[asterisk-commits] lmadsen: tag 1.4.32-rc1 r261543 - /tags/1.4.32-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu May 6 09:36:56 CDT 2010
Author: lmadsen
Date: Thu May 6 09:36:51 2010
New Revision: 261543
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=261543
Log:
Importing files for 1.4.32-rc1 release.
Added:
tags/1.4.32-rc1/.lastclean (with props)
tags/1.4.32-rc1/.version (with props)
tags/1.4.32-rc1/ChangeLog (with props)
Added: tags/1.4.32-rc1/.lastclean
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+2010-05-06 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.32-rc1 Released
+
+2010-05-05 16:42 +0000 [r261274] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_sip.c: Registration fix for SIP realtime. Make sure
+ realtime fields are not empty. (closes issue #17266) Reported by:
+ Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
+ Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
+ https://reviewboard.asterisk.org/r/643/
+
+2010-05-04 23:47 +0000 [r261093-261094] Tilghman Lesher <tlesher at digium.com>
+
+ * main/channel.c: Add a tiny corner case to the previous commit
+
+ * main/channel.c: Protect against overflow, when calculating how
+ long to wait for a frame. (closes issue #17128) Reported by:
+ under Patches: d.diff uploaded by under (license 914)
+
+2010-05-04 18:46 +0000 [r260923] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Voicemail transfer to operator should occur
+ immediately, not after main menu. There were two scenarios in the
+ advanced options that while using the operator=yes and review=yes
+ options, the transfer occurred only after exiting the main menu
+ (after sending a reply or leaving a message for an extension).
+ Now after the audio is processed for the reply or message the
+ transfer occurs immediately as expected. ABE-2107 ABE-2108
+
+2010-05-04 17:40 +0000 [r260887] tringenbach <tringenbach at localhost>:
+
+ * README-SERIOUSLY.bestpractices.txt: Fix FILTER() examples to work
+ in 1.4 Review: https://reviewboard.asterisk.org/r/644/
+
+2010-05-04 15:49 +0000 [r260801] Jason Parker <jparker at digium.com>
+
+ * build_tools/make_build_h: Fix fallout from removing from
+ configure script. Pointed out by philipp64 on #asterisk-dev
+
+2010-05-03 16:54 +0000 [r260661-260662] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * Makefile: Should have removed /usr/lib/ part. Thanks Qwell.
+
+ * Makefile: non-root make install PREFIX=/tmp fails. Prepend libdir
+ when executing mkpkgconfig allowing non-root installs to work.
+ (closes issue #17268) Reported by: pabelanger Patches:
+ issue17268.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger
+
+2010-05-03 14:57 +0000 [r260569] Leif Madsen <lmadsen at digium.com>
+
+ * doc/HOWTO_collect_debug_information.txt: Minor typo pointed out
+ by pabelanger on IRC.
+
+2010-04-30 22:22 +0000 [r260434] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Ensure channel state is not incorrectly
+ set in the case of a very early answer. The needringing bit was
+ being read in dahdi_read after answering thereby setting the
+ state to ringing from up. This clears needringing upon answering
+ so that is no longer possible. (closes issue #17067) Reported by:
+ tzafrir Patches: needringing.diff uploaded by tzafrir (license
+ 46)
+
+2010-04-30 20:08 +0000 [r260345] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_musiconhold.c: Fix potential crash from race condition
+ due to accessing channel data without the channel locked. In
+ res_musiconhold.c, there are several places where a channel's
+ stream's existence is checked prior to calling ast_closestream on
+ it. The issue here is that in several cases, the channel was not
+ locked while checking the stream. The result was that if two
+ threads checked the state of the channel's stream at
+ approximately the same time, then there could be a situation
+ where both threads attempt to call ast_closestream on the
+ channel's stream. The result here is that the refcount for the
+ stream would go below 0, resulting in a crash. I have added
+ proper channel locking to res_musiconhold.c to ensure that we do
+ not try to check chan->stream without the channel locked. A
+ Digium customer has been using this patch for several weeks and
