[asterisk-commits] bebuild: tag 1.8.23.0-rc1 r391247 - /tags/1.8.23.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jun 10 09:39:13 CDT 2013
Author: bebuild
Date: Mon Jun 10 09:39:10 2013
New Revision: 391247
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=391247
Log:
Importing files for 1.8.23.0-rc1 release.
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tags/1.8.23.0-rc1/.version (with props)
tags/1.8.23.0-rc1/ChangeLog (with props)
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--- tags/1.8.23.0-rc1/ChangeLog (added)
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+2013-06-10 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.23.0-rc1 Released.
+
+2013-06-10 14:15 +0000 [r391215] Matthew Jordan <mjordan at digium.com>
+
+ * UPGRADE.txt, apps/app_queue.c, configs/queues.conf.sample: 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
+
+2013-06-10 09:30 +0000 [r391062-391143] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/chan_iax2.c: chan_iax2: nativebridge refactor, missed
+ unlock bridgecallno
+
+ * channels/chan_iax2.c: fix bad edit after conflict resolution
+
+ * 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/
+
+ * 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/
+
+2013-05-31 08:10 +0000 [r390181] Alexandr Anikin <may at telecom-service.ru>
+
+ * 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
+
+2013-05-29 20:10 +0000 [r390044] 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
+
+2013-05-28 17:35 +0000 [r389895] 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)
+
+2013-05-24 11:42 +0000 [r389676] 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)
+
+2013-05-20 17:43 +0000 [r389244] Jason Parker <jparker at digium.com>
+
+ * /: Add doxygen.log to svn:ignore property.
+
+2013-05-15 15:54 +0000 [r388838] 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)
+
+2013-05-15 12:37 +0000 [r388768] 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)
+
+2013-05-13 20:34 +0000 [r388596] 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.
+
+2013-05-13 18:16 +0000 [r388532] 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)
+
+2013-05-13 18:05 +0000 [r388529] 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
+
+2013-05-13 14:24 +0000 [r388477] 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
+
+2013-05-10 22:09 +0000 [r388423-388425] 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
+
+ * utils: Add version.c to list of ignored files in the utils
+ directory.
+
+2013-05-10 20:28 +0000 [r388376] 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)
+
+2013-05-09 03:58 +0000 [r388111] 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/
+
+2013-05-08 07:17 +0000 [r387875] 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/
+
+2013-05-06 15:52 +0000 [r387688] 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.
+
+2013-05-02 17:11 +0000 [r387421] 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.
+
+2013-05-02 07:53 +0000 [r387294-387344] 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/
+
+ * 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/
+
+ * 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)
+
+ * 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)
+
+2013-05-01 21:15 +0000 [r387036-387213] 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)
+
+ * 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)
+
+2013-04-30 13:45 +0000 [r386929] 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/
+
+2013-04-29 23:34 +0000 [r386877] 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)
+
+2013-04-29 08:36 +0000 [r386792] Olle Johansson <oej at edvina.net>
+
+ * CHANGES, apps/app_queue.c: Play periodic prompst for first call
+ in a call queue Review: https://reviewboard.asterisk.org/r/2263/
+
+2013-04-26 21:26 +0000 [r386641-386672] 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)
+
+ * 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)
+
+2013-04-25 02:43 +0000 [r386483] Michael L. Young <elgueromexicano at gmail.com>
+
+ * channels/chan_sip.c: Change Case On Forcerport For Consistency *
+ Change "ForcerPort" to "Forcerport" to match everywhere else it
+ is displayed
+
+2013-04-22 16:10 +0000 [r386256] 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
+
+2013-04-19 15:59 +0000 [r386109] 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/
+
+2013-04-19 05:18 +0000 [r386049] 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.
+
+2013-04-16 23:11 +0000 [r385916] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * 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/
+
+2013-04-15 17:07 +0000 [r385745] Jason Parker <jparker at digium.com>
+
+ * Makefile: Don't unnecessarily rebuild things on every run of
+ 'make'. Review: https://reviewboard.asterisk.org/r/2449/
+
+2013-04-15 14:38 +0000 [r385683] David M. Lee <dlee at digium.com>
+
+ * 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, tests/test_locale.c,
+ include/asterisk/bridging_features.h, doc/Makefile,
+ tests/test_poll.c, res/res_timing_kqueue.c,
+ contrib/realtime/mysql/musiconhold.sql,
+ contrib/realtime/mysql/queue_log.sql, channels/sig_ss7.c,
+ channels/sig_ss7.h, channels/chan_multicast_rtp.c,
+ tests/test_expr.c, apps/app_saycounted.c,
+ contrib/realtime/mysql/voicemail_messages.sql: 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.
