[asterisk-commits] lmadsen: tag 1.4.23-testing r168838 - /tags/1.4.23-testing/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jan 16 12:58:00 CST 2009
Author: lmadsen
Date: Fri Jan 16 12:58:00 2009
New Revision: 168838
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=168838
Log:
Importing files for 1.4.23-testing release
Added:
tags/1.4.23-testing/.lastclean (with props)
tags/1.4.23-testing/.version (with props)
tags/1.4.23-testing/ChangeLog (with props)
Added: tags/1.4.23-testing/.lastclean
URL: http://svn.digium.com/svn-view/asterisk/tags/1.4.23-testing/.lastclean?view=auto&rev=168838
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--- tags/1.4.23-testing/ChangeLog (added)
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+2009-01-16 Leif Madsen <lmadsen at digium.com>
+
+ * Creating a testing release. THIS IS NOT FOR PUBLIC USE!
+
+2009-01-16 18:41 +0000 [r168828] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/say.h, apps/app_voicemail.c, main/say.c: Fix the
+ conjugation of Russian and Ukrainian languages. (related to issue
+ #12475) Reported by: chappell Patches: vm_multilang.patch
+ uploaded by chappell (license 8)
+
+2009-01-16 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.23-rc4 released.
+
+2009-01-16 00:19 +0000 [r168745] Steve Murphy <murf at digium.com>
+
+ * pbx/pbx_ael.c: This patch fixes a problem where a goto (or jump,
+ in this case) fails a consistency check because it can't find a
+ matching extension. The problem was a missing instruction to end
+ the range notation in the code where it converts the pattern into
+ a regex and uses the regex code to determine the match. I tested
+ using the AEL code the user supplied, and now, the consistency
+ check passes. (closes issue #14141) Reported by: dimas
+
+2009-01-15 18:43 +0000 [r168721] Olle Johansson <oej at edvina.net>
+
+ * configs/extconfig.conf.sample: Meetme actually has realtime but
+ wasn't documented
+
+2009-01-15 18:22 +0000 [r168716] Terry Wilson <twilson at digium.com>
+
+ * res/res_features.c: Convert call to park_call_full to
+ masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE
+ return value, we need to use masqueraded parking, otherwise we
+ will try to call ast_hangup() in __pbx_run() and in
+ do_parking_thread() and then promptly crash. (closes issue
+ #14215) Reported by: waverly360 Tested by: otherwiseguy (closes
+ issue #14228) Reported by: kobaz Tested by: otherwiseguy
+
+2009-01-15 01:20 +0000 [r168633] Tilghman Lesher <tlesher at digium.com>
+
+ * /: Blocked revision 168632 from /branches/1.2: 1.2 regression on
+ security fix AST-2009-001 (Closes issue #14238)
+
+2009-01-15 00:11 +0000 [r168628] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Fix some crashes from bad datastore handling in
+ app_queue.c * The queue_transfer_fixup function was searching for
+ and removing the datastore from the incorrect channel, so this
+ was fixed. * Most datastore operations regarding the
+ queue_transfer datastore were being done without the channel
+ locked, so proper channel locking was added, too. (closes issue
+ #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by
+ putnopvut (license 60) Tested by: ZX81, festr
+
+2009-01-14 21:48 +0000 [r168622] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn/isdn_lib.c: * Fixed create_process() allocation of
+ process ID values. The allocated process IDs could overflow their
+ respective NT and TE fields. Affects outgoing calls.
+
+2009-01-14 20:52 +0000 [r168614] Sean Bright <sean.bright at gmail.com>
+
+ * contrib/scripts/autosupport: Update autosupport script to supply
+ info for both Zaptel and DAHDI in 1.4 and be sure to run
+ dahdi_test in 1.6.x and trunk instead of zttest. (closes issue
+ #14132) Reported by: dsedivec Patches:
+ asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638)
+ asterisk-trunk-autosupport.patch uploaded by dsedivec (license
+ 638)
+
+2009-01-14 19:34 +0000 [r168608] Steve Murphy <murf at digium.com>
+
+ * apps/app_page.c: app_page was failing to compile in dev-mode on
+ my gcc-4.2.4 system. This change gets rid of the warning.