+ has not had any crashes since applying the patch. ABE-2147
+
+2010-04-29 22:11 +0000 [r260195] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: DTMF CallerID detection problems. The code
+ handling DTMF CallerID drops digits on long CallerID numbers and
+ may timeout waiting for the first ring with shorter numbers. The
+ DTMF emulation mode was not turned off when processing DTMF
+ CallerID. When the emulation code gets behind in processing the
+ DTMF digits it can skip a digit. For shorter numbers, the timeout
+ may have been too short. I increased it from 2 seconds to 4
+ seconds. Four seconds is a typical time between rings for many
+ countries. (closes issue #16460) Reported by: sum Patches:
+ issue16460.patch uploaded by rmudgett (license 664)
+ issue16460_v1.6.2.patch uploaded by rmudgett (license 664) Tested
+ by: sum, rmudgett Review: https://reviewboard.asterisk.org/r/634/
+ JIRA SWP-562 JIRA AST-334 JIRA SWP-901
+
+2010-04-29 15:31 +0000 [r259858-260049] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in
+ audiohook_write_list The middle_frame in the audiohook_write_list
+ function was being freed if a audiohook manipulator returned a
+ failure. This is incorrect logic. This patch resolves this and
+ adds detailed descriptions of how this function should work and
+ why manipulator failures must be ignored. (closes issue #17052)
+ Reported by: dvossel Tested by: dvossel (closes issue #16196)
+ Reported by: atis Review: https://reviewboard.asterisk.org/r/623/
+
+ * main/channel.c, channels/chan_local.c: resolves deadlocks in
+ chan_local Issue_1. In the local_hangup() 3 locks must be held at
+ the same time... pvt, pvt->chan, and pvt->owner. Proper deadlock
+ avoidance is done when the channel to hangup is the outbound
+ chan_local channel, but when it is not the outbound channel we
+ have an issue... We attempt to do deadlock avoidance only on the
+ tech pvt, when both the tech pvt and the pvt->owner are locked
+ coming into that loop. By never giving up the pvt->owner channel
+ deadlock avoidance is not entirely possible. This patch resolves
+ that by doing deadlock avoidance on both the pvt->owner and the
+ pvt when trying to get the pvt->chan lock. Issue_2. ast_prod() is
+ used in ast_activate_generator() to queue a frame on the channel
+ and make the channel's read function get called. This function is
+ used in ast_activate_generator() while the channel is locked,
+ which mean's the channel will have a lock both from the generator
+ code and the frame_queue code by the time it gets to
+ chan_local.c's local_queue_frame code... local_queue_frame
+ contains some of the same crazy deadlock avoidance that
+ local_hangup requires, and this recursive lock prevents that
+ deadlock avoidance from happening correctly. This patch removes
+ ast_prod() from the channel lock so only one lock is held during
+ the local_queue_frame function. (closes issue #17185) Reported
+ by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
+ (license 671) issue_17185_v2.diff uploaded by dvossel (license
+ 671) Tested by: schmoozecom, GameGamer43 Review:
+ https://reviewboard.asterisk.org/r/631/
+
+2010-04-28 21:07 +0000 [r259852] Leif Madsen <lmadsen at digium.com>
+
+ * config.guess: Update config.guess. Updating config.guess because
+ after installing Ubuntu Server 9.10 and running all the update
+ scripts, running ./configure would not continue because it was
+ unable to determine what kind of system I had. After updating
+ config.guess things started working again.
+
+2010-04-28 20:30 +0000 [r259748-259847] Jason Parker <jparker at digium.com>
+
+ * configure, configure.ac: Add AC_CONFIG_AUX_DIR to configure
+ script, so systems without install can use install-sh from our
+ source dir.
+
+ * makeopts.in: Missed this when removing $ID
+
+ * Makefile, configure, configure.ac: Remove usage of `id` since it
+ isn't useful and was causing breakge. Solaris `id` doesn't
+ support the -u argument. Instead of figuring out how to fix this
+ to work on Solaris, I decided to check why it was necessary and
+ where else it was used. It was only used in one place, and it
+ hasn't been needed for a very long time (I question whether it
+ was ever needed).