+
+2013-04-14 02:58 +0000 [r385633-385636] 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)
+
+ * 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)
+
+2013-04-12 22:34 +0000 [r385551-385593] 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/
+
+ * 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
+ return status. After adding the initial patch to fix this issue,
+ Jaco Kroon (jkroon) added some fixes to memory leaks he had
+ discovered. During review, Walter Doekes (wdoekes) suggested
+ adding a helper function in order to determine if we had a valid
+ configuration or not. This patch does the following: * Creates a
+ helper function to check if the configuration is valid * Adds
+ calls to the new helper function where appropiate * Fixes memory
+ leaks where the code returned without running
+ ast_config_destroy() on the configuration that was loaded (closes
+ issue ASTERISK-21302) Reported by: Jaco Kroon Tested by: Jaco
+ Kroon, Michael L. Young Patches:
+ asterisk-11.3.0-app_voicemail-ast_config-fixes.patch Jaco Kroon
+ (license 5671) asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2443/
+
+2013-04-12 08:46 +0000 [r385402-385429] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/chan_iax2.c: IAX2 defer_full_frames fail to get sent
+ Ensure iax2_process_thread is signalled when a deferred frame is
+ queued to it. (issue ASTERISK-18827) Reported by: alecdavis
+ Tested by: alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2426/
+
+ * channels/chan_iax2.c: IAX2, prevent network thread starting
+ before all helper threads are ready On startup, it's possible for
+ a frame to arrive before the processing threads were ready. In
+ iax2_process_thread() the first pass through falls into
+ ast_cond_wait, should a frame arrive before we are at
+ ast_cond_wait, the signal will be ignored. The result
+ iax2_process_thread stays at ast_cond_wait forever, with deferred
+ frames being queued. Fix: When creating initial idle
+ iax2_process_threads, wait for init_cond to be signalled after
+ each thread is started. (issue ASTERISK-18827) Reported by:
+ alecdavis Tested by: alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2427/
+
+2013-04-10 14:22 +0000 [r385170-385190] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_config_ldap.c: Use LDAP memory management functions
+ instead of Asterisk's When MALLOC_DEBUG is enabled with
+ res_config_ldap, issues (munmap_chunk: invalid pointer errors)
+ can occur as the memory is being allocated with Asterisk's
+ wrappers around malloc/calloc/free/strdup, as opposed to the LDAP
+ library's wrappers. This patch uses the LDAP library's wrappers
+ where appropriate, so that compiling with MALLOC_DEBUG doesn't
+ cause more problems than it solves. Note that the patch listed
+ below was modified slightly for this commit to account for some
+ additional memory allocation/deallocations. (closes issue
+ ASTERISK-17386) Reported by: John Covert Tested by: Andrew Latham
+ patches: issue18789-1.8-r316873.patch uploaded by seanbright
+ (License 5060)
+
+ * channels/chan_sip.c: Fix crash in chan_sip when a core initiated
+ op occurs at the same time as a BYE When a BYE request is
+ processed in chan_sip, the current SIP dialog is detached from
+ its associated Asterisk channel structure. The tech_pvt pointer
+ in the channel object is set to NULL, and the dialog persists for
+ an RFC mandated period of time to handle re-transmits. While this
+ process occurs, the channel is locked (which is good).
+ Unfortunately, operations that are initiated externally have no
+ way of knowing that the channel they've just obtained (which is
+ still valid) and that they are attempting to lock is about to
+ have its tech_pvt pointer removed. By the time they obtain the
+ channel lock and call the channel technology callback, the
+ tech_pvt is NULL. This patch adds a few checks to some channel
+ callbacks that make sure the tech_pvt isn't NULL before using it.
+ Prime offenders were the DTMF digit callbacks, which would crash
+ if AMI initiated a DTMF on the channel at the same time as a BYE
+ was received from the UA. This patch also adds checks on
+ sip_transfer (as AMI can also cause a callback into this
+ function), as well as sip_indicate (as lots of things can queue
+ an indication onto a channel). Review:
+ https://reviewboard.asterisk.org/r/2434/ (closes issue
+ ASTERISK-20225) Reported by: Jeff Hoppe
+
+2013-04-08 23:34 +0000 [r385047] Rusty Newton <rnewton at digium.com>
+
+ * configs/extconfig.conf.sample: Modified the list of keys for the
+ driver backends for sake of sample clarity Added a line showing
+ the mapping of "mysql" to res_config_mysql available in add-ons.
+ We used "mysql" as an example driver key in the sample, but
+ didn't show what module it mapped too. Also added a subtitle
+ above the list of keys for driver backends.