+
+2009-01-14 19:02 +0000 [r168603] Tilghman Lesher <tlesher at digium.com>
+
+ * main/udptl.c: Don't read into a buffer without first checking if
+ a value is beyond the end. (closes issue #13600) Reported by:
+ atis Patches: 20090106__bug13600.diff.txt uploaded by Corydon76
+ (license 14) Tested by: atis
+
+2009-01-14 16:19 +0000 [r168598] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_agent.c: Fix a logic error I found while searching
+ through chan_agent.c I found that the allow_multiple_logins
+ function would never return 0 due to an incorrect comparison
+ being used when traversing the list of agents. While I was
+ modifying this function, I also did a little bit of coding
+ guidelines cleanup, too.
+
+2009-01-14 01:27 +0000 [r168593] Terry Wilson <twilson at digium.com>
+
+ * apps/app_page.c: Don't overflow when paging more than 128
+ extensions The number of available slots for calls in app_page
+ was hardcoded to 128. Proper bounds checking was not in place to
+ enforce this limit, so if more than 128 extensions were passed to
+ the Page() app, Asterisk would crash. This patch instead
+ dynamically allocates memory for the ast_dial structures and
+ removes the (non-functional) arbitrary limit. This issue would
+ have special importance to anyone who is dynamically creating the
+ argument passed to the Page application and allowing more than
+ 128 extensions to be added by an outside user via some external
+ interface. The patch posted by a_villacis was slightly modified
+ for some coding guidelines and other cleanups. Thanks,
+ a_villacis! (closes issue #14217) Reported by: a_villacis
+ Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch
+ uploaded by a (license 660) Tested by: otherwiseguy
+
+2009-01-13 19:13 +0000 [r168561] Russell Bryant <russell at digium.com>
+
+ * main/indications.c, main/channel.c, apps/app_read.c,
+ channels/chan_misdn.c, funcs/func_channel.c,
+ include/asterisk/indications.h, apps/app_disa.c, main/app.c,
+ res/snmp/agent.c, include/asterisk/channel.h,
+ res/res_indications.c: Revert unnecessary indications API change
+ from rev 122314
+
+2009-01-13 18:34 +0000 [r168551] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Don't pass a value with a side effect to a
+ macro (closes issue #14176) Reported by: paraeco Patches:
+ chan_sip.c.diff uploaded by paraeco (license 658)
+
+2009-01-13 17:48 +0000 [r168546] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_logic.c: If either conditional is NULL, don't try
+ copying it. (closes issue #14226) Reported by: caspy Patches:
+ 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
+
+2009-01-12 21:42 +0000 [r168507-168516] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_agi.c: (closes issue #13881) Reported by: hoowa Update
+ the app CDR field for AGI commands that are not executing an
+ application via "exec".
+
+ * channels/chan_agent.c: (closes issue #12269) Reported by: IgorG
+ Tested by: denisgalvao This gits rid of the notion of an
+ owning_app allowing the request and hangup to be initiated by
+ different threads. Originating from an active agent channel
+ requires this. The implementation primarily changes __login_exec
+ to wait on a condition variable rather than a lock. Review:
+ http://reviewboard.digium.com/r/35/
+
+2009-01-12 14:58 +0000 [r168482] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: I am reverting the fix made in revision
+ 168128 (and its upward merges) after being contacted by Olle
+ Johansson and being shown how this fix is incorrect. Thanks to
+ Olle for clearing this up for me.
+
+2009-01-12 14:57 +0000 [r168480] Russell Bryant <russell at digium.com>
+
+ * configs/indications.conf.sample: s/ringdance/ringcadence/ for
+ Bulgaria
+
+2009-01-10 20:47 +0000 [r168267-168382] Kevin P. Fleming <kpfleming at digium.com>
+
+ * README: small commit to test new server
+
+ * README: small commit to test new server
+
+ * sounds/Makefile: update to use new sound file packages that
+ include license files
+
+2009-01-09 22:14 +0000 [r168198] Russell Bryant <russell at digium.com>
+
+ * res/res_musiconhold.c: Make this compile for mvanbaak
+
+2009-01-09 21:28 +0000 [r168191] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: * Fix for JIRA AST-175/ABE-1757 *
+ Miscellaneous doxygen comments added.
+
+2009-01-09 20:08 +0000 [r168128] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Add check_via calls to more request handlers
+ INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not
+ checking the topmost Via to determine where to send the response.
+ Adding check_via calls to those request handlers solves this.
+ (closes issue #13071) Reported by: baron Patches: check_via.patch
+ uploaded by baron (license 531) Tested by: baron
+
+2009-01-08 22:08 +0000 [r167840] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_agi.c: Don't truncate database results at 255 chars.
