[asterisk-commits] lmadsen: tag 1.8.3-rc1 r302176 - /tags/1.8.3-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jan 18 12:16:44 CST 2011
Author: lmadsen
Date: Tue Jan 18 12:16:39 2011
New Revision: 302176
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=302176
Log:
Importing files for 1.8.3-rc1 release.
Added:
tags/1.8.3-rc1/.lastclean (with props)
tags/1.8.3-rc1/.version (with props)
tags/1.8.3-rc1/ChangeLog (with props)
Added: tags/1.8.3-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.3-rc1/.lastclean?view=auto&rev=302176
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+2011-01-17 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.3-rc1 Released.
+
+2011-01-18 18:11 +0000 [r302174] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/features.c: Merged revisions 302173 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600
+ (Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011)
+ | 88 lines Issues with DTMF triggered attended transfers. Issue
+ #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in
+ features.conf for attended transfer). 3) A hears MOH. B dial
+ number C 4) C ringing. A hears MOH. 5) B hangup. A still hears
+ MOH. C ringing. 6) A hangup. C still ringing until
+ "atxfernoanswertimeout" expires. For v1.4 C will ring forever
+ until C answers the dead line. (Issue #17096) Problem: When A and
+ B hangup, C is still ringing. Issue #18395 SIP call limit of B is
+ 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C
+ ringing 4. Timeout waiting for C to answer 5. Recall to B fails
+ because B has reached its call limit. Because B reached its call
+ limit, it cannot do anything until the transfer it started
+ completes. Issue #17273 Same scenario as issue 18395 but party B
+ is an FXS port. Party B cannot do anything until the transfer it
+ started completes. If B goes back off hook before C answers, B
+ hears ringback instead of the expected dialtone. ********** Note
+ for the issue #17273 and #18395 fix: DTMF attended transfer works
+ within the channel bridge. Unfortunately, when either party A or
+ B in the channel bridge hangs up, that channel is not completely
+ hung up until the transfer completes. This is a real problem
+ depending upon the channel technology involved. For chan_dahdi,
+ the channel is crippled until the hangup is complete. Either the
+ channel is not useable (analog) or the protocol disconnect
+ messages are held up (PRI/BRI/SS7) and the media is not released.
+ For chan_sip, a call limit of one is going to block that endpoint
+ from any further calls until the hangup is complete. For party A
+ this is a minor problem. The party A channel will only be in this
+ condition while party B is dialing and when party B and C are
+ conferring. The conversation between party B and C is expected to
+ be a short one. Party B is either asking a question of party C or
+ announcing party A. Also party A does not have much incentive to
+ hangup at this point. For party B this can be a major problem
+ during a blonde transfer. (A blonde transfer is our term for an
+ attended transfer that is converted into a blind transfer. :))
+ Party B could be the operator. When party B hangs up, he assumes
+ that he is out of the original call entirely. The party B channel
+ will be in this condition while party C is ringing, while
+ attempting to recall party B, and while waiting between call
+ attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to
+ fix the problem. It will replace the party B channel technology
+ with a NULL channel driver to complete hanging up the party B
+ channel technology. The consequences of this code is that the 'h'
+ extension will not be able to access any channel technology
+ specific information like SIP statistics for the call.
+ ATXFER_NULL_TECH is not defined by default. ********** (closes
+ issue #17999) Reported by: iskatel Tested by: rmudgett JIRA
+ SWP-2246 (closes issue #17096) Reported by: gelo Tested by:
+ rmudgett JIRA SWP-1192 (closes issue #18395) Reported by:
+ shihchuan Tested by: rmudgett (closes issue #17273) Reported by:
+ grecco Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1047/ ........
+ ................
+
+2011-01-17 15:04 +0000 [r302005] Terry Wilson <twilson at digium.com>
+
+ * configs/sip.conf.sample: Document "encryption" option in
+ sip.conf.sample
+
+2011-01-14 21:09 +0000 [r301946] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c: Deadlock between dahdi_request() and
+ pri_dchannel() processing an incomming call. The
+ sig_pri_new_ast_channel() is called with the channel private lock
+ held when pri_dchannel() calls it and no channel private lock
+ held when dahdi_request() calls it. The use of pri_grab() in
+ sig_pri_new_ast_channel() could leave the channel private lock
+ held when it returns if the lock was not held before calling it.
