+
+ * apps/app_voicemail.c: Fix VM snapshot handling for combined
+ INBOX. The snapshot API contains an option that allow for
+ combining of new and old messages within a single snapshot. New
+ messages, however, include options beyond just 'INBOX' - it also
+ includes the Urgent folder. A previous patch that combined INBOX
+ and Urgent accidentally impacted snapshots that attempted to gain
+ messages from just the Old folder. This patch fixes the snapshot
+ gathering such that the API returns the appropriate messages for
+ the folder selected, with and without the combine option. This
+ should make it more clear about what's happening. Review:
+ https://reviewboard.asterisk.org/r/2539/
+
+2013-05-15 12:39 +0000 [r388769] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_srtp.c, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Use srtp_shutdown when available This allows the
+ SRTP library to be shut down properly when the functionality is
+ offered by libsrtp. Review:
+ https://reviewboard.asterisk.org/r/2538/ (closes issue
+ ASTERISK-21719) ........ Merged revisions 388768 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-14 18:55 +0000 [r388700] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/astobj2.h, main/astobj2.c: Make ao2 global
+ objects not always use the debug version of the ao2_ref() calls.
+ The debug versions of ao2_ref() should only be used if REF_DEBUG
+ is enabled so nothing is written to /tmp/refs unexpectedly.
+ (closes issue ASTERISK-21785) Reported by: abelbeck Patches:
+ jira_asterisk_21785_v11.patch (license #5621) patch uploaded by
+ rmudgett Tested by: abelbeck
+
+2013-05-13 21:17 +0000 [r388601-388605] Michael L. Young <elgueromexicano at gmail.com>
+
+ * main/logger.c: Fix Missing CALL-ID When Logging Through Syslog
+ The CALL-ID (ie [C-00000074]) is missing when logging to syslog.
+ This was just an oversight when this feature was added. * Add
+ CALL-IDs when using syslog (closes issue ASTERISK-21430) Reported
+ by: Nikola Ciprich Tested by: Nikola Ciprich, Michael L. Young
+ Patches: asterisk-21430-syslog-callid_trunk.diff by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2526/
+
+ * channels/chan_sip.c: Fix Crash Caused By One-way Audio With
+ auto_* NAT Settings Fix The prior code committed, r385473, failed
+ to take into consideration that not all outgoing calls will be to
+ a peer. My fault. This patch does the following: * Check if there
+ is a related peer involved. If there is, check and set NAT
+ settings according to the peer's settings. * Fix a problem with
+ realtime peers. If the global setting has auto_force_rport set
+ and we issued a "sip reload" while a peer is still registered,
+ the peer's flags for NAT are reset to off. When this happens, we
+ were always setting the contact address of the peer to that of
+ the full contact info that we had. (closes issue ASTERISK-21374)
+ Reported by: jmls Tested by: Michael L. Young Patches:
+ asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2524/
+
+2013-05-13 20:35 +0000 [r388597] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_srtp.c, /: Revert r388529 for now Adding the cleanup
+ function needs some deeper thought since it apparently doesn't
+ exist for all variants of libsrtp. ........ Merged revisions
+ 388596 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-13 19:24 +0000 [r388578] Jonathan Rose <jrose at digium.com>
+
+ * main/pbx.c, /: pbx: Fix lack of cleanup on macrolock and
+ context_table (closes issue ASTERISK-21723) Reported by: Corey
+ Farrell Patches: core-pbx-cleanup.patch uploaded by Correy
+ Farrell (license 5909) ........ Merged revisions 388532 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-13 18:09 +0000 [r388530] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_srtp.c, /: Close libsrtp properly Ensure that libsrtp is
+ shutdown properly when res_srtp is unloaded. (closes issue
+ ASTERISK-21719) Reported by: Corey Farrell Patches:
+ res_srtp-library-shutdown.patch uploaded by Corey Farrell
+ ........ Merged revisions 388529 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-13 14:26 +0000 [r388478] Richard Mudgett <rmudgett at digium.com>
+
+ * main/manager.c, /: Fix SendText AMI action to never return
+ non-zero. AMI actions must never return non-zero unless they
+ intend to close the AMI connection. (Which is almost never.)
