[svn-commits] lmadsen: tag 1.4.28-rc1 r233860 - /tags/1.4.28-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Dec 9 10:46:03 CST 2009
Author: lmadsen
Date: Wed Dec 9 10:45:59 2009
New Revision: 233860
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=233860
Log:
Importing files for 1.4.28-rc1 release.
Added:
tags/1.4.28-rc1/.lastclean (with props)
tags/1.4.28-rc1/.version (with props)
tags/1.4.28-rc1/ChangeLog (with props)
Added: tags/1.4.28-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.4.28-rc1/.lastclean?view=auto&rev=233860
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--- tags/1.4.28-rc1/ChangeLog (added)
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+2009-12-09 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.28-rc1
+
+2009-12-09 15:14 +0000 [r233782] Russell Bryant <russell at digium.com>
+
+ * main/loader.c, formats/format_ilbc.c, formats/format_vox.c,
+ include/asterisk/module.h, formats/format_pcm.c,
+ formats/format_h263.c, formats/format_g723.c,
+ formats/format_h264.c, formats/format_jpeg.c,
+ formats/format_g726.c, formats/format_gsm.c,
+ formats/format_g729.c, formats/format_sln.c,
+ formats/format_wav.c, formats/format_ogg_vorbis.c,
+ formats/format_wav_gsm.c: Set a module load priority for format
+ modules. A recent change to app_voicemail made it such that the
+ module now assumes that all format modules are available while
+ processing voicemail configuration. However, when autoloading
+ modules, it was possible that app_voicemail was loaded before the
+ format modules. Since format modules don't depend on anything,
+ set a module load priority on them to ensure that they get loaded
+ first when autoloading. This version of the patch is specific to
+ Asterisk 1.4 and 1.6.0. These versions did not already support
+ module load priority in the module API. This adds a trivial
+ version of this which is just a module flag to include it in a
+ pass before loading "everything". Thanks to mmichelson for the
+ review! (closes issue #16412) Reported by: jiddings Tested by:
+ russell Review: https://reviewboard.asterisk.org/r/445/
+
+2009-12-08 00:02 +0000 [r233618] Atis Lezdins <atis at iq-labs.net>
+
+ * contrib/valgrind.supp: Merged revisions 233577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r233577 |
+ atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8 lines Fix
+ compatibility with valgrind 3.3 and older. (noticed in issue
+ #16388) Reported by: parisioa Patches: valgrind.supp uloaded by
+ atis (license 242) Tested by: atis, parisioa ........
+
+2009-12-07 23:24 +0000 [r233471-233609] David Vossel <dvossel at digium.com>
+
+ * main/utils.c: hex escape control and non 7-bit clean characters
+ in uri_encode In ast_uri_encode, non 7-bit clean characters were
+ being hex escaped correctly, but control characters were not.
+ (issue #16299)
+
+ * channels/chan_sip.c: fixes missing Contact header angle brackets
+ (closes issue #16298) Reported by: mgernoth Patches:
+ reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested
+ by: dvossel
+
+2009-12-07 16:11 +0000 [r233392] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Allow SDP packets with only video session
+ information. (closes issue #16387) Reported by: zalex1953 Tested
+ by: mnicholson, zalex1953
+
+2009-12-04 21:54 +0000 [r233116-233279] David Vossel <dvossel at digium.com>
+
+ * configs/iax.conf.sample: clarify requirecalltoken option in
+ iax.sample.conf (closes issue #16223) Reported by: bklang
+ Patches: clarify-iax-requirecalltoken.patch uploaded by bklang
+ (license 919)
+
+ * apps/app_voicemail.c: document and rename strip_control() in
+ app_voicemail (closes issue #16291) Reported by: wdoekes
+
+2009-12-04 17:12 +0000 [r233092] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Only do frame payload check for HOLD frames. This
+ code was added for helping to debug the source of invalid HOLD
+ frames. However, a side effect of this is that it will
+ incorrectly report errors for frames that have an integer
+ payload. Make the check for this block specific to the HOLD frame
+ case.