+ Make sig_pri_new_ast_channel() just lock the PRI span lock
+ instead of using pri_grab(). It is safe to do this because
+ dahdi_request() does not have the channel private lock and the
+ deadlock potential with the PRI span lock is only between
+ pri_dchannel() and other threads.
+
+2011-01-14 20:11 +0000 [r301851] Brett Bryant <bbryant at digium.com>
+
+ * channels/chan_multicast_rtp.c: Changing previous revisions
+ 301845/301847 to use ast_sockaddr_setnull() instead of setting
+ the field manually to avoid uninitialized data. Review:
+ https://reviewboard.asterisk.org/r/1076/
+
+2011-01-14 20:05 +0000 [r301849] Andrew Latham <lathama at gmail.com>
+
+ * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to
+ function documentation. Fix amatuer type mistake
+
+2011-01-14 19:35 +0000 [r301845] Brett Bryant <bbryant at digium.com>
+
+ * channels/chan_multicast_rtp.c: Fix for a consistent MulticastRTP
+ channel driver crash due to use of unitilized data. (closes issue
+ #18290) (closes issue #18602) Reported by: voipgate, wybecom
+ Review: https://reviewboard.asterisk.org/r/1076/
+
+2011-01-14 19:35 +0000 [r301844] Andrew Latham <lathama at gmail.com>
+
+ * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to
+ function documentation.
+
+2011-01-14 17:32 +0000 [r301790] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_sip.c: Resolve deadlock involving REFER. Two fixes:
+ 1) One must always have the private unlocked before calling
+ pbx_builtin_setvar_helper to not invalidate locking order since
+ it locks the channel. 2) Unlock the channel before calling
+ pbx_find_extension, which starts and stops autoservice during the
+ lookup. The problem scenario as illustrated by the reporter:
+ Thread: do_monitor ----------------------- handle_request_do
+ handle_incoming handle_request_refer ast_parking_ext_valid
+ pbx_find_extension ast_autoservice_stop while (chan_list_state ==
+ as_chan_list_state) { usleep(1000); } Thread: autoservice_run
+ ----------------------- autoservice_run chan = ast_waitfor_n
+ ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple /
+ complex (depending on your system) ast_channel_lock(c[x]);
+ handle_request_do and schedule_process_request_queue locks the
+ owner if it exists. The autoservice thread is waiting for the
+ channel lock, which wasn't ever released since the do_monitor
+ thread was waiting for autoservice operations to complete. Solved
+ by unlocking the channel but keeping a reference to guarantee
+ safety. (closes issue #18403) Reported by: jthurman Patches:
+ 20110103-blind_deadlock.diff uploaded by jthurman (license 614)
+ issue18403.patch uploaded by jpeeler (license 325) Tested by:
+ jthurman
+
+2011-01-13 17:01 +0000 [r301731] Leif Madsen <lmadsen at digium.com>
+
+ * configs/phoneprov.conf.sample, /: Merged revisions 301730 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011)
+ | 7 lines Add static entry for split Polycom 332 firmware.
+ (closes issue #18607) Reported by: cjacobsen Patches:
+ polycom_331.diff uploaded by cjacobsen (license 1029) Tested by:
+ lathama ........
+
+2011-01-12 21:19 +0000 [r301683] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 301682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011)
+ | 9 lines Don't reject all SUBSCRIBE auth requests When merging
+ another SUBSCRIBE fix from 1.4, some braces were put in the wrong
+ place. This patch fixes that. (closes issue #18597) Reported by:
+ thsgmbh ........
+
+2011-01-12 18:51 +0000 [r301595] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/manager.c, /: Merged revisions 301594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r301594 | mnicholson | 2011-01-12 12:50:31 -0600
+ (Wed, 12 Jan 2011) | 15 lines Removed a usleep(1) that shouldn't
+ be necessary in session_do, and removed the ms_t member from the
+ mansession_session structure. Merged revisions 301591 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan
+ 2011) | 5 lines Don't store the thread id for the manager session
+ in the structure we pass to the thread for the manager session.
+ ABE-2543 ........ ................
+
+2011-01-12 18:12 +0000 [r301504] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, /: Merged revisions 301503 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600
+ (Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011)
+ | 12 lines Fix CPU spike when pressing DTMF after agent login.