+ (closes issue #14069) Reported by: evandro Patches:
+ 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
+
+2009-01-08 17:24 +0000 [r167620-167714] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c: remove an unnecessary argument to
+ queue_request()
+
+ * channels/chan_sip.c: When a SIP request or response arrives for a
+ dialog with an associated Asterisk channel, and the lock on that
+ channel cannot be obtained because it is held by another thread,
+ instead of dropping the request/response, queue it for later
+ processing when the channel lock becomes available.
+ http://reviewboard.digium.com/r/117/
+
+2009-01-07 22:35 +0000 [r167432-167566] Russell Bryant <russell at digium.com>
+
+ * main/file.c: Fix the last couple of places where free() was
+ improperly used directly.
+
+ * main/file.c: Don't fclose() the file early, the filestream
+ destructor will handle it.
+
+ * main/file.c: Only try to close the file if one was actually
+ opened
+
+ * main/file.c: Don't use free() directly. This caused a crash since
+ ast_filestream is now an ao2 object. Reported by JunK-Y on IRC,
+ #asterisk-dev
+
+ * main/indications.c: Treat an empty string the same way as a NULL
+ country argument. In passing, simplify the handling of returning
+ a default tone zone.
+
+2009-01-06 21:35 +0000 [r167299] Mark Michelson <mmichelson at digium.com>
+
+ * main/db.c: Use the correct variable when creating the format
+ string (closes issue #14177) Reported by: nic_bellamy Patches:
+ asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic
+ (license 299)
+
+2009-01-06 20:48 +0000 [r167260] Tilghman Lesher <tlesher at digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 167259 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06
+ Jan 2009) | 2 lines Security fix AST-2009-001. ........
+
+2009-01-05 16:51 +0000 [r167179] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: A couple of changes to T.38 SDP attribute
+ handling There are some boolean attributes for T.38 such as
+ T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
+ T38FaxTranscodingJBIG. By simply being present, we should treat
+ these as a "true" value. The current code, however, was requiring
+ a 1 or 0 as the value of the attribute in order to parse it. This
+ is due to the fact that there are some T.38 endpoints and
+ gateways that also transmit this information incorrectly. This
+ patch follows the "be liberal in what you accept and strict in
+ what you send" philosophy by accepting both the correctly- and
+ incorrectly-formatted attributes, but only sending information as
+ it is supposed to be sent. It was also discovered that a
+ particular type of T.38 gateway sends some non-standard T.38 SDP
+ attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate,
+ it used T38MaxDatagram and T38FaxMaxRate respectively. We now
+ will properly accept these attributes as well. Note that there
+ are a lot of patches cited in the below commit message template.
+ This is because the person who submitted these patches is an
+ awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes
+ issue #13976) Reported by: linulin Patches:
+ chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov
+ (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded
+ by arcivanov (license 648)
+ chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov
+ (license 648) Tested by: arcivanov
+
+2009-01-01 00:01 +0000 [r166953-167095] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_alsa.c: Repeat attempts to write when we receive
+ -EAGAIN from the driver, as detailed in the ALSA sample code (see
+ http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
+ Reported by: Jerry Geis (via the -users list) Fixed by: me
+ (license 14)
+
+ * channels/chan_local.c: Also inherit the musiconhold class.
+ (Closes #14153) Reported by: Jerry Geis, via the users list.
+ Patch by: me (license 14)
+
+2008-12-28 15:13 +0000 [r166772] Russell Bryant <russell at digium.com>
+
+ * channels/misdn_config.c: Use strncat() instead of an sprintf() in
+ which source and target buffers overlap
+ http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html
+
+2008-12-23 15:35 +0000 [r166592] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, channels/chan_iax2.c: Compile, even if both
+ DAHDI and Zaptel are not installed. (Closes issue #14120)
+
+2008-12-23 15:16 +0000 [r166568] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Fix a crash resulting from a datastore with
+ inheritance but no duplicate callback The fix for this is to
+ simply set the newly created datastore's data pointer to NULL if
+ it is inherited but has no duplicate callback. (closes issue
+ #14113) Reported by: francesco_r Patches: 14113.patch uploaded by
+ putnopvut (license 60) Tested by: francesco_r
+
+2008-12-23 04:05 +0000 [r166509] Tilghman Lesher <tlesher at digium.com>
+
+ * main/channel.c: Use the integer form of condition for integer
+ comparisons. (closes issue #14127) Reported by: andrew
+
+2008-12-22 20:56 +0000 [r166380] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_dahdi.c: Fix a deadlock relating to channel locks
+ and autoservice It has been discovered that if a channel is
+ locked prior to a call to ast_autoservice_stop, then it is likely
+ that a deadlock will occur. The reason is that the call to
+ ast_autoservice_stop has a check built into it to be sure that
+ the thread running autoservice is not currently trying to
+ manipulate the channel we are about to pull out of autoservice.