+ (closes issue ASTERISK-21779) Reported by: Paul Goldbaum ........
+ Merged revisions 388477 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-10 22:11 +0000 [r388424-388426] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/misdn/isdn_msg_parser.c: Allow mISDN to send PROGRESS
+ messsage. * Made isdn_msg_parser.c build a progress message with
+ the mandatory progress indicator IE. (The mISDNuser NT state
+ machine rejected sending the incomplete message.) Note: The
+ associated mISDN and mISDNuser patches respectively are viewable
+ here: http://svnview.digium.com/svn/thirdparty?view=rev&rev=200
+ http://svnview.digium.com/svn/thirdparty?view=rev&rev=201 (closes
+ issue AST-1153) Reported by: Guenther Kelleter Patches:
+ progress-chan_misdn.diff (license #6372) patch uploaded by
+ Guenther Kelleter progress-misdn.diff (license #6372) mISDN patch
+ uploaded by Guenther Kelleter progress-misdnuser.diff (license
+ #6372) mISDNuser patch uploaded by Guenther Kelleter ........
+ Merged revisions 388425 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * utils, /: Add version.c to list of ignored files in the utils
+ directory. ........ Merged revisions 388423 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-10 20:41 +0000 [r388378] Mark Michelson <mmichelson at digium.com>
+
+ * /, pbx/pbx_dundi.c: Fix memory leak in pbx_dundi pbx_dundi added
+ an io context without removing it. This caused a memory leak when
+ the module was unloaded. (closes ASTERISK-21718) Reported by
+ Corey Farrell Patches: pbx_dundi-ast_io_remove.patch uploaded by
+ Corey Farrell (License #5909) ........ Merged revisions 388376
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-10 11:46 +0000 [r388253] Sean Bright <sean at malleable.com>
+
+ * channels/chan_sip.c: Fix copy/paste error in one-touch-recording
+ implementation.
+
+2013-05-09 04:10 +0000 [r388108-388112] Michael L. Young <elgueromexicano at gmail.com>
+
+ * res/res_rtp_asterisk.c, /: Fix The Payload Being Set On CN
+ Packets And Do Not Set Marker Bit When we send out a CN packet
+ (for instance, in the case of using rtpkeepalives), we are not
+ setting the payload code properly. Also, we are setting the
+ marker bit when we shouldn't be according to RFC 3389, section 4.
+ AST_RTP_CN is not defined by AST_FORMAT codes. Therefore, we
+ should be using ast_rtp_codecs_payload_code() rather than
+ ast_rtp_codecs_payload_lookup(). 11 and trunk already use the
+ appropriate function. * In 1.8, use ast_rtp_codecs_payload_code()
+ * Remove the setting of the marker bit * Fix the debug message by
+ incrementing the seqno after the debug message is set in order to
+ display the correct seqno that was sent out (closes issue
+ ASTERISK-21246) Reported by: Peter Katzmann Tested by: Peter
+ Katzmann, Michael L. Young Patches:
+ asterisk-21246-rtp-cng-payload-error_1.8_v2.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2500/ ........ Merged
+ revisions 388111 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_queue.c: Fix Segfault In app_queue When
+ "persistentmembers" Is Enabled And Using Realtime When the
+ "ignorebusy" setting was deprecated, we added some code to allow
+ us to be compatible with older setups that are still using the
+ "ignorebusy" setting instead of "ringinuse". We set a char
+ *variable with the column name to use, which helps the realtime
+ functions to use the correct column in their SQL queries. When
+ "persistentmembers" is enabled, we are not setting this variable
+ before the realtime functions were called to load members. This
+ results in the variable being NULL and therefore causing a
+ segfault when loading members during the module's process of
+ loading. The solution was to move the code that sets that
+ variable to be before these realtime functions are called during
+ the loading of the module. (closes issue ASTERISK-21738) Reported
+ by: JoshE Tested by: JoshE Patches:
+ asterisk-21738-rt-ringinuse-field-not-set.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2499/
+
+2013-05-08 07:19 +0000 [r387880] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * /, channels/chan_sip.c: chan_sip: NOTIFYs for BLF start queuing
+ up and fail to be sent out after retries fail RFC6665 4.2.2: ...