+ The problem here is that DTMF was being continuously deferred and
+ requeued since ast_safe_sleep is called in a loop. There are
+ serveral other places in the code that sleeps and then loops in a
+ similar fashion. Because of this fact I opted to not defer DTMF
+ any more, which will not affect the original fix:
+ https://reviewboard.asterisk.org/r/674 (closes issue #18130)
+ Reported by: rgj ........ ................
+
+2011-01-12 16:05 +0000 [r301446] David Vossel <dvossel at digium.com>
+
+ * main/file.c: Removal of unused variables so Asterisk will
+ compile.
+
+2011-01-12 15:57 +0000 [r301444] Stefan Schmidt <sst at sil.at>
+
+ * Makefile: fix wrong text of rerun menuselect after user interface
+ warning the warning, if no user interface for menuselect warning
+ was found is not right. you have to rerun configure before make
+ menuselect after installing a proper user interface. (closes
+ issue #18594) Reported by: Dovid
+
+2011-01-12 00:26 +0000 [r301402] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * main/file.c: Call execl() directly for a better solution for
+ paths with spaces. (closes issue #18600) Reported by: ebroad
+ Patches: 20110111__issue18600__2.diff.txt uploaded by tilghman
+ (license 14)
+
+2011-01-11 19:16 +0000 [r301311] Paul Belanger <pabelanger at digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 301310 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301310 | pabelanger | 2011-01-11 14:14:31 -0500 (Tue, 11 Jan
+ 2011) | 2 lines Fix a logic issue when passing context ARG
+ ........
+
+2011-01-11 18:51 +0000 [r301308] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/utils.c, /: Merged revisions 301307 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r301307 | mnicholson | 2011-01-11 12:42:05 -0600
+ (Tue, 11 Jan 2011) | 11 lines Merged revisions 301305 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan
+ 2011) | 4 lines Prevent buffer overflows in ast_uri_encode()
+ ABE-2705 ........ ................
+
+2011-01-10 22:39 +0000 [r301263] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * main/strcompat.c: Little endian machines were not converted
+ properly. (closes issue #18583) Reported by: jcovert Patches:
+ 20110110__issue18583.diff.txt uploaded by tilghman (license 14)
+ Tested by: jcovert
+
+2011-01-09 21:40 +0000 [r301177-301221] Paul Belanger <pabelanger at digium.com>
+
+ * autoconf/ast_ext_lib.m4, /, configure, configure.ac: Merged
+ revisions 301220 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan
+ 2011) | 14 lines SOUND_CACHE_DIR now defaults to empty Sounds
+ files included in the Asterisk tarball were being ignored and
+ re-downloaded. Users wanting to cache the files can still
+ override the setting using the --with-sounds-cache option.
+ (closes issue #18589) Reported by: pabelanger Patches:
+ issue18589.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger Review: https://reviewboard.asterisk.org/r/1074/
+ ........
+
+ * apps/app_verbose.c, /: Merged revisions 301176 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301176 | pabelanger | 2011-01-08 16:58:24 -0500 (Sat, 08 Jan
+ 2011) | 7 lines Indicate log level argument for Log() is not
+ optional (closes issue #18586) Reported by: kshumard Patches:
+ app_verbose.c.patch uploaded by kshumard (license 92) ........
+
+2011-01-08 01:11 +0000 [r301134] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: The DTMF attended transfer feature cannot
+ callback a chan_dahdi BRI phone. The DAHDI ISDN channel name is
+ not dialable. Make a channel name like DAHDI/i3/400-12 dialable
+ when the sequence number is stripped off of the name.
+
+2011-01-07 20:53 +0000 [r301090] Jason Parker <jparker at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 301089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) |
+ 8 lines Initialize useropts/adminopts in case there is no column
+ in the realtime DB. (closes issue #18182) Reported by: dimas
+ Patches: v1-18182.patch uploaded by dimas (license 88) Tested by:
+ dimas ........
+
+2011-01-07 19:58 +0000 [r300955-301047] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 301046 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07
+ Jan 2011) | 8 lines Fix regression causing forwarding voicemails
+ to not work with file storage. I had actually already fixed this
+ in 295200 in 1.4 and thought it wasn't missing in the other
+ branches for some reason. (closes issue #18358) Reported by:
+ cabal95 ........