+ The autoservice thread, however, cannot advance beyond where it
+ currently is, though, because it is trying to acquire the lock of
+ the channel for which autoservice is attempting to be stopped.
+ The gist of all this is that a channel MUST NOT be locked when
+ attempting to stop autoservice on the channel. In this particular
+ case, the channel was locked by a call to ast_read. A call to
+ ast_exists_extension led to autoservice being started and stopped
+ due to the existence of dialplan switches. It may be that there
+ are future commits which handle the same symptoms but in a
+ different location, but based on my looks through the code, it is
+ very rare to see a construct such as this one. (closes issue
+ #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded
+ by putnopvut (license 60) Tested by: rtrauntvein Review:
+ http://reviewboard.digium.com/r/107/
+
+2008-12-22 17:22 +0000 [r166262-166297] Russell Bryant <russell at digium.com>
+
+ * main/utils.c: Fix up timeout handling in ast_carefulwrite().
+
+ * include/asterisk/strings.h, res/res_musiconhold.c: Re-work ref
+ count handling of MoH classes using astobj2 to resolve crashes.
+ (closes issue #13566) Reported by: igorcarneiro Tested by:
+ russell Review: http://reviewboard.digium.com/r/106/
+
+2008-12-19 23:34 +0000 [r166157] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c, funcs/func_audiohookinherit.c (added),
+ channels/chan_sip.c, include/asterisk/audiohook.h,
+ main/audiohook.c, CHANGES: Backport of AUDIOHOOK_INHERIT for
+ Asterisk 1.4 (closes issue #13538) Reported by: mbit Patches:
+ 13538.patch uploaded by putnopvut (license 60) Tested by:
+ putnopvut
+
+2008-12-19 22:30 +0000 [r166093] Steve Murphy <murf at digium.com>
+
+ * apps/app_dial.c, res/res_features.c, include/asterisk/pbx.h,
+ apps/app_queue.c: This merges the masqpark branch into 1.4 These
+ changes eliminate the need for (and use of) the KEEPALIVE return
+ code in res_features.c; There are other places that use this
+ result code for similar purposes at a higher level, these appear
+ to be left alone in 1.4, but attacked in trunk. The reason these
+ changes are being made in 1.4, is that parking ends a channel's
+ life, in some situations, and the code in the bridge (and some
+ other places), was not checking the result code properly, and
+ dereferencing the channel pointer, which could lead to memory
+ corruption and crashes. Calling the masq_park function eliminates
+ this danger in higher levels. A series of previous commits have
+ replaced some parking calls with masq_park, but this patch puts
+ them ALL to rest, (except one, purposely left alone because a
+ masquerade is done anyway), and gets rid of the code that tests
+ the KEEPALIVE result, and the NOHANGUP_PEER result codes. While
+ bug 13820 inspired this work, this patch does not solve all the
+ problems mentioned there. I have tested this patch (again) to
+ make sure I have not introduced regressions. Crashes that
+ occurred when a parked party hung up while the parking party was
+ listening to the numbers of the parking stall being assigned, is
+ eliminated. These are the cases where parking code may be
+ activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to
+ parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip
+ xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via
+ manager. The interesting testing cases for parking are: I. A
+ calls B, A parks B a. B hangs up while A is getting the numbers
+ announced. b. B hangs up after A gets the announcement, but
+ before the parking time expires c. B waits, time expires, A is
+ redialed, A answers, B and A are connected, after which, B hangs
+ up. d. C picks up B while still in parking lot. II. A calls B, B
+ parks A a. A hangs up while B is getting the numbers announced.