+
+2010-04-28 17:13 +0000 [r259664] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Do not play goodbye prompt after timeout of
+ message review. ABE-2124
+
+2010-04-27 21:53 +0000 [r259531] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: DAHDI "WARNING" message is confusing and
+ vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
+ failed: Success" Changed the warning to "Failed to decode
+ CallerID on channel 'name'". The message before it is likely more
+ specific about why the CallerID decode failed. SWP-501 AST-283
+
+2010-04-27 21:48 +0000 [r259526] Leif Madsen <lmadsen at digium.com>
+
+ * sounds/Makefile: Update sounds files. * Add additional sounds
+ prompts for say_enumeration * Update the English conference
+ sounds prompts so they are better quality and all sound more
+ consistent * Clean up the core-sounds-XX.txt and
+ extra-sounds-XX.txt files to include all present sound files Both
+ core (en, fr, es) and extra (en, fr) sounds files have been
+ updated. (closes issue #16200) Reported by: murf (closes issue
+ #17137) Reported by: lmadsen
+
+2010-04-27 21:15 +0000 [r259352-259441] Jason Parker <jparker at digium.com>
+
+ * main/editline/configure, main/editline/configure.in: Add gar to
+ the check for AR for those silly OSes (Solaris) that don't have
+ ar.
+
+ * configure, configure.ac: Support the silly OSes that don't have
+ ar and strip. Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when
+ path isn't specified, and AC_PATH_TOOLS doesn't exist, we'll just
+ switch to AC_CHECK_TOOLS.
+
+2010-04-27 18:14 +0000 [r259270] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample:
+ hidecalleridname parameter in chan_dahdi.conf Issue #7321
+ implements a new chan_dahdi configuration option. However, a
+ change mentioned in the issue was never implemented. This is the
+ change that will allow the feature to work. I added a note to
+ chan_dahdi.conf.sample about the feature. (closes issue #17143)
+ Reported by: djensen99 Patches: diff.txt uploaded by djensen99
+ (license NA) (One line change) Tested by: djensen99
+
+2010-04-26 21:44 +0000 [r259018-259104] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Let compilation succeed warning-free when
+ DONT_OPTIMIZE is turned off.
+
+ * main/channel.c: Prevent Newchannel manager events for dummy
+ channels. No Newchannel manager event will be fired for channels
+ that are allocated to not match a registered technology type.
+ Thus bogus channels allocated solely for variable substitution or
+ CDR operations do not result in a Newchannel event. (closes issue
+ #16957) Reported by: atis Review:
+ https://reviewboard.asterisk.org/r/601
+
+2010-04-25 18:09 +0000 [r258775] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_monitor.c: When StopMonitor is called, ensure that it
+ will not be restarted by a channel event. (closes issue #16590)
+ Reported by: kkm Patches: resmonitor-16590-trunk.239289.diff
+ uploaded by kkm (license 888)
+
+2010-04-22 21:49 +0000 [r258670] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/cdr.c, main/channel.c, res/res_features.c: Fix broken CDR
+ behavior. This change allows a CDR record previously marked with
+ disposition ANSWERED to be set as BUSY or NO ANSWER. Additionally
+ this change partially reverts r235635 and does not set the
+ AST_CDR_FLAG_ORIGINATED flag on CDRs generated from ast_call().
+ To preserve proper CDR behavior, the AST_CDR_FLAG_DIALED flag is
+ now cleared from all brige CDRs in ast_bridge_call(). (closes
+ issue #16797) Reported by: VarnishedOtter Tested by: mnicholson
+
+2010-04-21 21:45 +0000 [r258432] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Fix looping forever when no input received
+ in certain voicemail menu scenarios. Specifically, prompting for
+ an extension (when leaving or forwarding a message) or when
+ prompting for a digit (when saving a message or changing
+ folders). ABE-2122 SWP-1268
+
+2010-04-20 16:16 +0000 [r257856-258029] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Play correct prompt when voicemail store
+ failure occurs after attempted forward. If a user's mailbox was
+ full and a message was attempted to be forwarded to said box,
+ warnings on the console would indicate failure. However, the
+ played prompt was that of success (vm-msgsaved). Now storage
+ failure is taken into account and the correct prompt
+ (vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262
+
+ * apps/app_voicemail.c: make app_voicemail compile with
+ IMAP_STORAGE
+
+2010-04-16 21:15 +0000 [r257686] Dwayne M. Hubbard <dwayne.hubbard at gmail.com>
+
+ * apps/app_mixmonitor.c: Make the mixmonitor thread process audio
+ frames faster Mantis issue 17078 reports MixMonitor recordings
+ have shorter durations than the call duration. This was because
+ the mixmonitor thread was not processing frames from the
+ audiohook fast enough. The mixmonitor thread would slowly fall
+ behind the most recent audio frame and when the channel hangs up,
+ the mixmonitor thread would exit without processing the same
+ number of frames as the channel; leaving the mixmonitor recording
+ shorter than actual call duration. This revision fixes this issue
+ by moving the ast_audiohook_trigger_wait() and the subsequent
+ audiohook.status check into the block where the
+ ast_audiohook_read_frame() function returns NULL. (closes issue
+ #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded
+ by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review:
+ https://reviewboard.asterisk.org/r/611/
+
+2010-04-15 21:23 +0000 [r257467-257544] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/app.h, main/app.c: Allow application options
+ with arguments to contain parentheses, through a variety of
+ escaping techniques. Fixes SWP-1194 (ABE-2143). Review:
+ https://reviewboard.asterisk.org/r/604/
+
+ * channels/chan_sip.c: Don't recreate peer, when responding to a
+ repeated deregistration attempt. When a reply to a deregistration
+ is lost in transmit, the client retries the deregistration.