+
+2013-04-08 19:55 +0000 [r385008] Michael L. Young <elgueromexicano at gmail.com>
+
+ * UPGRADE.txt, channels/chan_sip.c: Fix For Not Overriding The
+ Default Settings In chan_sip The initial report was that the
+ "nat" setting in the [general] section was not having any effect
+ in overriding the default setting. Upon confirming that this was
+ happening and looking into what was causing this, it was
+ discovered that other default settings would not be overriden as
+ well. This patch works similar to what occurs in build_peer(). We
+ create a temporary ast_flags structure and using a mask, we
+ override the default settings with whatever is set in the
+ [general] section. In the bug report, the reporter who helped to
+ test this patch noted that the directmedia settings were being
+ overriden properly as well as the nat settings. (closes issue
+ ASTERISK-21225) Reported by: Alexandre Vezina Tested by:
+ Alexandre Vezina, Michael L. Young Patches:
+ asterisk-21225-handle-options-default-prob_1.8_v4.diff.diff
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2386/
+
+2013-04-04 19:31 +0000 [r384779] Michael L. Young <elgueromexicano at gmail.com>
+
+ * contrib/realtime/postgresql/realtime.sql,
+ contrib/realtime/mysql/sippeers.sql, channels/chan_sip.c:
+ Backport Appropiate NAT Setting Cleanup In ASTERISK-20904, the
+ focus was around the changes to NAT that took place in Asterisk
+ 11. Since the report stated that 1.8 was fine, we didn't take a
+ look at 1.8 at the time. While working on ASTERISK-21225, I could
+ see that 1.8 would benefit from having some of those changes
+ applied to it. This patch does the following: * The important
+ part of this patch is that it sets the peer's flags earlier in
+ build_peer so that the code properly uses the peer's flags based
+ on the peer's configuration. * constify req parameter in
+ check_via() * update realtime schemas under the contrib directory
+ to handle properly the NAT settings available in 1.8 as well as
+ to handle the changes made in 11 to make upgrading easier when
+ installing newer versions of Asterisk (closes issue
+ ASTERISK-21243) Reported by: Michael L. Young Patches:
+ asterisk-20904-changes_for_1.8.diff Michael L. Young (license
+ 5026) Review: https://reviewboard.asterisk.org/r/2422/
+
+2013-04-03 20:13 +0000 [r384685] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample: chan_dahdi: Add
+ inband_on_proceeding compatibility option. The new
+ inband_on_proceeding option causes Asterisk to assume inband
+ audio may be present when a PROCEEDING message is received. Q.931
+ Section 5.1.2 says the network cannot assume that the CPE side
+ has attached to the B channel at this time without explicitly
+ sending the progress indicator ie informing the CPE side to
+ attach to the B channel for audio. However, some non-compliant
+ ISDN switches send a PROCEEDING without the progress indicator ie
+ indicating inband audio is available and assume that the CPE
+ device has connected the media path for listening to ringback and
+ other messages. ASTERISK-17834 which causes this issue was
+ dealing with a non-compliant network switch. (closes issue
+ ASTERISK-21151) Reported by: Gianluca Merlo Tested by: rmudgett
+
+2013-04-03 17:05 +0000 [r384640] Matthew Jordan <mjordan at digium.com>
+
+ * funcs/func_channel.c: Update documentation for CHANNEL function
+ Document that you can read/write the 'accountcode' and 'amaflags'
+ on a channel.
+
+2013-04-02 17:33 +0000 [r384544] David M. Lee <dlee at digium.com>
+
+ * Makefile: Fixed spurious rebuilds of func_version.
+ func_version.so was being rebuilt every time, because build.h was
+ changing every build, because of the cleantest dependency that
+ was added in r384410 to fix parallel make bugs. Now build.h will
+ only be created if it does not exist, which was the original
+ behavior of the Makefile.
+
+2013-04-01 13:18 +0000 [r384410] David M. Lee <dlee at digium.com>
+
+ * Makefile: Fix parallel make problems. Occasionally, make -j would
+ fail due to missing includes, or other unusual errors. This was
+ due to the 'cleantest' target, which was designed to force a make
+ clean when some change in the code would cause the typical
+ depedency checking to fail. Several targets in the main Makefile
+ did not depend upon cleantest, hence would run in parallel to it.
+ By adding the dependency, make -j runs happily now. Review:
+ https://reviewboard.asterisk.org/r/2418/
+
+2013-03-29 16:23 +0000 [r384325] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_voicemail.c: app_voicemail: Add blank argument to
+ externnotify if no context argument At least one call to
+ run_externnotify provides a NULL context parameter and because
+ the snprintf statement doesn't account for a NULL context
+ parameter, it simply writes '(null)' to the arguments string
+ instead. This patch makes it write two quotes back to back for
+ that argument instead in the event of a NULL context. (closes
+ issue ASTERISK-18207) Reported by: Barry L. Kline Patches:
+ modified from patch-20130306 uploaded by Karsten Wemheuer
+ (License 5930)
+
+2013-05-17 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.22.0 Released.
+
+2013-05-13 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.22.0-rc2 Released.
+
+ * 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.
+
+ * 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'
[... 44954 lines stripped ...]
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