+
+2009-12-04 16:59 +0000 [r233014-233091] Matthias Nick <mnick at digium.com>
+
+ * pbx/pbx_config.c: Parse global variables or expressions in hint
+ extensions Parse global variables or expressions in hint
+ extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)}
+ (closes issue #16166) Reported by: rmudgett Tested by: mnick,
+ rmudgett
+
+ * main/dsp.c: Warning message gets displayed only once Added
+ additional field 'int display_inband_dtmf_warning', which when
+ set to '1' displays the warning ('Inband DTMF is not supported on
+ codec %s. Use RFC2833'), and when set to '0' doesn't display the
+ warning. Otherwise you would get hundreds of warnings every
+ second. (closes issue #15769) Reported by: falves11 Patches:
+ patch_15769_14.txt uploaded by mnick (license 874) Tested by:
+ mnick, falves11
+
+2009-12-03 20:10 +0000 [r232820] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Deprecate "cz" in favor of "cs". Also,
+ change the use of language codes so that language registers as a
+ prefix, rather than an exact match. (closes issue #16272)
+ Reported by: patrol-cz Patches: 20091203__issue16272.diff.txt
+ uploaded by tilghman (license 14)
+
+2009-12-02 21:57 +0000 [r232581] Jeff Peeler <jpeeler at digium.com>
+
+ * main/manager.c: Send ack (response/message) after receiving
+ manager action userevent (closes issue #16264) Reported by: dimas
+ Patches: event-ack.patch uploaded by dimas (license 88)
+
+2009-12-02 19:03 +0000 [r232444] David Vossel <dvossel at digium.com>
+
+ * apps/app_queue.c: fixes app_queue ao2 error (closes issue #16369)
+ Reported by: vrban Patches: queue_issue_1.4.diff uploaded by
+ dvossel (license 671) Tested by: dvossel
+
+2009-12-02 17:04 +0000 [r232355] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_amd.c: Fix a bug where if you hung up very quickly after
+ calling AMD it would overwrite the AMDSTATUS of HANGUP with
+ TOOLONG. (closes issue #16239) Reported by: CGMChris
+
+2009-12-02 16:59 +0000 [r232268-232350] David Vossel <dvossel at digium.com>
+
+ * main/acl.c: ast_outaddrfor doesn't do htons() on port, looks odd
+ in strace. (closes issue #16290) Reported by: wdoekes
+
+ * funcs/func_groupcount.c: fixes segfault in func_groupcount closes
+ issue #16337) Reported by: Parantido Patches: issue_16337.diff
+ uploaded by dvossel (license 671) Tested by: Parantido, dvossel
+
+2009-12-02 04:05 +0000 [r232165] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c: Fix compiling without devmode (closes issue
+ #16367) Reported by: falves11
+
+2009-12-02 00:42 +0000 [r232090] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Do not modify the gain settings on data
+ calls. (The digital flag actually represents a data call.)
+ (closes issue #15972) Reported by: udosw Patches:
+ transcap_digital_fix.diff.txt uploaded by alecdavis (license 585)
+ Tested by: alecdavis
+
+2009-12-01 23:25 +0000 [r232007] Russell Bryant <russell at digium.com>
+
+ * main/file.c: Fix a warning pointed out by buildbot.
+
+2009-12-01 21:52 +0000 [r231911-231926] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c: log channel name in dev mode as well
+
+ * main/channel.c: Fix crash with invalid frame data The crash was
+ happening as a result of a frame containing an invalid data
+ pointer, but was set with data length of zero. The few times the
+ issue was reproduced it _seemed_ that the frame was queued
+ properly, that is the data pointer was set to NULL. I never could
+ reproduce the crash so as a last resort the crash has been fixed,
+ but a check in __ast_read has been added to give as much
+ information about the source of problematic frames in the future.
+ (closes issue #16058) Reported by: atis
+
+2009-12-01 21:14 +0000 [r231853] David Vossel <dvossel at digium.com>
+
+ * main/pbx.c: WaitExten m option with no parameters generates frame
+ with zero datalen but non-null data ptr
+
+2009-12-01 15:34 +0000 [r231614-231740] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/file.c: Ignore unknown formats in ast_format_str_reduce()
+ and return an error if no know formats are found.
+
+ * apps/app_voicemail.c, include/asterisk/file.h, main/file.c,
+ main/app.c: Remove duplicate entries from voicemail format lists.
+ This prevents app_voicemail from entering an infinite loop when
+ the same format is specified twice in the format list. (closes
+ issue #15625) Reported by: Shagg63 Tested by: mnicholson Review:
+ https://reviewboard.asterisk.org/r/429/
+
+2009-11-30 17:14 +0000 [r231437-231441] David Vossel <dvossel at digium.com>
+
+ * main/rtp.c: fixes crash caused by RTP comfort noise payload
+ greater than 24 bytes AST-2009-010 (closes issue #16242) Reported
+ by: amorsen Patches: issue16242.diff uploaded by oej (license
+ 306) Tested by: amorsen, oej, dvossel
+
+ * apps/app_queue.c: app_queue crashes randomly, often during
+ call-transfers In app_queue, it is possible for a call_queue to
+ be destroyed while another object still holds a pointer to it.