+
+ * apps/app_voicemail.c, /: Merged revisions 300951 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r300951 | jpeeler | 2011-01-07 11:23:37 -0600
+ (Fri, 07 Jan 2011) | 14 lines Merged revisions 300918 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011)
+ | 7 lines Ensure good bye prompt in voicemail is played at the
+ correct time. Specifically in the case of timing out but not
+ leaving voicemail nothing should be heard. And when leaving
+ voicemail it should be heard. ABE-2647 ........ ................
+
+2011-01-06 06:28 +0000 [r300798] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * addons/res_config_mysql.c: Don't destroy handle not created by
+ use (because the caller will). (closes issue #18526) Reported by:
+ makoto Patches: res-config-mysql-include.patch uploaded by makoto
+ (license 38) Tested by: makoto
+
+2011-01-05 20:54 +0000 [r300714] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c: Merged revision 300711 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed,
+ 05 Jan 2011) | 14 lines A call retrieved from hold may wind up
+ with no audio. If the retrieved call is natively bridged then the
+ call may not have any audio path. The following warning message
+ is given: "Failed to add <dfd> to conference <chan>/<chan>:
+ Invalid argument". * Open the media on a B channel when
+ pri_fixup_principle() moves the call from a no_b_channel channel
+ to a real channel. * Added lock protection while
+ pri_fixup_principle() moves a call from one private structure to
+ another. * Made some pri_fixup_principle() messages more
+ meaningful. ..........
+
+2011-01-05 18:56 +0000 [r300623] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * res/res_odbc.c, /: Merged revisions 300622 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r300622 | tilghman | 2011-01-05 12:54:58 -0600
+ (Wed, 05 Jan 2011) | 17 lines Merged revisions 300621 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011)
+ | 10 lines Use the sanity check in place of the
+ disconnect/connect cycle. The disconnect/connect cycle has the
+ potential to cause random crashes. (closes issue #18243) Reported
+ by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147)
+ Tested by: ks3 ........ ................
+
+2011-01-05 16:29 +0000 [r300575] Paul Belanger <pabelanger at digium.com>
+
+ * /, cdr/cdr_sqlite.c: Merged revisions 300574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r300574 | pabelanger | 2011-01-05 11:28:07 -0500 (Wed, 05 Jan
+ 2011) | 6 lines Change deprecated message to LOG_WARNING Also
+ removed latter part of message Discussed on #asterisk-dev
+ ........
+
+2011-01-04 21:53 +0000 [r300433-300521] Leif Madsen <lmadsen at digium.com>
+
+ * channels/chan_iax2.c, main/xmldoc.c, /, channels/chan_sip.c,
+ channels/chan_agent.c: Merged revisions 300520 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011)
+ | 9 lines Fix backwards and broken XML documentation. (closes
+ issue #18547) Reported by: jcovert Patches: xmldoc.c.patch
+ uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded
+ by jcovert (license 551) chan_sip.c.patch uploaded by jcovert
+ (license 551) chan_agent.c.patch uploaded by jcovert (license
+ 551) ........
+
+ * configs/users.conf.sample, /: Merged revisions 300431 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r300431 | lmadsen | 2011-01-04 15:00:29 -0600 (Tue, 04 Jan 2011)
+ | 7 lines Add some documentation to users.conf.sample. (closes
+ issue #18531) Reported by: lathama Patches:
+ users.conf.sample2.diff uploaded by lathama (license 1028) Tested
+ by: lathama ........
+
+2011-01-04 21:00 +0000 [r300430] Russell Bryant <russell at digium.com>
+
+ * contrib/scripts/autosupport, /, contrib/scripts/autosupport.8:
+ Merged revisions 300429 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r300429 | russell | 2011-01-04 14:59:56 -0600
+ (Tue, 04 Jan 2011) | 11 lines Merged revisions 300428 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011)
+ | 4 lines Update the autosupport script from Digium support.
+ (closes AST-395) ........ ................