+ b. A hangs up after B gets the announcement, but before the
+ parking time expires c. A waits, time expires, B is redialed, B
+ answers, A and B are connected, after which, A hangs up. d. C
+ picks up A while still in parking lot. Testing this throroughly
+ involves acting all the permutations of I and II, in situations
+ 1,2,3, and 4. Since I added a few more changes (ALL references to
+ KEEPALIVE in the bridge code eliimated (I missed one earlier), I
+ retested most of the above cases, and no crashes. H-extension
+ weirdness. Current h-extension execution is not completely
+ correct for several of the cases. For the case where A calls B,
+ and A parks B, the 'h' exten is run on A's channel as soon as the
+ park is accomplished. This is expected behavior. But when A calls
+ B, and B parks A, this will be current behavior: After B parks A,
+ B is hung up by the system, and the 'h' (hangup) exten gets run,
+ but the channel mentioned will be a derivative of A's... Thus, if
+ A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on
+ channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info
+ will be those relating to Channel A. And, in the case where A is
+ reconnected to B after the park time expires, when both parties
+ hang up after the joyful reunion, no h-exten will be run at all.
+ In the case where C picks up A from the parking lot, when either
+ A or C hang up, the h-exten will be run for the C channel. CDR's
+ are a separate issue, and not addressed here. As to WHY this
+ strange behavior occurs, the answer lies in the procedure
+ followed to accomplish handing over the channel to the parking
+ manager thread. This procedure is called masquerading. In the
+ process, a duplicate copy of the channel is created, and most of
+ the active data is given to the new copy. The original channel
+ gets its name changed to XXX<ZOMBIE> and keeps the PBX
+ information for the sake of the original thread (preserving its
+ role as a call originator, if it had this role to begin with),
+ while the new channel is without this info and becomes a call
+ target (a "peer"). In this case, the parking lot manager thread
+ is handed the new (masqueraded) channel. It will not run an
+ h-exten on the channel if it hangs up while in the parking lot.
+ The h exten will be run on the original channel instead, in the
+ original thread, after the bridge completes. See bug 13820 for
+ our intentions as to how to clean up the h exten behavior.
+ Review: http://reviewboard.digium.com/r/29/
+
+2008-12-19 19:48 +0000 [r165991] Jeff Peeler <jpeeler at digium.com>
+
+ * include/asterisk/dahdi_compat.h, main/asterisk.c, main/channel.c,
+ apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_meetme.c,
+ apps/app_dahdiscan.c, codecs/codec_dahdi.c,
+ res/res_musiconhold.c, channels/chan_iax2.c: (closes issue
+ #13480) Reported by: tzafrir Replace a bunch of if defined checks
+ for Zaptel/DAHDI through several new defines in dahdi_compat.h.
+ This removes a lot of code duplication. Example from bug: #ifdef
+ HAVE_ZAPTEL fd = open("/dev/zap/pseudo", O_RDWR); #else fd =
+ open("/dev/dahdi/pseudo", O_RDWR); #endif is replaced with: fd =
+ open(DAHDI_FILE_PSEUDO, O_RDRW);
+
+2008-12-19 15:03 +0000 [r165796-165889] Russell Bryant <russell at digium.com>
+
+ * apps/app_chanspy.c: Ensure that the chanspy datastore is fully
+ initialized. This patch resolved some random crash issues
+ observed by a user on a BSD system (closes issue #14111) Reported
+ by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281)
+
+ * main/utils.c: Make ast_carefulwrite() be more careful. This patch
+ handles some additional cases that could result in partial writes
+ to the file description. This was done to address complaints
+ about partial writes on AMI. (issue #13546) (more changes needed
+ to address potential problems in 1.6) Reported by: srt Tested by:
+ russell Review: http://reviewboard.digium.com/r/99/
+
+2008-12-18 21:14 +0000 [r165767] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Add mutexes around accesses to the IMAP
+ library interface. This prevents certain crashes, especially when
+ shared mailboxes are used. (closes issue #13653) Reported by:
+ howardwilkinson Patches:
+ asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by
+ howardwilkinson (license 590) Tested by: jpeeler
+
+2008-12-18 18:52 +0000 [r165661] Russell Bryant <russell at digium.com>
+
+ * res/res_musiconhold.c: Set the process group ID on the MOH
+ process so that all children will get killed (closes issue
+ #14099) Reported by: caspy Patches:
+ res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license
+ 645)
+
+2008-12-18 17:11 +0000 [r165537-165591] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: Only care about a compatible codec for early bridging
+ if we are actually bridging to another channel. If we are not we
+ actually want to bring the audio back to us. (closes issue
+ #13545) Reported by: davidw
+
+ * apps/app_followme.c: Do not crash if we are not passed in a
+ followme id. (closes issue #14106) Reported by: ys Patches:
+ app_followme.c.2.diff uploaded by ys (license 281)
+
+2008-12-17 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.23-rc3 released.