+ This patch converts call_queue objects to ao2 objects allowing
+ them to be ref counted. This makes it safe for the queue_ent
+ object in queue_exec() to reference it's parent call_queue even
+ after it has left the queue. (closes issue #15686) Reported by:
+ Hatrix Patches: v2_queue_ao2.diff uploaded by dvossel (license
+ 671) Tested by: dvossel, aragon Review:
+ https://reviewboard.asterisk.org/r/427/
+
+2009-11-25 22:31 +0000 [r231298] Tilghman Lesher <tlesher at digium.com>
+
+ * main/channel.c: After a frame duplication failure, unlock the
+ channel before returning.
+
+2009-11-25 21:38 +0000 [r231233-231235] David Vossel <dvossel at digium.com>
+
+ * apps/app_dial.c: fixes solaris segfault on dial with verbosity >=
+ 3 (closes issue #16193) Reported by: asgaroth Patches:
+ bug_16193_1.4.21.2_vers.diff uploaded by snuffy (license 35)
+ Tested by: asgaroth, snuffy
+
+ * channels/chan_sip.c: fixes conditional jump or move depending on
+ uninitialised STACK value (closes issue #16261) Reported by:
+ edguy3 Patches: edguy16261.patch uploaded by edguy3 (license 917)
+
+2009-11-23 15:31 +0000 [r230772-230875] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c: When 'sip set debug' is enabled, and the
+ last line of an incoming SIP message is not properly newline
+ terminated, ensure that that line is included in the debug
+ output. (part of issue #16268)
+
+ * main/editline/makelist.in, channels/chan_sip.c,
+ channels/ring_tone.h, channels/busy_tone.h: Correct fix for issue
+ #16268... the reporter's original patch was very close to
+ correct.
+
+ * channels/chan_sip.c: Ensure that SDP parsing does not ignore the
+ last line of the SDP. (closes issue #16268) Reported by: sgimeno
+
+2009-11-20 20:53 +0000 [r230627] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_features.c: Copy the peer CDR's userfield to the bridge
+ CDR if it exists. This is necessary for the recordagentcalls
+ option in chan_agent to store the recorded file name in the
+ bridge CDR. (closes issue #14590) Reported by: msetim Patches:
+ queue_agent_userfield.patch uploaded by Laureano (license 265)
+ Tested by: Laureano, mnicholson
+
+2009-11-19 21:22 +0000 [r230508] David Vossel <dvossel at digium.com>
+
+ * apps/app_mixmonitor.c: fixes MixMonitor thread not exiting when
+ StopMixMonitor is used (closes issue #16152) Reported by: AlexMS
+ Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license
+ 671) Tested by: dvossel, AlexMS Review:
+ https://reviewboard.asterisk.org/r/424/
+
+2009-11-19 16:09 +0000 [r230469] Michiel van Baak <michiel at vanbaak.info>
+
+ * main/asterisk.c: Update copyright year in visible output. (cli)
+ Spotted by Stuart Henderson
+
+2009-11-30 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.27.1
+
+ * AST-2009-010
+
+ * SDP parser regression fix (issue #16268)
+
+2009-11-18 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.27
+
+2009-11-13 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.27-rc5
+
+2009-11-12 16:41 +0000 [r229669] David Vossel <dvossel at digium.com>
+
+ * funcs/func_audiohookinherit.c: fixes merging error, datastore was
+ being freed in the wrong function. (closes issue #16219) Reported
+ by: aragon
+
+2009-11-11 19:46 +0000 [r229498] David Brooks <dbrooks at digium.com>
+
+ * main/pbx.c: Solaris doesn't like NULL going to ast_log Solaris
+ will crash if NULL is passed to ast_log. This simple patch simply
+ uses S_OR to get around this. (closes issue #15392) Reported by:
+ yrashk
+
+2009-11-10 22:09 +0000 [r229360] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c: If two pattern classes start with the same digit and
+ have the same number of characters, they will compare equal. The
+ example given in the issue report is that of [234] and [246],
+ which have these characteristics, yet they are clearly not
+ equivalent. The code still uses these two characteristics, yet
+ when the two scores compare equal, an additional check will be
+ done to compare all characters within the class to verify