+
+2011-01-04 19:45 +0000 [r300384] Leif Madsen <lmadsen at digium.com>
+
+ * phoneprov/000000000000.cfg: Update STAT() to use the comma
+ instead of the pipe. (closes issue #18503) Reported by: cjacobsen
+ Patches: old_separator.diff uploaded by cjacobsen (license 1029)
+ Tested by: lathama
+
+2011-01-04 17:54 +0000 [r300301] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 300298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r300298 | twilson | 2011-01-04 11:37:26 -0600
+ (Tue, 04 Jan 2011) | 22 lines Merged revisions 300216 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011)
+ | 15 lines Don't authenticate SUBSCRIBE re-transmissions This
+ only skips authentication on retransmissions that are already
+ authenticated. A similar method is already used for INVITES. This
+ is the kind of thing we end up having to do when we don't have a
+ transaction layer... (closes issue #18075) Reported by: mdu113
+ Patches: diff.txt uploaded by twilson (license 396) Tested by:
+ twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/
+ ........ ................
+
+2011-01-04 17:01 +0000 [r300214] Jan Kalab <pitlicek at gmail.com>
+
+ * res/res_calendar_exchange.c, res/res_calendar_icalendar.c: Memory
+ leaking in calendars ne_request_destroy() was missing in
+ icalendar and exchange calendar modules, causing memory leak.
+ (closes issue #18521) Review:
+ https://reviewboard.asterisk.org/r/1068/
+
+2011-01-03 23:14 +0000 [r300166] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/features.c: Merged revisions 300165 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r300165 | rmudgett | 2011-01-03 17:02:13 -0600 (Mon, 03 Jan 2011)
+ | 4 lines Use correct variable for atxfercallbackretries config
+ option. * Misc formatting changes. ........
+
+2011-01-03 13:14 +0000 [r300082] Leif Madsen <lmadsen at digium.com>
+
+ * pbx/pbx_dundi.c: Increase side of mapping response field. I've
+ increased the size of the response field in a DUNDi mapping
+ because of some documentation I'm writing. Previously it was set
+ to AST_MAX_EXTENSION which is only 80 characters, which is far
+ too small when you're using some dialplan functions to craft a
+ response. The example I'm using is: extensions =>
+ RegisteredDevices,0,SIP,dundi:very_awesome_password/${IF($[${DB_EXISTS(phones/${NUMBER}/device)}]?${DB(phones/${NUMBER}/device)}:None)},nopartial
+
+2010-12-29 22:02 +0000 [r299989] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * apps/app_voicemail.c, main/file.c: Quote arguments, just in case
+ there's a space in a pathname. (Diagnosed by pabelanger on
+ #asterisk-dev, fixed by me.)
+
+2010-12-29 19:28 +0000 [r299865-299948] Paul Belanger <pabelanger at digium.com>
+
+ * sounds/Makefile: Only remove /tmp/astdatadir, not
+ /var/lib/asterisk
+
+ * build_tools/make_sample_voicemail, sounds/Makefile, Makefile:
+ Properly quote varibles for MAC OS X
+
+ * apps/app_chanspy.c, /: Merged revisions 299864 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r299864 | pabelanger | 2010-12-28 13:51:13 -0500 (Tue, 28 Dec
+ 2010) | 2 lines Documentation typo ........
+
+2010-12-27 21:23 +0000 [r299752-299820] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * sounds/Makefile: More space-in-pathname issues.
+
+ * sounds/Makefile, Makefile, Makefile.moddir_rules: Mac OS X
+ spaces-in-pathnames fix.
+
+ * configure: Regen configure
+
+ * configure.ac: Properly quote path on Darwin.
+
+2010-12-25 16:12 +0000 [r299711] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooh245.c: Change
+ order of sending TCS and MSD packets Change order of sending
+ Terminal Capability Set and MasterSlave Determination packets,
+ MSD send when TCS exchange procedure is done (we send tcs ack to
+ remote and we have remote tcs ack already or we receive tcs ack
+ from remote and we have send our tcs ack to remote already). Some
+ endpoints can work in this sequence only, i suggest they can't
+ work with both (tcs and msd) exchange procedures simultaneously.