+ after a failed State NOTIFY transaction remove the subscription
+ The problem is that the State Notify requests rely on the 200OK
+ reponse for pacing control and to not confuse the notify
+ susbsystem. The issue is, the pendinginvite isn't cleared if a
+ response isn't received, thus further notify's are never sent.
+ The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the
+ subscription after failure. (closes issue ASTERISK-21677)
+ Reported by: Dan Martens Tested by: Dan Martens, David Brillert,
+ alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2475/ ........ Merged
+ revisions 387875 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-07 18:29 +0000 [r387823] David M. Lee <dlee at digium.com>
+
+ * res/res_config_pgsql.c, main/manager.c: Minor fixups to Doxygen
+ comments. The \example tags marks an entire file as an example,
+ not a code snippet.
+
+2013-05-06 15:55 +0000 [r387689] Russell Bryant <russell at russellbryant.com>
+
+ * /, apps/app_meetme.c: Make SLA reload more paranoid. Reload
+ support was originally not included for SLA. It was added later,
+ but in a fairly non-traditional way. It basically sets a flag
+ indicating that a reload is pending, and then waits for a time
+ where it thinks everything SLA related is idle and unused, and
+ *then* executes the reload. It does this because the reload
+ process is destructive. It starts by throwing everything away and
+ starting over. There are a number of problems with this approach.
+ One of them is that the check to see if anything in use was
+ incomplete. This patch makes it more complete and thus less
+ likely for a crash to occur during reload processing. However,
+ this approach still has problems so some much more significant
+ reworking of this code will need to come in as a next step. Patch
+ credit and testing by CoreDial, LLC. ........ Merged revisions
+ 387688 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-02 17:15 +0000 [r387422] Matthew Jordan <mjordan at digium.com>
+
+ * utils/Makefile, /: Update utils Makefile to handle r387294 Alec's
+ patch that added the Asterisk version to 'core show locks'
+ angered the items in utils, as they exist somewhat outside of the
+ Asterisk build system. Some day, this Makefile should get nuked
+ from high orbit, but for now, include version.c in its list of
+ stuff to pile in. ........ Merged revisions 387421 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-02 08:09 +0000 [r387295-387345] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
+ Session-Expires: Set timer to correctly expire at (~2/3) of the
+ interval when not the refresher RFC 4028 Section 10 if the side
+ not performing refreshes does not receive a session refresh
+ request before the session expiration, it SHOULD send a BYE to
+ terminate the session, slightly before the session expiration.
+ The minimum of 32 seconds and one third of the session interval
+ is RECOMMENDED. Prior to this asterisk would refresh at 1/2 the
+ Session-Expires interval, or if the remote device was the
+ refresher, asterisk would timeout at interval end. Now, when not
+ refresher, timeout as per RFC noted above. (closes issue
+ ASTERISK-21742) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2488/ ........ Merged
+ revisions 387344 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: chan_sip: Honor Session-Expires in 200OK
+ response when it's a RE-INVITE when asterisk is the refresher.
+ RFC 4028 Section 7.2 "UACs MUST be prepared to receive a
+ Session-Expires header field in a response, even if none were
+ present in the request." What changed After ASTERISK-20787,
+ inbound calls to asterisk with no Session-Expires in the INVITE
+ are now are offered a Session-Expires (1800 asterisk default) in
+ the response, with asterisk as the refresher. Symptom: After 900
+ seconds (asterisk default refresher period 1800), asterisk
+ RE-INVITEs the device, the device may respond with a much lower
+ Session-Expires (180 in our case) value that it is now using.