+ equality. (closes issue #15421) Reported by: jsmith Patches:
+ 20091109__issue15421__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: jsmith, thedavidfactor
+
+2009-11-10 21:45 +0000 [r229355] David Ruggles <thedavidfactor at gmail.com>
+
+ * doc/externalivr.txt: Fix ExternalIVR Documentation Remove
+ documentation for event that doesn't function (closes issue
+ #16220) Reported by: thedavidfactor Patches:
+ externalivr.txt.20091110.1622.patch uploaded by thedavidfactor
+ (license 903)
+
+2009-11-10 20:03 +0000 [r229281] Joshua Colp <jcolp at digium.com>
+
+ * codecs/codec_g726.c: Remove broken support for direct transcoding
+ between G.726 RFC3551 and G.726 AAL2. On some systems the
+ translation core would actually consider g726aal2 -> g726 ->
+ signed linear to be a quicker path then g726aal2 -> signed linear
+ which exposed this problem. (closes issue #15504) Reported by:
+ globalnetinc
+
+2009-11-10 17:23 +0000 [r229191] David Ruggles <thedavidfactor at gmail.com>
+
+ * doc/externalivr.txt: Document ExternalIVR event tag collision
+ ExternalIVR uses the D tag for two different event types. This
+ documents that behavior and how to differentiate between the two
+ cases. Also includes a minor spelling fix and clarification
+ (closes issue #16211) Reported by: thedavidfactor Patches:
+ externalivr.txt.20091109.1507.patch uploaded by thedavidfactor
+ (license 903)
+
+2009-11-10 17:15 +0000 [r229167] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: don't crash on log message in solaris
+ AST-2009-006 (closes issue #16206) Reported by: bklang Tested by:
+ bklang
+
+2009-11-10 15:22 +0000 [r229091] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Reverted revision 202022. (closes issue
+ #16175) Reported by: paul-tg
+
+2009-11-09 Leif Madsen <lmadsen at digium.com>
+
+ * Release Astersik 1.4.27-rc4
+
+2009-11-09 15:37 +0000 [r228896] Leif Madsen <lmadsen at digium.com>
+
+ * main/channel.c: Update WARNING message. Update a WARNING message
+ to give a suggested fix when encountered. (closes issue #16198)
+ Reported by: atis Tested by: atis
+
+2009-11-09 14:16 +0000 [r228827] Matthew Nicholson <mnicholson at digium.com>
+
+ * include/asterisk/lock.h: Perform limited bounds checking when
+ destroying ast_mutex_t structures to make sure we don't try to
+ use negative indices. (closes issue #15588) Reported by: zerohalo
+ Patches: 20090820__issue15588.diff.txt uploaded by tilghman
+ (license 14) Tested by: zerohalo
+
+2009-11-06 22:33 +0000 [r228692] David Vossel <dvossel at digium.com>
+
+ * main/channel.c: fixes audiohook write crash occuring in chan_spy
+ whisper mode. After writing to the audiohook list in ast_write(),
+ frames were being freed incorrectly. Under certain conditions
+ this resulted in a double free crash. (closes issue #16133)
+ Reported by: wetwired (closes issue #16045) Reported by:
+ bluecrow76 Patches: issue16045.diff uploaded by dvossel (license
+ 671) Tested by: bluecrow76, dvossel, habile
+
+2009-11-06 18:32 +0000 [r228547] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Don't overwrite caller ID name on a trunk
+ with the configured fullname when using users.conf (issue
+ ABE-1989)
+
+2009-11-06 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.27-rc3
+
+2009-11-06 17:07 +0000 [r228418] David Vossel <dvossel at digium.com>
+
+ * codecs/codec_ilbc.c: fixes segfault in iLBC For reasons not yet
+ known, it appears possible for an ast_frame to have a datalen
+ greater than zero while the actual data is NULL during Packet
+ Loss Concealment. Most codecs don't support PLC so this doesn't
+ affect them. This patch catches the malformed frame and prevents
+ the crash from occuring. Additional efforts to determine why it
+ is possible for a frame to look like this are still being
+ investigated. (issue #16979)
+
+2009-11-06 16:41 +0000 [r228409] Joshua Colp <jcolp at digium.com>
+
+ * main/abstract_jb.c: Fix a bug caused by a partially invalid frame
+ (from the jitterbuffer) passing through the Asterisk core.