+ Previously, this would cause a realtime/autocreate peer to be
+ loaded back into memory, after it had already been correctly
+ purged. Instead, we just want to resend the reply without loading
+ the peer. (closes issue #16908) Reported by: kkm Patches:
+ 20100412__issue16908.diff.txt uploaded by tilghman (license 14)
+ Tested by: kkm
+
+2010-04-15 19:40 +0000 [r257342-257426] Leif Madsen <lmadsen at digium.com>
+
+ * doc/backtrace.txt: Update backtrace.txt documentation. Update the
+ backtrace.txt documentation so it conforms to the same layout as
+ other documents we've been working on recently. Additionally, add
+ a bunch of new information about gathering backtraces for crashes
+ and deadlocks, along with ways of verifying your file before
+ uploading it. Create a couple of one line commands for people to
+ generate the files we need. (closes issue #17190) Reported by:
+ lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen
+ (license 10) Tested by: lmadsen, pabelanger
+
+ * doc/backtrace.txt: Update address of the bug tracker.
+
+2010-04-14 23:08 +0000 [r257266] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: When forwarding a message, ensure that
+ prepending works correctly. This is a regression in 1.4, only.
+ (closes issue #17103) Reported by: mglazer Patches:
+ 20100408__issue17103.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+2010-04-13 16:46 +0000 [r257070] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/manager.c, configs/manager.conf.sample: Add an option to
+ restore past broken behavor of the Events manager action Before
+ r238915, certain values for the EventMask parameter of the Events
+ action would result in no response being returned. This patch
+ adds an option to restore that broken behavior. Also while fixing
+ this bug I discovered that passing an empty EventMasks parameter
+ would also result in no response being returned, this has been
+ fixed as well while being preserved when the broken behavior is
+ requested. (closes issue #17023) Reported by: nblasgen Review:
+ https://reviewboard.asterisk.org/r/602/
+
+2010-04-12 17:29 +0000 [r256900] Leif Madsen <lmadsen at digium.com>
+
+ * doc/HOWTO_collect_debug_information.txt (added): Add How-To
+ document on collecting debugging info for issues.asterisk.org
+ Paul Belanger has been helping a lot with bug tracking recently
+ and created this document that we can now point to when
+ additional debugging information is required. This document will
+ help those filing issues to know how to get the information
+ required when filing their issues. This will make things easier
+ on the developers. Initial text and changes by pabelanger. Tweaks
+ and editing by myself. (closes issue #17159) Reported by:
+ pabelanger Patches: HOWTO_collect_debug_information.txt.patch
+ uploaded by lmadsen (license 10) Tested by: tzafrir, pabelanger,
+ lmadsen
+
+2010-04-06 00:10 +0000 [r256225] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: DAHDI/PRI call to pri_channel_bridge() not
+ protected by PRI lock. SWP-1231 ABE-2163
+
+2010-04-05 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.31-rc1 Released
+
+2010-04-02 23:45 +0000 [r256009-256014] Russell Bryant <russell at digium.com>
+
+ * channels/chan_local.c: Resolve a deadlock that occurs due to a
+ pointless call to ast_bridged_channel() (closes issue #16840)
+ Reported by: bzing2 Patches: patch.txt uploaded by bzing2
+ (license 902) issue_16840.rev1.diff uploaded by russell (license
+ 2) Tested by: bzing2, russell
+
+ * main/channel.c: Remove extremely verbose debug message.