+ Asterisk ignores this response, as it's deemed both an INBOUND
+ CALL, and a RE-INVITE. After 180 seconds the device times out and
+ sends BYE (hangs up), asterisk is still working with the
+ refresher period of 1800 as it ignored the 'Session Expires: 180'
+ in the previous 200OK response. Fix: handle_response_invite()
+ when 200OK, remove check for outbound and reinvite. (closes issue
+ ASTERISK-21664) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2463/ ........ Merged
+ revisions 387312 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_dahdi.c, /: chan_dahdi: fix lower bound check with
+ -ve integer conversion from a float Lower bound of a 16bit signed
+ int is -32768 not -32767 (closes issue ASTERISK-21744) Reported
+ by: alecdavis Tested by: alecdavis alecdavis (license 585)
+ ........ Merged revisions 387297 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/utils.c: Add Asterisk Version to core show locks Assist
+ with reporting 'core show locks' when submitting bug reports.
+ Example below: =========================== == SVN-branch-1.8-...
+ == Currently Held Locks =========================== (closes issue
+ ASTERISK-21743) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) ........ Merged revisions 387294 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-01 21:17 +0000 [r387038-387216] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_rtp_asterisk.c, /: Clear the DTMF sending digit tracking
+ on off nominal paths In certain situations, when the RTP engine
+ goes to send a DTMF end digit it may be in a situation where the
+ remote address is no longer available, or the digit that was
+ supposed to be sent is invalid. In such cases, we need to clear
+ the RTP counters appropriately. Otherwise, when the RTP source is
+ set again, we'll continue to think that we're in the middle of
+ sending a DTMF digit, which can confuse the remote party
+ (signficantly). (closes issue ASTERISK-21522) Reported by: Corey
+ Farrell patches: rtp_dtmf_process_end.patch uploaded by Corey
+ Farrell (License 5909) ........ Merged revisions 387213 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_sip.c: Prevent crash in 'sip show peers' when the
+ number of peers on a system is large When you have lots of SIP
+ peers (according to the issue reporter, around 3500), the 'sip
+ show peers' CLI command or AMI action can crash due to a poorly
+ placed string duplication that occurs on the stack. This patch
+ refactors the command to not allocate the string on the stack,
+ and handles the formatting of a single peer in a separate
+ function call. (closes issue ASTERISK-21466) Reported by:
+ Guillaume Knispel patches:
+ fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch
+ uploaded by gknispel (License 6492)
+
+ * /, main/features.c: Fix CDR not being created during an
+ externally initiated blind transfer Way back when in the dark
+ days of Asterisk 1.8.9, blind transferring a call in a context
+ that included the 'h' extension would inadvertently execute the
+ hangup code logic on the transferred channel. This was a "bad
+ thing". The fix was to properly check for the softhangup flags on
+ the channel and only execute the 'h' extension logic (and, in
+ later versions, hangup handler logic) if the channel was well and
+ truly dead (Jim). Unfortunately, CDRs are fickle. Setting the
+ softhangup flag when we detected that the channel was leaving the
+ bridge (but not to die) caused some crucial snippet of CDR code,
+ lying in ambush in the middle of the bridging code, to not get
+ executed. This had the effect of blowing away one of the CDRs
+ that is typically created during a blind transfer. While we live
+ and die by the adage "don't touch CDRs in release branches", this
+ was our bad. The attached patch restores the CDR behavior, and
+ still manages to not run the 'h' extension during a blind
+ transfer (at least not when it's supposed to). Thanks to Steve