+ (closes issue #15560) Reported by: jvandal (closes issue #15709)
+ Reported by: covici
+
+2009-11-06 16:26 +0000 [r228378] Matthew Nicholson <mnicholson at digium.com>
+
+ * funcs/func_base64.c, main/utils.c: Properly handle '=' while
+ decoding base64 messages and null terminate strings returned from
+ BASE64_DECODE. (closes issue #15271) Reported by: chappell
+ Patches: base64_fix.patch uploaded by chappell (license 8) Tested
+ by: kobaz
+
+2009-11-06 15:41 +0000 [r228272-228338] David Vossel <dvossel at digium.com>
+
+ * main/astfd.c: fixes crash in astfd.c (closes issue #15981)
+ Reported by: slavon
+
+ * funcs/func_audiohookinherit.c: fixes memory leak in
+ func_audiohookinherit.c (closes issue 0015394) Reported by:
+ boroda Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks
+ (license 790) Tested by: dbrooks, boroda
+
+2009-11-05 19:14 +0000 [r228079] Jason Parker <jparker at digium.com>
+
+ * channels/chan_vpb.cc: Fix crash on VPB exception when no hardware
+ is present. (closes issue #14970) Reported by: tzafrir Patches:
+ vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
+ markwaters
+
+2009-11-05 18:59 +0000 [r228078] David Brooks <dbrooks at digium.com>
+
+ * channels/chan_misdn.c: chan_misdn Asterisk 1.4.27-rc2 crash Crash
+ related to chan_misdn connection. Patch submitted by
+ gknispel_proformatique, tested by francesco_r. "I have many crash
+ since i have upgraded to Asterisk 1.4.27-rc2. Attached a full
+ bt." This patch zeros out an ast_frame. (closes issue #16041)
+ Reported by: francesco_r
+
+2009-11-04 23:47 +0000 [r227944] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_monitor.c: Fix incorrect filename comparsion after
+ monitor file change The logic to detect if a requested file is
+ indeed a different file from the current file was incorrect. The
+ main issue being confusion of the use of filename_base which was
+ previously set without pathing information and then compared to
+ another full path. Robust file comparison logic has been added to
+ properly check if two files are the same even if symlinks are
+ used. (closes issue #15313) Reported by: caspy Patches:
+ 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
+ 325) but mostly tilghman's work
+
+2009-11-04 20:52 +0000 [r227758-227827] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_dial.c: This patch modifies the Dial application to
+ monitor the calling channel for hangups while playing back
+ announcements. (closes issue #16005) Reported by: falves11
+ Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: mnicholson, falves11 Review:
+ https://reviewboard.asterisk.org/r/407/
+
+ * channels/chan_sip.c: Modify the SDP parsing code to parse session
+ and media level items separately. With the new code, media level
+ proprieties should no longer be confused with session level
+ proprieties. This change also reorganizes some of the SDP parsing
+ code which should make it easier to manage in the future. (closes
+ issue #14994) Reported by: frawd Tested by: frawd, mnicholson,
+ file Review: https://reviewboard.asterisk.org/r/385/
+
+2009-11-04 19:25 +0000 [r227700-227735] Joshua Colp <jcolp at digium.com>
+
+ * static-http/prototype.js: Fix a security issue where it may be
+ possible for someone to execute a cross-site AJAX request
+ exploit. (AST-2009-009)
+
+ * channels/chan_sip.c: Fix a security issue where sending a
+ REGISTER with a differing username in the From URI and
+ Authorization header would reveal whether it was valid or not.
+ (AST-2009-008)
+
+2009-11-03 17:55 +0000 [r227275] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Make sure the outgoing flag is cleared if
+ a new channel fails to get created for outgoing calls. This is
+ the relevant portion of asterisk/trunk -r226648
+
+2009-11-03 15:36 +0000 [r227166] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix a bug where an RPID header could be
+ generated with a blank username in the URI. (closes issue #15909)
+ Reported by: kobaz
+
+2009-11-03 10:48 +0000 [r227088-227090] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Fixing bug before someone reports it...
+
+ * channels/chan_sip.c: Adding IP address in Contact ACL log message
+ and removing redundant message (based on kpfleming's feedback)
+
+ * channels/chan_sip.c: Use proper response code when violating
+ Contact ACL's. Review: https://reviewboard.asterisk.org/r/415/
+ Thanks kpfleming for a quick review. (EDVX-003)
+
+2009-11-02 20:52 +0000 [r226972] David Brooks <dbrooks at digium.com>
+
+ * channels/chan_sip.c: SIP channel name uniqueness SIP channel
+ names were supposed to be unique by way of a name suffix derived
+ from the pointer to the channel's private data. Uniqueness was
+ preserved on 32-bit systems, but not on 64-bit systems. This
+ patch, as suggested by kpfleming, replaces this suffix with a
+ simple incremented unsigned int. (closes issue #15152) Reported
+ by: palbrecht Review: https://reviewboard.asterisk.org/r/420/
+
+2009-11-02 18:08 +0000 [r226889] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_dial.c: Fix a bug where the recorded privacy
+ introduction file would not get removed if the caller hung up
+ while the called party had not yet answered. This was fixed by
+ introducing an argument to the 'n' option which, when enabled,
+ removes the introduction file under all scenarios. This was done
+ to preserve the behavior that has existed for quite some time.
+ (closes issue #14674) Reported by: ulogic Patches: bug14674.patch
+ uploaded by jpeeler (license 325)
+
+2009-11-02 17:14 +0000 [r226811] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/init.d/rc.redhat.asterisk: Don't allow two separate
+ instances of safe_asterisk when restarting from the init script.