+
+2008-12-17 21:14 +0000 [r165317] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_macro.c: Reverse the fix from issue #6176 and add proper
+ handling for that issue. (Closes issue #13962, closes issue
+ #13363) Fixed by myself (license 14)
+
+2008-12-17 20:51 +0000 [r164977-165255] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_meetme.c, apps/app_realtime.c, apps/app_directory.c,
+ apps/app_queue.c: Fix some memory leaks found while looking at
+ how realtime configs are handled. Also cleaned up some coding
+ guidelines violations in app_realtime.c, mostly related to
+ spacing
+
+ * channels/chan_sip.c: After looking through SIP registration code
+ most of the day, this is one of the few things I could find that
+ was just plain wrong. Even though it probably isn't possible for
+ it to happen, it seems weird to have code that checks if a
+ pointer is NULL and then immediately dereferences that pointer if
+ it was NULL.
+
+2008-12-16 21:38 +0000 [r164672-164881] Russell Bryant <russell at digium.com>
+
+ * main/utils.c: Fix an issue where DEBUG_THREADS may erroneously
+ report that a thread is exiting while holding a lock. If the last
+ lock attempt was a trylock, and it failed, it will still be in
+ the list of locks so that it can be reported. (closes issue
+ #13219) Reported by: pj
+
+ * apps/app_macro.c: Do not dereference the channel if
+ AST_PBX_KEEPALIVE has been returned. This is a bug I noticed
+ while looking at the code for app_macro. This return code means
+ that another thread has assumed ownership of the channel and it
+ can no longer be touched. (I hate this return code with a
+ passion, by the way.)
+
+ * main/manager.c: Add "restart gracefully" to the AMI blacklist of
+ CLI commands. "module unload" was already identified as a command
+ that can not be used from the AMI. "restart gracefully"
+ effectively unloads all modules, and will run in to the same
+ problems. (closes issue #13894) Reported by: kernelsensei
+
+ * include/asterisk/threadstorage.h, main/threadstorage.c: Fix
+ memory leak and invalid reporting issues with DEBUG_THREADLOCALS.
+ One issue was that the ast_mutex_* API was being used within the
+ context of the thread local data destructors. We would go off and
+ allocate more thread local data while the pthread lib was in the
+ middle of destroying it all. This led to a memory leak. Another
+ issue was an invalid argument being provided to the the
+ object_add API call. (closes issue #13678) Reported by: ys Tested
+ by: Russell
+
+ * channels/chan_sip.c: Fix a memory leak related to the use of the
+ "setvar" configuration option. The problem was that these
+ variables were being appended to the list of vars on the sip_pvt
+ every time a re-registration or re-subscription came in. Since
+ it's just a waste of memory to put them there unless the request
+ was an INVITE, then the fix is to check the request type before
+ copying the vars. (closes issue #14037) Reported by: marvinek
+ Tested by: russell
+
+2008-12-16 15:15 +0000 [r164634] Steve Murphy <murf at digium.com>
+
+ * main/pbx.c: I added a sentence to clarify why - and ' ' are
+ ignored in patterns as per bug 14076. Leif says he'll put some
+ stuff about it in the extensions.conf sample, etc.
+
+2008-12-16 14:28 +0000 [r164605] Russell Bryant <russell at digium.com>
+
+ * res/res_musiconhold.c: Don't try to change working directory if a
+ directory was not configured. (closes issue #14089) Reported by:
+ caspy
+
+2008-12-15 19:53 +0000 [r164416-164422] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/pbx.h: Add the deadlock note to
+ ast_spawn_extension as well
+
+ * include/asterisk/channel.h, include/asterisk/pbx.h: Add notes to
+ autoservice and pbx doxygen regarding a potential deadlock
+ scenario so that it is avoided in the future
+
+2008-12-15 18:11 +0000 [r164204-164350] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Do not try to unlock a non-existant channel
+ if the transfer fails. (closes issue #13800) Reported by: dwagner
+ Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety
+ (license 608)
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/channel.h: Use autoconf logic to determine
+ whether the system has timersub or not. Do not blindly assume
+ Solaris does not. (closes issue #13838) Reported by: ano
+
+ * apps/app_dial.c: Can we try not to assign an unsigned int to -1?