+
+2010-03-31 19:09 +0000 [r255591] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Ensure line terminators in email are
+ consistent. Fixes an issue with certain Mail Transport Agents,
+ where attachments are not interpreted correctly. (closes issue
+ #16557) Reported by: jcovert Patches:
+ 20100308__issue16557__1.4.diff.txt uploaded by tilghman (license
+ 14) 20100308__issue16557__1.6.0.diff.txt uploaded by tilghman
+ (license 14) 20100308__issue16557__trunk.diff.txt uploaded by
+ tilghman (license 14) Tested by: ebroad, zktech Reviewboard:
+ https://reviewboard.asterisk.org/r/544/
+
+2010-03-31 17:42 +0000 [r255503] Leif Madsen <lmadsen at digium.com>
+
+ * apps/app_dial.c, configs/sip.conf.sample: Add documentation
+ clarifying when 't' and 'T' can be used. (closes issue #17021)
+ Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad
+
+2010-03-30 20:56 +0000 [r255322-255409] Russell Bryant <russell at digium.com>
+
+ * channels/chan_h323.c: Don't kill Asterisk if the H323 listener
+ does not start.
+
+ * pbx/pbx_dundi.c: Don't make Asterisk not start if pbx_dundi fails
+ to initialize.
+
+2010-03-25 20:41 +0000 [r254714-254800] Jason Parker <jparker at digium.com>
+
+ * utils/Makefile: Don't remove local copies of utils in uninstall.
+
+ * main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS
+ issue with out-of-tree modules. Take 2, without ABI breakage this
+ time. Review: https://reviewboard.asterisk.org/r/588/
+
+2010-03-25 18:51 +0000 [r254639] Russell Bryant <russell at digium.com>
+
+ * Makefile, /: Update Asterisk 1.4 to use menuselect trunk. Review:
+ https://reviewboard.asterisk.org/r/590/
+
+2010-03-25 17:33 +0000 [r254452-254552] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/acl.h: Add doxygen for acl.h Review:
+ https://reviewboard.asterisk.org/r/528
+
+ * main/rtp.c: Several fixes regarding RFC2833 DTMF detection. Here
+ is a copy and paste of the details from my request on reviewboard
+ that dealt with these changes: Fix 1. The first change in place
+ is to fix Mantis issue 15811, which deals with a situation where
+ Asterisk will incorrectly interpret out of order RFC2833 frames
+ as duplicate DTMF digits. For instance, we would receive a
+ sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1
+ seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 seqno 7:
+ DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch when we
+ received the frame with seqno 5, we would interpret this as a new
+ DTMF 1. With this patch, we will check the seqno of the incoming
+ digit and not process the frame if the seqno is lower than the
+ last recorded seqno. Note that we do not record the seqno of the
+ dropped DTMF frame for future processing. While the above
+ situation is what was designed to be fixed, the patch is written
+ in such a way that the following would also be fixed too: seqno
+ 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) seqno 13:
+ DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno 15: DTMF 2
+ (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In this
+ second situation, the beginning of the DTMF 2 arrives before the
+ final end frame of the DTMF 1. With the patch, seqno 12 is no
+ processed and thus we properly interpret the DTMF. Fix 2. The
+ second change in place is to fix an issue like the following:
+ seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet
+ lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end)
+ *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had
+ code in place that was supposed to properly end the previously
+ unended DTMF 1. The problem was that the code was essentially a
+ no-op. The code would set up an end frame for the DTMF 1 but
+ would immediately overwrite the frame with the begin for DTMF 2.