+ (closes issue #14562) Reported by: davidw Patches: Initially
+ 20091022__issue14562.diff.txt uploaded by tilghman (license 14)
+ Modified to 20091030__Issue14562_diff.txt uploaded by davidw
+ (license 780) Tested by: davidw
+
+2009-11-02 15:31 +0000 [r226688-226736] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: fixes crash on iterator_destroy on
+ uninitialized iterator (closes issue #16162) Reported by: krn
+
+ * channels/chan_iax2.c: changes calltoken debug messages from
+ LOG_NOTICE to LOG_DEBUG like they are supposed to be (closes
+ issue #16144) Reported by: aragon
+
+2009-10-29 18:11 +0000 [r226531] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_local.c, doc/localchannel.txt: Add an option to
+ enabling passing music on hold start and stop requests through
+ instead of acting on them in chan_local. (closes issue #14709)
+ Reported by: dimas
+
+2009-10-28 20:06 +0000 [r226377-226382] Leif Madsen <lmadsen at digium.com>
+
+ * configs/sip.conf.sample: Update documentation in sip.conf.sample.
+ Update the documentation in sip.conf.sample in order to make it
+ more clear that directmedia/canreinvite do not cause Asterisk to
+ ignore reINVITEs. It is only used to stop Asterisk from
+ generating a reINVITE, but does not stop it from accepting them
+ if necessary. (closes issue #15644) Reported by: lmadsen
+
+ * doc/channelvariables.txt: Update CALLINGSUBADDR channel variable
+ documentation. (closes issue #15734) Reported by: alecdavis
+ Patches: channelvariables.tex.diff.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis
+
+2009-10-28 18:02 +0000 [r226138-226304] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/linkedlists.h: Fix documentation (pointed out by
+ TheDavidFactor on #-dev)
+
+ * main/manager.c: Manager output is not always NULL-terminated, so
+ force a NULL at the end of the filestream. (closes issue #15495)
+ Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
+ by tilghman (license 14) Tested by: pdf
+
+2009-10-26 22:13 +0000 [r225957] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: detect
+ ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi * Set
+ OSARCH to linux-gnu even if host_os is linux-gnueabi * When
+ checking if we are Linux, check OSARCH rather than host_os The
+ newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather
+ than 'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even
+ in such a case. OSARCH is tested for the value of 'linux-gnu' in
+ one or two places in the tree. This patch also fixes the check
+ libcap to check for $OSARCH rather than $host_os . See also:
+ http://wiki.debian.org/ArmEabiPort
+
+2009-10-23 14:00 +0000 [r225581] Kevin P. Fleming <kpfleming at digium.com>
+
+ * Makefile: Don't force menuselect.makeopts to be rebuilt on every
+ build. For some reason the menuselect.makeopts file was listed as
+ PHONY in the Makefile, resulting in 'make' needing to rebuild it
+ for every build. This then resulted in the embedded module rules
+ being rebuilt on every build, which can be slow and is
+ unnecessary. This patch fixes the problem by properly allowing
+ 'make' to know when the menuselect.makeopts file needs to be
+ rebuilt (defining the proper dependencies).
+
+2009-10-22 21:51 +0000 [r225484] Leif Madsen <lmadsen at digium.com>
+
+ * doc/valgrind.txt, contrib/valgrind.supp (added): Clean valgrind
+ output by suppressing false errors. Update valgrind.txt
+ documentation and add valgrind.supp file in order to allow those
+ who are creating valgrind output to have less false errors in the
+ logfile. (closes issue #16007) Reported by: atis Patches:
+ valgrind.txt.diff uploaded by atis (license 242) asterisk2.supp
+ uploaded by atis (license 242) Tested by: atis, amorsen
+
+2009-10-21 20:58 +0000 [r225243] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: IAX2: VNAK loop caused by signaling frames
+ with no destination call number It is possible for the PBX thread
+ to queue up signaling frames before a destination call number is
+ received. This can result in signaling frames being sent out with
+ no destination call number. Since recent versions of Asterisk
+ require accurate destination callnumbers for all Full Frames,
+ this can cause a VNAK loop to occur. To resolve this no signaling
+ frames are sent until a destination callnumber is received, and
+ destination call numbers are now only required for iax_pvt
+ matching when the frame is an ACK. Review:
+ https://reviewboard.asterisk.org/r/413/
+
+2009-10-21 16:44 +0000 [r225169-225171] Russell Bryant <russell at digium.com>
+
+ * main/translate.c: Revert 225169, as this doesn't account for the
+ possibility of a list of frames.