+ Davies for diagnosing this and providing a fix. Review:
+ https://reviewboard.asterisk.org/r/2476 (closes issue
+ ASTERISK-21394) Reported by: Ishfaq Malik Tested by: Ishfaq
+ Malik, mjordan patches: fix_missing_blindXfer_cdr2 uploaded by
+ one47 (License 5012) ........ Merged revisions 387036 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-30 22:15 +0000 [r387030] Jonathan Rose <jrose at digium.com>
+
+ * main/event.c: Add forgotten event types to event_names array
+
+2013-04-30 13:46 +0000 [r386930] Sean Bright <sean at malleable.com>
+
+ * include/asterisk/utils.h, /: Use the proper lower bound when
+ doing saturation arithmetic. 16 bit signed integers have a range
+ of [-32768, 32768). The existing code was using the interval
+ (-32768, 32768) instead. This patch fixes that. Review:
+ https://reviewboard.asterisk.org/r/2479/ ........ Merged
+ revisions 386929 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-29 23:35 +0000 [r386878] Rusty Newton <rnewton at digium.com>
+
+ * /, sounds/Makefile: Modifying sounds/Makefile to pull down 1.4.24
+ core sounds 1.4.24 core sounds includes a full set of Italian
+ prompts for core sounds and a fix for the missing voicemail
+ prompts in the Russian language. (closes issue ASTERISK-19431)
+ (closes issue ASTERISK-19721) ........ Merged revisions 386877
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-29 08:54 +0000 [r386794] Olle Johansson <oej at edvina.net>
+
+ * /, CHANGES, apps/app_queue.c: Play periodic prompts for first
+ call in a call queue Review:
+ https://reviewboard.asterisk.org/r/2263/ ........ Merged
+ revisions 386792 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-26 21:27 +0000 [r386642-386677] Matthew Jordan <mjordan at digium.com>
+
+ * main/config.c, /: Clean up memory leak in config file on off
+ nominal paths when glob is allowed If a system allows for its
+ usage, Asterisk will use glob to help parse Asterisk .conf files.
+ The config file loading routine was leaking the memory allocated
+ by the glob() routine when the config file was in an unmodified
+ or invalid state. This patch properly calls globfree in those off
+ nominal paths. (closes issue ASTERISK-21412) Reported by: Corey
+ Farrell patches: config_glob_leak.patch uploaded by Corey Farrell
+ (license 5909) ........ Merged revisions 386672 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/features.c: Clean up resources in features on exit This
+ patch cleans up two things features: * It properly unregisters
+ the CLI commands that features registered * It cancels and
+ performs a pthread_join on the created parking thread. This not
+ only properly joins a non-detached thread, but also prevents
+ disposing of the parking lots prior to the parking thread
+ completely exiting. (closes issue ASTERISK-21407) Reported by:
+ Corey Farrell patches: features_shutdown-r2.patch uploaded by
+ Corey Farrell (License 5909) ........ Merged revisions 386641
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-25 03:02 +0000 [r386484-386486] Michael L. Young <elgueromexicano at gmail.com>
+
+ * channels/chan_sip.c: Fix Displaying Symmetric RTP Global Setting
+ * Use comedia_string() to display correctly the symmetric rtp
+ setting when running "sip show settings"
+
+ * /, channels/chan_sip.c: Change Case On Forcerport For Consistency
+ * Change "ForcerPort" to "Forcerport" to match everywhere else it
+ is displayed ........ Merged revisions 386483 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-22 16:30 +0000 [r386286] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, /: Fix crash when AMI redirect action redirects
+ two channels out of a bridge. The two party bridging loops were
+ changing the bridge peer pointers without the channel locks held.
+ Thus when ast_channel_massquerade() tested and used the pointer
+ there is a small window of opportunity for the pointers to become
+ NULL even though the masquerade code has the channels locked.