+ I changed process_dtmf_rfc2833() so that instead of returning a
+ single frame, it is given as an output parameter a list of
+ frames. Each frame that needs to be returned is appended to this
+ list. Fix 3. The final change is a minor one where an
+ AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco
+ DTMF or an RFC 3389 frame and no frame was returned, then we
+ would return &ast_null_frame. The problem is that earlier in the
+ function, we may have generated an AST_CONTROL_SRCCHANGE frame
+ and put it in the list of frames we wish to return. This frame
+ would be lost in such a case. The patch fixes this problem
+ Review: https://reviewboard.asterisk.org/r/558
+
+2010-03-25 15:57 +0000 [r254451] Terry Wilson <twilson at digium.com>
+
+ * main/file.c: Handle new SRCCHANGE control message here too
+
+2010-03-24 00:37 +0000 [r254235] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_monitor.c: Ensure that monitor recordings are written to
+ the correct location (again) This is an extension to 248860. As
+ such the dialplan test has been extended: ; non absolute path,
+ not combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
+ exten => 5040, n, dial(sip/5001) ; absolute path, not combined
+ exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten =>
+ 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1,
+ monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
+ exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n,
+ dial(sip/5001) ; combined: changemonitor from no path to non
+ absolute path exten => 5044, 1, monitor(wav,monitor_test6,m)
+ exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this
+ wasn't possible before exten => 5044, n, dial(sip/5001) ; non
+ absolute path, combined exten => 5045, 1,
+ monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n,
+ dial(sip/5001) ; absolute path, combined exten => 5046, 1,
+ monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n,
+ dial(sip/5001) ; no path, combined exten => 5047, 1,
+ monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to absolute (leaves
+ tmp/jeff) exten => 5048, 1,
+ monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n,
+ changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5049, 1,
+ monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n,
+ changemonitor(tmp/jeff/monitor_test14) exten => 5049, n,
+ dial(sip/5001) ; combined: changemonitor from no path to absolute
+ exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n,
+ changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to no path
+ (leaves /tmp/jeff) exten => 5051, 1,
+ monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n,
+ changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ;
+ not combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
+ exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to non
+ absolute exten => 5053, 1, monitor(wav,monitor_test21) exten =>
+ 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n,
+ dial(sip/5001) ; not combined: changemonitor from non absolute to
+ absolute (leaves tmp/jeff) exten => 5054, 1,
+ monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n,
+ changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n,
+ dial(sip/5001) ; not combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5055, 1,
+ monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n,
+ changemonitor(tmp/jeff/monitor_test25) exten => 5055, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to
+ absolute exten => 5056, 1, monitor(wav,monitor_test26) exten =>
+ 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056,
+ n, dial(sip/5001) ; not combined: changemonitor from absolute to
+ no path (leaves /tmp/jeff) exten => 5057, 1,
+ monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n,
+ changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001)
+
+2010-03-23 22:45 +0000 [r254046-254161] Jason Parker <jparker at digium.com>
+
+ * main/astobj2.c, main/lock.c (removed), main/channel.c,
+ main/Makefile, include/asterisk/astobj2.h, UPGRADE.txt,
+ include/asterisk/lock.h: Revert revisions 254046 and 254098.
+
+ * UPGRADE.txt: Add note about the out-of-tree module ABI changes.
+
+ * main/astobj2.c, main/lock.c (added), main/channel.c,
+ main/Makefile, include/asterisk/astobj2.h,
+ include/asterisk/lock.h: Allow out-of-tree modules to load,
+ regardless of DEBUG_THREADS/DEBUG_CHANNEL_LOCKS differences. This
+ can be guaranteed by forcing the ABI to no longer change when
+ these compiler flags are set. An unfortunate side-effect to this
+ is that there is an ABI change here. However, there is some
+ mitigation. Existing modules *will* fail to load since they would
+ require functions that no longer exist. Review:
+ https://reviewboard.asterisk.org/r/508/
+
+2010-03-22 19:50 +0000 [r253799] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_features.c: Unconditionally copy the caller's account
+ code to the called party. (related to issue #16331)
+
+2010-03-21 14:26 +0000 [r253631-253670] Russell Bryant <russell at digium.com>
+
+ * main/Makefile: Fix final link on FreeBSD by adding the
+ PTHREAD_CFLAGS.
+
+ * main/sched.c, Makefile, apps/app_dial.c, channels/chan_dahdi.c,
+ main/manager.c, res/res_features.c, main/http.c, main/utils.c,
+ pbx/pbx_dundi.c, apps/app_followme.c: Resolve a number of FreeBSD
+ build issues.
+
+2010-03-18 17:57 +0000 [r253252-253349] Leif Madsen <lmadsen at digium.com>
+
+ * apps/app_userevent.c: Typo found while fixing issue #16961.