+
+ * main/translate.c: Isolate the frame returned from
+ ast_translate().
+
+2009-10-21 16:02 +0000 [r225103-225105] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c, apps/app_meetme.c, include/asterisk/channel.h: Fix
+ documentation for ast_softhangup() and correct the misuse
+ thereof. (closes issue #16103) Reported by: majorbloodnok
+
+ * apps/app_voicemail.c: Suffix is not needed for a match
+
+2009-10-21 14:37 +0000 [r225032] David Vossel <dvossel at digium.com>
+
+ * configs/iax.conf.sample, channels/chan_sip.c,
+ configs/sip.conf.sample, channels/chan_iax2.c: IAX/SIP
+ shrinkcallerid option The shrinking of caller id removes '(', '
+ ', ')', non-trailing '.', and '-' from the string. This means
+ values such as 555.5555 and test-test result in 555555 and
+ testtest. There are instances, such as Skype integration, where a
+ specific value is passed via caller id that must be preserved
+ unmodified. This patch makes the shrinking of caller id optional
+ in chan_sip and chan_iax in order to support such cases. By
+ default this option is on to preserve previous expected behavior.
+ (closes issue #15940) Reported by: dimas Patches: v2-15940.patch
+ uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c
+ uploaded by dvossel (license 671) Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/408/
+
+2009-10-21 02:59 +0000 [r224931] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/translate.h, main/dsp.c, main/frame.c,
+ main/translate.c, include/asterisk/dsp.h, codecs/codec_dahdi.c,
+ include/asterisk/frame.h: Isolate frames returned from a DSP
+ instance or codec translator. The reasoning for these changes are
+ the same as what I wrote in the commit message for rev 222878.
+
+2009-10-20 22:07 +0000 [r224855] Tilghman Lesher <tlesher at digium.com>
+
+ * main/audiohook.c: Pay attention to the return value of the
+ manipulate function. While this looks like an optimization, it
+ prevents a crash from occurring when used with certain audiohook
+ callbacks (diagnosed with SVN trunk, backported to 1.4 to keep
+ the source consistent across versions).
+
+2009-10-20 17:46 +0000 [r224773] Joshua Colp <jcolp at digium.com>
+
+ * res/res_features.c: Add support for relaying early media in the
+ features attended transfer option. (closes issue #14828) Reported
+ by: licedey
+
+2009-10-19 23:44 +0000 [r224670] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/rtp.c: Correct timestamp calculations when RTP sample rates
+ over 8kHz are used. While testing some endpoints that support
+ 16kHz and 32kHz sample rates, some log messages were generated
+ due to calc_rxstamp() computing timestamps in a way that produced
+ odd results, so this patch sanitizes the result of the
+ computations.
+
+2009-10-19 19:47 +0000 [r224565] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_dial.c: Do not attempt early media bridging (ie: direct
+ RTP setup) if options are enabled that should prevent it. (closes
+ issue #14763) Reported by: cupotka
+
+2009-10-17 01:32 +0000 [r224330] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Fix stale caller id data from being
+ reported in AMI NewChannel event The problem here is that
+ chan_dahdi is designed in such a way to set certain values in the
+ dahdi_pvt only once. One of those such values is the configured
+ caller id data in chan_dahdi.conf. For PRI, the configured caller
+ id data could be overwritten during a call. Instead of saving the
+ data and restoring, it was decided that for all non-analog
+ channels it was simply best to not set the configured caller id
+ in the first place and also clear it at the end of the call.
+ (closes issue #15883) Reported by: jsmith
+
+2009-10-16 20:25 +0000 [r224260] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Never released PRI channels when using
+ Busy() or Congestion() dialplan apps. When the Busy() or
+ Congestion() application is used towards ISDN (an ISDN progress
+ is sent), the responding ISDN Disconnect or Release may contain
+ the ISDN cause user busy or one of the congestion causes. In
+ chan_dahdi.c these causes will only set the needbusy or
+ needcongestion flags and not activate the softhangup procedure.
+ Unfortunately only the latter can interrupt the endless wait loop
+ of Busy()/Congestion(). Result: PRI channels staying in state
+ busy for the rest of asterisk life or until the other end times
+ out and forces the call to clear. (in issue 0014292) Reported by:
+ tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso
+ (license 564) (This patch is unrelated to the issue.)
+
+2009-10-13 20:58 +0000 [r223955] Jean Galarneau <jgalarneau at digium.com>
+
+ * channels/chan_dahdi.c: Fix PRI timer T309 operation
+
+2009-10-12 23:12 +0000 [r223804] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_dial.c: Ensure ringing continues for branched calls
+ after progress is received While waiting for an answer, don't
+ send progress for branched calls for which ringing was sent.