+ (closes issue #14074) Reported by: wetwired
+
+2008-12-15 14:31 +0000 [r164201] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, res/res_features.c: Handle a case where a call
+ can be bridged to a channel that is still ringing. The issue that
+ was reported was about a case where a RINGING channel got
+ redirected to an extension to pick up a call from parking. Once
+ the parked call got taken out of parking, it heard silence until
+ the other side answered. Ideally, the caller that was parked
+ would get a ringing indication. This patch fixes this case so
+ that the caller receives ringback once it comes out of parking
+ until the other side answers. The fixes are: - Make sure we
+ remember that a channel was an outgoing channel when doing a
+ masquerade. This prevents an erroneous ast_answer() call on the
+ channel, which causes a bogus 200 OK to be sent in the case of
+ SIP. - Add some additional comments to explain related parts of
+ code. - Update the handling of the ast_channel visible_indication
+ field. Storing values that are not stateful is pointless. Control
+ frames that are events or commands should be ignored. - When a
+ bridge first starts, check to see if the peer channel needs to be
+ given ringing indication because the calling side is still
+ ringing. - Rework ast_indicate_data() a bit for the sake of
+ readability. (closes issue #13747) Reported by: davidw Tested by:
+ russell Review: http://reviewboard.digium.com/r/90/
+
+2008-12-13 23:22 +0000 [r164082] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_dial.c: Change the default calldurationlimit from the
+ special value 0 to -1, so we can better detect an exceptional
+ case. This follows on to the changes made in revision 156386.
+ Related to issue #13851. (closes issue #13974) Reported by:
+ paradise Patches: 20081208__bug13974.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: file, blitzrage, ZX81
+
+2008-12-12 22:20 +0000 [r163785] Russell Bryant <russell at digium.com>
+
+ * /: Set the reviewboard:url property on 1.4, as well
+
+2008-12-12 22:03 +0000 [r163761] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, main/editline/read.c: Simple fix for Ctrl-C not
+ immediately exiting Asterisk, but also add a pointer inside
+ editline to look back to asterisk.c, so others don't spend as
+ much time as I did looking (in the wrong place) for the
+ appropriate function. Reported by: ZX81, via the #asterisk-users
+ channel Fixed by: me (license 14)
+
+2008-12-12 14:40 +0000 [r163448-163511] Russell Bryant <russell at digium.com>
+
+ * pbx/pbx_dundi.c: Specify uint32_t for variables storing a CRC32
+ so that it is actually 32 bits on 64-bit machines, as well.
+ (inspired by issue #13879)
+
+ * main/channel.c, main/autoservice.c, include/asterisk/channel.h:
+ Resolve issues that could cause DTMF to be processed out of
+ order. These changes come from team/russell/issue_12658 1) Change
+ autoservice to put digits on the head of the channel's frame
+ readq instead of the tail. If there were frames on the readq that
+ autoservice had not yet read, the previous code would have
+ resulted in out of order processing. This required a new API call
+ to queue a frame to the head of the queue instead of the tail. 2)
+ Change up the processing of DTMF in ast_read(). Some of the
+ problems were the result of having two sources of pending DTMF
+ frames. There was the dtmfq and the more generic readq. Both were
+ used for pending DTMF in various scenarios. Simplifying things to
+ only use the frame readq avoids some of the problems. 3) Fix a
+ bug where a DTMF END frame could get passed through when it
+ shouldn't have. If code set END_DTMF_ONLY in the middle of digit
+ emulation, and a digit arrived before emulation was complete,
+ digits would get processed out of order. (closes issue #12658)
+ Reported by: dimas Tested by: russell, file Review:
+ http://reviewboard.digium.com/r/85/
+
+2008-12-11 23:35 +0000 [r163383] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: When a Ctrl-C or Ctrl-D ends a remote console,
+ on certain shells, the terminal is messed up. By intercepting
+ those events with a signal handler in the remote console, we can
+ avoid those issues. (closes issue #13464) Reported by: tzafrir
+ Patches: 20081110__bug13464.diff.txt uploaded by Corydon76
+ (license 14) Tested by: blitzrage
+
+2008-12-11 22:44 +0000 [r163316] Matt Nicholson <mnicholson at digium.com>
+
+ * pbx/pbx_dundi.c: Clean up the dundi cache every 5 minutes.