+ (closes issue ASTERISK-21356) Reported by: William luke Patches:
+ jira_asterisk_21356_v11.patch (license #5621) patch uploaded by
+ rmudgett Tested by: William luke ........ Merged revisions 386256
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-19 22:25 +0000 [r386159] Matthew Jordan <mjordan at digium.com>
+
+ * /, res/res_timing_pthread.c: Prevent res_timing_pthread from
+ blocking callers There were several reports of deadlock when
+ using res_timing_pthread. Backtraces indicated that one thread
+ was blocked waiting for the write to the pipe to complete and
+ this thread held the container lock for the timers. Therefore any
+ thread that wanted to create a new timer or read an existing
+ timer would block waiting for either the timer lock or the
+ container lock and deadlock ensued. This patch changes the way
+ the pipe is used to eliminate this source of deadlocks: 1) The
+ pipe is placed in non-blocking mode so that it would never block
+ even if the following changes someone fail... 2) Instead of
+ writing bytes into the pipe for each "tick" that's fired the pipe
+ now has two states--signaled and unsignaled. If signaled, the
+ pipe is hot and any pollers of the read side filedescriptor will
+ be woken up. If unsigned the pipe is idle. This eliminates even
+ the chance of filling up the pipe and reduces the potential
+ overhead of calling unnecessary writes. 3) Since we're tracking
+ the signaled / unsignaled state, we can eliminate the exta poll
+ system call for every firing because we know that there is data
+ to be read. (closes issue ASTERISK-21389) Reported by: Matt
+ Jordan Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis patches:
+ 0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch
+ uploaded by sruffell (License 5417) (closes issue ASTERISK-19754)
+ Reported by: Nikola Ciprich (closes issue ASTERISK-20577)
+ Reported by: Kien Kennedy (closes issue ASTERISK-17436) Reported
+ by: Henry Fernandes (closes issue ASTERISK-17467) Reported by:
+ isrl (closes issue ASTERISK-17458) Reported by: isrl Review:
+ https://reviewboard.asterisk.org/r/2441/ ........ Merged
+ revisions 386109 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-19 05:18 +0000 [r386006-386051] David M. Lee <dlee at digium.com>
+
+ * main/cli.c, /: cli.c: Properly initialize debug_modules and
+ verbose_modules. This avoids some lock errors on the core set
+ {debug,verbose} commands. ........ Merged revisions 386049 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/message.c: Fix lock errors on startup. In messages.c, there
+ are several places in the code where we create a tmp_tech_holder
+ and pass that into an ao2_find call. Unfortunately, we weren't
+ initializing the rwlock on the tmp_tech_holder, which the hash
+ function was locking. It's apparently harmless, but still not the
+ best code. This patch extracts all that copy/pasted code into two
+ functions, msg_find_by_tech and msg_find_by_tech_name, which
+ properly initialize and destroy the rwlock on the
+ tmp_tech_holder. Review: https://reviewboard.asterisk.org/r/2454/
+
+2013-04-16 23:27 +0000 [r385917-385938] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * res/res_xmpp.c: Distributed Device State broken at sites using
+ res_xmpp or res_jabber where Secuity Advisory AST-2012-015 is
+ inplace res_xmpp was not adding AST_EVENT_IE_CACHABLE to the
+ event as each message came in, then
+ devstate_change_collector_cb() was unable to find
+ AST_EVENT_IE_CACHABLE in the event, so defaulted incorrectly to
+ AST_DEVSTATE_NOT_CACHABLE. (issue ASTERISK-20175) (closes issue
+ ASTERISK-21429) (closes issue ASTERISK-21069) (closes issue
+ ASTERISK-21164) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2452/
+
+ * /, main/devicestate.c, res/res_jabber.c: Distributed Device State
+ broken at sites using res_xmpp or res_jabber where Secuity
+ Advisory AST-2012-015 is inplace res_jabber/res_xmpp were not
+ adding AST_EVENT_IE_CACHABLE to the event as each message came
+ in, then devstate_change_collector_cb() was unable to find
+ AST_EVENT_IE_CACHABLE in the event, so defaulted incorrectly to
+ AST_DEVSTATE_NOT_CACHABLE. (issue ASTERISK-20175) (closes issue
+ ASTERISK-21429) (closes issue ASTERISK-21069) (closes issue
+ ASTERISK-21164) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2452/ ........ Merged
+ revisions 385916 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-15 17:23 +0000 [r385768] Jason Parker <jparker at digium.com>
+
+ * Makefile, /: Don't unnecessarily rebuild things on every run of
+ 'make'. Review: https://reviewboard.asterisk.org/r/2449/ ........