+ (closes issue #15028) Reported by: fnordian
+
+2009-10-12 15:30 +0000 [r223692] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c: Remove automatic switching from T.38 to
+ voice mode in chan_sip. chan_sip has some code to automatically
+ switch from T.38 mode to voice mode when a voice frame is written
+ to the channel while it is in T.38 mode; this was intended to
+ handle the situation when a FAX transmission has ended and the
+ channel is not yet hung up, but is causing problems at the
+ beginning of FAX sessions as well when there are still voice
+ frames 'in flight' at the time the T.38 negotiation completes.
+ This patch removes the automatic switchover. (issue #16025)
+ Reported by: jamicque
+
+2009-10-11 18:34 +0000 [r223485-223550] Russell Bryant <russell at digium.com>
+
+ * apps/app_queue.c: Remove a duplicate ao2_iterator_destroy().
+
+ * main/autoservice.c: Remove some unnecessary code.
+
+ * main/autoservice.c: Don't use data outside of its scope. The
+ purpose of this code was to have a hangup frame put on the list
+ of deferred frames. However, the code that read the hangup frame
+ was outside of the scope of where the hangup frame was declared.
+
+2009-10-09 18:20 +0000 [r223225] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c: Signal timeouts by returning AST_CONTROL_RINGING
+ when originating calls. (closes issue #15104) Reported by:
+ nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
+ (license 96) Tested by: nblasgen, mnicholson
+
+2009-10-09 18:17 +0000 [r223213] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_dial.c: Fix potential memory leak in app_dial.c
+
+2009-10-09 17:52 +0000 [r223142-223205] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: fixes sip registration using authuser in
+ user.conf (closes issue #14954) Reported by: tornblad Tested by:
+ mmichelson, tornblad, dvossel
+
+ * channels/chan_sip.c: 'auth=' did not parse md5 secret correctly
+ (closes issue https://issues.asterisk.org/view.php?id=15949)
+ Reported by: ebroad Patches: authparsefix.patch uploaded by
+ ebroad (license 878) 15949_trunk.diff uploaded by dvossel
+ (license 671) Tested by: ebroad
+
+2009-10-08 19:45 +0000 [r222878] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/file.h, main/frame.c, main/file.c,
+ include/asterisk/frame.h: Make filestream frame handling safer by
+ isolating frames before returning them. This patch is related to
+ a number of issues on the bug tracker that show crashes related
+ to freeing frames that came from a filestream. A number of fixes
+ have been made over time while trying to figure out these
+ problems, but there re still people seeing the crash. (Note that
+ some of these bug reports include information about other
+ problems. I am specifically addressing the filestream frame crash
+ here.) I'm still not clear on what the exact problem is. However,
+ what is _very_ clear is that we have seen quite a few problems
+ over time related to unexpected behavior when we try to use
+ embedded frames as an optimization. In some cases, this
+ optimization doesn't really provide much due to improvements made
+ in other areas. In this case, the patch modifies filestream
+ handling such that the embedded frame will not be returned.
+ ast_frisolate() is used to ensure that we end up with a
+ completely mallocd frame. In reality, though, we will not
+ actually have to malloc every time. For filestreams, the frame
+ will almost always be allocated and freed in the same thread.
+ That means that the thread local frame cache will be used. So,
+ going this route doesn't hurt. With this patch in place, some
+ people have reported success in not seeing the crash anymore.
+ (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon
+ Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell
+ (license 2) Tested by: aragon, russell (closes issue #15817)
+ Reported by: zerohalo Tested by: zerohalo (closes issue #15845)
+ Reported by: marhbere Review:
+ https://reviewboard.asterisk.org/r/386/
+
+2009-10-08 19:45 +0000 [r222877] David Vossel <dvossel at digium.com>
+
+ * main/netsock.c, include/asterisk/netsock.h: fixes an
+ ast_netsock_list memory leak. ABE-1998 Review:
+ https://reviewboard.asterisk.org/r/395/
+
+2009-10-08 16:33 +0000 [r222691-222797] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn_config.c: Fix memory leak if chan_misdn config
+ parameter is repeated. Memory leak when the same config option is
+ set more than once in an misdn.conf section. Why must this be
+ considered? Templates! Defining a template with default port
+ options and later adding to or overriding some of them. Patches:
+ memleak-misdn.patch JIRA ABE-1998
+
+ * channels/chan_misdn.c: chan_misdn.c:process_ast_dsp() memory leak
+ misdn.conf: astdtmf must be set to "yes". With "no", buffer loss
+ does not occur. The translated frame "f2" when passing through
[... 26275 lines stripped ...]
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