+
+ * doc/localchannel.txt: Synchronize text in localchannels.txt and
+ localchannels.tex. (issue #16963)
+
+ * doc/localchannel.txt: Update new Local channel documentation. The
+ original reporter, Kobaz, of an issue with a Local channel that
+ inspired the Local channel documentation provided some tweaks to
+ the documentation after testing what I had written. Hopefully
+ anything that was vague or unclear has been cleaned up by these
+ changes. (closes issue #16963) Reported by: kobaz Patches:
+ localchannel-2.txt uploaded by kobaz (license 834) Tested by:
+ kobaz, lmadsen
+
+2010-03-17 16:25 +0000 [r253158] Terry Wilson <twilson at digium.com>
+
+ * main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c,
+ channels/chan_mgcp.c, channels/chan_sip.c,
+ include/asterisk/rtp.h: Revert API change in release branches
+ This re-renames ast_rtp_update_source to ast_rtp_new_source
+
+2010-03-17 00:26 +0000 [r253018] Leif Madsen <lmadsen at digium.com>
+
+ * configs/say.conf.sample: Add french snipset to say.conf. Add the
+ french snipset to say.conf. (Closes issue #15799)
+
+2010-03-16 20:52 +0000 [r252766-252928] Russell Bryant <russell at digium.com>
+
+ * Makefile.rules: Backport chan_sip build fix for Mac OSX 10.6 from
+ trunk.
+
+ * codecs/gsm/Makefile: Use uname -s, as done in trunk.
+
+ * codecs/gsm/Makefile: Apply codec_gsm Mac OS X 10.6 build fix that
+ is in trunk and 1.6.X.
+
+ * utils/Makefile: Don't treat warnings as errors for muted. muted
+ supports OS X, but uses functions marked as deprecated in 10.6.
+ However, the functions are still supported, so just ignore the
+ warnings for now and allow the build to proceed.
+
+2010-03-16 18:46 +0000 [r252761] Leif Madsen <lmadsen at digium.com>
+
+ * configs/extensions.ael.sample: Additional extensions.ael global
+ variable fixes. Fixing up a couple more overlapping global
+ variable namespaces shared with extensions.conf.sample. Also
+ noticed a few of the lines that were commented out didn't have
+ the closing semi-colon so I added that as well. (issue #17035)
+
+2010-03-15 21:43 +0000 [r252617] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/init.d/org.asterisk.asterisk.plist: Uh, yeah. Umask. I'm
+ stupid.
+
+2010-03-15 20:48 +0000 [r252531-252533] Leif Madsen <lmadsen at digium.com>
+
+ * configs/extensions.ael.sample: Update extensions.ael file to not
+ overlap extensions.conf. Updated the extensions.ael file so the
+ global variables don't overlap those that we have in
+ extensions.conf (sample files). This way unexpected things won't
+ happed hopefully if both pbx_ael and res_config are loaded.
+ (closes issue #17035) Reported by: pprindeville
+
+ * configure, configs/extensions.ael.sample: Revert last commit that
+ had bad changed to configure.
+
+ * configure, configs/extensions.ael.sample: Update extensions.ael
+ file to not overlap extensions.conf. Updated the extensions.ael
+ file so the global variables don't overlap those that we have in
+ extensions.conf (sample files). This way unexpected things won't
+ happed hopefully if both pbx_ael and res_config are loaded.
+ (closes issue #17035) Reported by: pprindeville
+
+2010-03-15 01:39 +0000 [r252361-252366] Tilghman Lesher <tlesher at digium.com>
+
+ * Makefile: Typo
+
+ * main/asterisk.c, Makefile,
+ contrib/init.d/org.asterisk.asterisk.plist (added): Launch
+ Asterisk on Mac OS X with launchd. Reviewboard:
+ https://reviewboard.asterisk.org/r/551/
+
+2010-03-13 00:30 +0000 [r252175] Terry Wilson <twilson at digium.com>
+
+ * main/rtp.c, channels/chan_mgcp.c, main/channel.c,
+ channels/chan_sip.c, channels/chan_skinny.c,
+ include/asterisk/rtp.h, channels/chan_h323.c,
+ configs/sip.conf.sample, include/asterisk/frame.h: Merged
+ revisions 252089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 |
+ twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
+ Only change the RTP ssrc when we see that it has changed This
+ change basically reverts the change reviewed in
+ https://reviewboard.asterisk.org/r/374/ and instead limits the
+ updating of the RTP synchronization source to only those times
+ when we detect that the other side of the conversation has
+ changed the ssrc. The problem is that SRCUPDATE control frames
+ are sent many times where we don't want a new ssrc, including
+ whenever Asterisk has to send DTMF in a normal bridge. This is
+ also not the first time that this mistake has been made. The
+ initial implementation of the ast_rtp_new_source function also
+ changed the ssrc--and then it was removed because of this same
+ issue. Then, we put it back in again to fix a different issue.