+ (closes issue #13819) Reported by: adomjan Patches:
+ pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
+ dundi_clearecache3.diff uploaded by mnicholson (license 96)
+ Tested by: adomjan
+
+2008-12-11 21:46 +0000 [r163092-163253] Russell Bryant <russell at digium.com>
+
+ * funcs/func_strings.c, funcs/func_cut.c: Fix some observed
+ slowdowns in dialplan processing. The change is to remove
+ autoservice usage from dialplan functions that do not need it
+ because they do not perform operations that potentially block.
+ (closes issue #13940) Reported by: tbelder
+
+ * res/res_features.c: Fix an issue that made it so you could only
+ have a single caller executing a custom feature at a time. This
+ was especially problematic when custom features ran for any
+ appreciable amount of time. The fix turned out to be quite
+ simple. The dynamic features are now stored in a read/write list
+ instead of a list using a mutex. (closes issue #13478) Reported
+ by: neutrino88 Fix suggested by file
+
+2008-12-11 16:51 +0000 [r163088] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_agi.c: Don't wait forever, if there's a specified
+ recording timeout. (closes issue #13885) Reported by: bamby
+ Patches: res_agi.c.patch uploaded by bamby (license 430)
+
+2008-12-11 16:46 +0000 [r163080-163084] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Revert this cast to long. Using time_t here
+ causes build failures on a FreeBSD 32-bit build.
+
+ * apps/app_queue.c: Fix a potential crash due to unsafe datastore
+ handling. This patch also contains a conversion from using long
+ to time_t for representing times for a queue, as well as some
+ whitespace fixes. (closes issue #14060) Reported by: nivek
+ Patches: datastore_fixup.patch.corrected uploaded by nivek
+ (license 636) with slight modification from me Tested by: nivek
+
+2008-12-10 22:52 +0000 [r162874-162926] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_musiconhold.c: Oops, inverted logic for a strcasecmp
+ check. Pointed out by mmichelson, thanks!
+
+ * res/res_musiconhold.c: (closes issue #13229) Reported by:
+ clegall_proformatique Ensure that moh_generate does not return
+ prematurely before local_ast_moh_stop is called. Also, the sleep
+ in mp3_spawn now only occurs for http locations since it seems to
+ have been added originally only for failing media streams.
+
+2008-12-10 19:01 +0000 [r162738-162804] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix subscription based MWI up a bit. We only
+ want to put sip: at the beginning of the URI if it is not already
+ there and revert code to ignore destination check if subscribing
+ for MWI. (closes issue #12560) Reported by: vsauer Patches:
+ patch001.diff uploaded by ramonpeek (license 266)
+
+ * channels/chan_sip.c: When a SIP peer unregisters set the expiry
+ time back to 0 so that the 200 OK contains an expires of 0.
+ (closes issue #13599) Reported by: hjourdain Patches:
+ chan_sip.c.diff uploaded by hjourdain (license 583)
+
+2008-12-10 16:45 +0000 [r162671] Steve Murphy <murf at digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/ael/ael.flex: (closes issue #14022)
+ Reported by: wetwired Tested by: murf I checked, and I added a
+ mod to the trunk version of Asterisk that would make it 8-bit
+ transparent on 27 Nov 2007, but I made no such updates to 1.4. My
+ best guess is that 1.4 was released, and it was not appropriate
+ to commit an enhancement. But I'm going to add the same fixes to
+ 1.4 now, for the following reasons: 1. wetwired is correct; 1.4
+ is **mostly** 8-bit transparent now. This is because the lexical
+ token forming rules use . in most 'word' state continuances. It's
+ just the beginning of a 'word' that is picky. 2. Accepting 8-bit
+ chars in some places and not others leads to bug reports like
+ this.
+
+2008-12-10 16:44 +0000 [r162659-162670] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/stringfields.h: Update to stringfield handling
+ so that side-effects on parameters are not evaluated multiple
+ times. An example where this caused a problem was in chan_sip.c,
+ with the line ast_string_field_set(p, fromdomain, ++fromdomain);
+ This patch was originally uploaded to issue #13783 by jamessan.
+ While the issue was closed for other reasons, this patch is valid
+ and fixes a separate problem, and is thus being committed.
+
+ * channels/chan_sip.c: Revert fix for issue 13570. It has caused
+ more problems than it helped to fix. (closes issue #13783)
+ Reported by: navkumar (closes issue #14025) Reported by: ffs
+
+ * doc/misdn.txt: Add missing documentation to misdn.txt (closes
[... 21347 lines stripped ...]
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