+ Merged revisions 385745 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-15 15:18 +0000 [r385689] David M. Lee <dlee at digium.com>
+
+ * channels/sig_ss7.c, channels/sip/include/security_events.h,
+ contrib/realtime/mysql/queue_log.sql,
+ channels/chan_multicast_rtp.c, channels/sig_ss7.h, /,
+ tests/test_expr.c, apps/app_saycounted.c,
+ channels/sip/security_events.c,
+ contrib/realtime/mysql/voicemail_messages.sql, BSDmakefile,
+ contrib/realtime/mysql/voicemail_data.sql,
+ build_tools/sha1sum-sh, res/res_mutestream.c,
+ configs/res_curl.conf.sample, tests/test_func_file.c,
+ include/asterisk/select.h, res/res_rtp_multicast.c,
+ include/asterisk/bridging_technology.h,
+ include/asterisk/bridging_features.h, tests/test_locale.c,
+ doc/Makefile, tests/test_poll.c,
+ contrib/realtime/mysql/musiconhold.sql, res/res_timing_kqueue.c:
+ Fix the svn:keywords property on several files. Normally I think
+ keyword expansion is silly, but the one time it would have been
+ good, it didn't work because the property had quotes in it. This
+ patch fixes obviously busted svn:keywords properties. ........
+ Merged revisions 385683 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-14 03:00 +0000 [r385634-385637] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_rtp_multicast.c, /: Calculate the timestamp for outbound
+ RTP if we don't have timing information This patch calculates the
+ timestamp for outbound RTP when we don't have timing information.
+ This uses the same approach in res_rtp_asterisk. Thanks to both
+ Pietro and Tzafrir for providing patches. (closes issue
+ ASTERISK-19883) Reported by: Giacomo Trovato Tested by: Pietro
+ Bertera, Tzafrir Cohen patches: rtp-timestamp-1.8.patch uploaded
+ by tzafrir (License 5035) rtp-timestamp.patch uploaded by
+ pbertera (License 5943) ........ Merged revisions 385636 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_alsa.c: Don't attempt to create a voice frame on
+ a read error Prior to this patch, a read error in snd_pcm_readi
+ would still be treated as a nominal result when constructing a
+ voice frame from the expected data. Since the value returned is
+ negative, as opposed to the number of samples read, this could
+ result in a crash. With this patch, we now return a null frame
+ when a read error is detected. Note that the patch on
+ ASTERISK-21329 was modified slightly for this commit, in that we
+ bail immediately on detecting the read error, rather than
+ bypassing the construction of the voice frame. (closes issue
+ ASTERISK-21329) Reported by: Keiichiro Kawasaki patches:
+ chan_alsa.diff uploaded by kawasaki (License 6489) ........
+ Merged revisions 385633 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-12 22:37 +0000 [r385594] Michael L. Young <elgueromexicano at gmail.com>
+
+ * /, apps/app_queue.c: Fix Manager Segfault When app_queue Is
+ Unloaded When app_queue is unloaded, some manager commands are
+ not being unregistered which result in a segfault. This patch
+ corrects this. (closes issue ASTERISK-21397) Reported by: Peter
+ Katzmann, Corey Farrell Tested by: Corey Farrell Patches:
+ asterisk-21397-missing-unreg-manager-cmd_1.8.diff Michael L.
+ Young (license 5026)
+ asterisk-21397-missing-unreg-manager-cmd_11.diff Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2444/
+ ........ Merged revisions 385593 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-12 22:25 +0000 [r385582] Kinsey Moore <kmoore at digium.com>
+
+ * codecs/codec_resample.c: Allow codec_resample to be unloaded
+ Ensure that trans_size is correct to prevent uninitialized
+ entries from preventing reload. (closes issue ASTERISK-21401)
+ Reported by: Corey Farrell Tested by: Corey Farrell Patches:
+ codec_resample-unload.patch uploaded by Corey Farrell
+
+2013-04-12 22:18 +0000 [r385473-385557] Michael L. Young <elgueromexicano at gmail.com>
+
+ * apps/app_voicemail.c, /: Fix app_voicemail Segfault And A Few
+ Memory Leaks The original report was that app_voicemail would
+ crash. This was caused by ast_config_load() returning
+ CONFIG_STATUS_FILEINVALID but no checks being performed for that
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