+ This patch attempts to only change the ssrc when we see that the
+ other side of the conversation has changed the ssrc. It also
+ renames some functions to make their purpose more clear. Review:
+ https://reviewboard.asterisk.org/r/540/ ........
+
+2010-03-12 19:58 +0000 [r251986-251997] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Forward declaring dahdi_pri was already
+ done.
+
+ * channels/chan_dahdi.c: Make chan_dahdi wakeup_sub() prototype not
+ conditional.
+
+2010-03-11 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.30 released
+
+2010-03-04 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.30-rc3 released
+
+2010-03-03 21:28 +0000 [r250613] Leif Madsen <lmadsen at digium.com>
+
+ * doc/localchannel.txt: Update existing Local channel
+ documentation. A complete re-write of the Local channel
+ documentation has been performed, with the existing information
+ from localchannel.txt and localchannel.tex merged in. (issue
+ #16637) Reported by: kobaz Patches: localchannel.tex uploaded by
+ lmadsen (license 10) localchannel.txt uploaded by lmadsen
+ (license 10) Tested by: lmadsen, jsmith, mmichelson
+
+2010-03-03 19:04 +0000 [r250480] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Make sure to clear red alarm after
+ polarity reversal. From the issue: The automatic overnight line
+ tests (or manual ones) used on UK (BT) lines causes a red alarm
+ on a dahdi / TDM400P connected channel. This is because the line
+ uses voltage tests (battery loss) and polarity reversal. The
+ polarity reversal causes chan_dahdi to initiate v23 CallerID
+ processing but during this the event DAHDI_EVENT_NOALARM is
+ ignored so that the alarm is never cleared. (closes issue #14163)
+ Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded
+ by jedi98 (license 653) Tested by: mattbrown, Chainsaw,
+ mikeeccleston
+
+2010-03-03 18:02 +0000 [r250394] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: fixes problem with duplicate TXREQ packets
+ When Asterisk receives an IAX2 TXREQ packet, try_transfer() will
+ call store_by_transfercallno() to link the chan_iax2_pvt struct
+ into iax_transfercallno_pvts. If a duplicate TXREQ packet is
+ received for the same call, the pvt struct will be linked into
+ iax_transfercallno_pvts multiple times. This patch fixes this.
+ Thanks rain for debugging this and providing a patch! (closes
+ issue #16904) Reported by: rain Patches:
+ iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
+ by: rain, dvossel
+
+2010-03-02 21:08 +0000 [r250041-250050] Leif Madsen <lmadsen at digium.com>
+
+ * doc/imapstorage.txt: Update IMAP documentation. Update the IMAP
+ documentation to make it clear that storing voicemails in the
+ same folder as a large number of emails could potentially cause
+ significant slow downs when writing or retrieving voicemails.
+ (closes issue #16704) Reported by: TimeHider Tested by: lmadsen,
+ TimeHider
+
+ * configs/cdr.conf.sample: Update documentation to clarify purpose
+ of unanswered option. (closes issue #16267) Reported by: elsto
+ Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license
+ 10) Tested by: davidw, elsto
+
+ * doc/configuration.txt: Update documentation to not imply we
+ support overriding options. (issue #16855) Reported by: davidw
+
+2010-03-02 19:36 +0000 [r249845-249946] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * apps/app_echo.c: revert ability to exit echo app caused a
+ regression, as only supported VOICE, not VIDEO etc. Left in small
+ formatting change. (issue #16880)
+
+ * apps/app_echo.c: fixes ability to exit echo app when called from
+ a ISDN channel, null frames prevent '#' exit. Now only echo back
+ VOICE and DTMF frames (issue #16880) Reported by: alecdavis
+ Patches: based on echo_exit_1-6-1.diff.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis
+
+2010-03-01 19:35 +0000 [r249671] Sean Bright <sean at malleable.com>
+
+ * apps/app_voicemail.c: Fix crash in app_voicemail related to
+ message counting. We were passing a 'struct inprocess **' and
+ treating it like a 'struct inprocess *' causing a segfault.
+ (closes issue #16921) Reported by: whardier Patches:
+ 20100301_issue16921.patch uploaded by seanbright (license 71)
+ Tested by: whardier
+
+2010-03-01 17:02 +0000 [r249536] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_local.c: Modify queued frames from local channels
+ to not set the other side to up In this case, attended transfers
+ were broken due to ast_feature_request_and_dial detecting the
+ channel being set to up before the answer frame could be read and
[... 27773 lines stripped ...]
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