+ Also changed StartH245 facility message sending. It send on
+ incoming calls only due to some endpoints can't proccess properly
+ this facility messages on their incoming calls. (issue #18433)
+ Reported by: MrHanMan Patches: tcs-msd-h245-3.patch uploaded by
+ may213 (license 454) Tested by: MrHanMan, may213
+
+2010-12-25 10:07 +0000 [r299583-299626] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * channels/chan_local.c, /: Merged revisions 299625 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r299625 | tilghman | 2010-12-25 04:05:00 -0600
+ (Sat, 25 Dec 2010) | 12 lines Merged revisions 299624 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010)
+ | 5 lines Move check for extension existence below variable
+ inheritance, due to the possible use of an eswitch. (closes issue
+ #16228) Reported by: jlaguilar ........ ................
+
+ * addons/res_config_mysql.c: Reset 'first' variable after usage.
+ (closes issue #18525) Reported by: makoto Patches:
+ res-config-mysql-update2.patch uploaded by makoto (license 38)
+
+2010-12-23 02:53 +0000 [r299531] Moises Silva <moises.silva at gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 299530 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r299530 | moy | 2010-12-22 21:28:37 -0500 (Wed, 22 Dec
+ 2010) | 7 lines Enqueue AST_CONTROL_PROGRESS after
+ AST_CONTROL_RINGING when MFC-R2 calls are accepted (closes issue
+ #18438) Reported by: mariner7 Tested by: moy ........
+
+2010-12-22 20:05 +0000 [r299449] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c,
+ pbx/ael/ael-test/ref.ael-vtest25,
+ pbx/ael/ael-test/ref.ael-vtest17, /,
+ pbx/ael/ael-test/ref.ael-test3: Merged revisions 299448 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r299448 | tilghman | 2010-12-22 14:03:30 -0600 (Wed, 22 Dec 2010)
+ | 8 lines Resolve warnings by disambiguating the "s" extension as
+ used by chan_dahdi from the "s" extension as used by the AEL
+ macros. (closes issue #18480) Reported by: nivek Patches:
+ 20101215__issue18480__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: nivek ........
+
+2010-12-22 02:10 +0000 [r299405] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c: Chan_dahdi sends an empty COLP on the bridged
+ channel. Chan_dahdi always inserts a connected party IE when you
+ call from one dahdi channel to another dahdi channel, even if no
+ such information was received on the 2nd channel. This clears the
+ display of many phones. * Removed leftover artifact from before
+ the valid flag was added. * Updated all of the channel's caller
+ id information with the new connected line information instead of
+ just the string parts. (closes issue #18508) Reported by: wimpy
+ Patches: issue18508_trunk.patch uploaded by rmudgett (license
+ 664) Tested by: wimpy, rmudgett
+
+2010-12-21 15:25 +0000 [r299353] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 299242 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r299242 | mnicholson | 2010-12-20 15:25:35 -0600
+ (Mon, 20 Dec 2010) | 23 lines Merged revisions
+ 299194,299198,299220 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec
+ 2010) | 6 lines Respond as soon as possible with a 202 Accepted
+ to refer requests. This change also plugs a few memory leaks that
+ can occur when parking sip calls. ABE-2656 ........ r299198 |
+ mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2
+ lines Remove changes to via processing that were not supposed to
+ go into the last commit. ........ r299220 | mnicholson |
+ 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use
+ ast_free() instead of free() ABE-2656 ........ ................
+
+2010-12-21 00:44 +0000 [r299312] Paul Belanger <pabelanger at digium.com>
+
+ * configs/cel.conf.sample: Correct typo with USER_DEFINED event.
+ (closes issue #18461) Reported by: joscas Patches:
+ cel.conf.sample.diff uploaded by lathama (license 1028) Tested
+ by: lathama, joscas
+
+2010-12-20 21:38 +0000 [r299248] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Fix a couple of CCSS issues. * Make sure to
+ allocate a cc_params structure when creating autopeers. * Use
+ sip_uri_cmp when retrieving SIP CC agents and monitors in case
+ parameters appear in the URI. (closes issue #18504) Reported by:
+ kkm (closes issue #18338) Reported by: GeorgeKonopacki Patches:
+ 18338.diff uploaded by mmichelson (license 60) Tested by:
+ GeorgeKonopacki
+
+2010-12-20 18:17 +0000 [r299131-299138] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * sample.call, /: Merged revisions 299136 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r299136 | tilghman | 2010-12-20 12:16:37 -0600 (Mon, 20 Dec 2010)
+ | 2 lines Documentation fix ........
+
+ * cdr/cdr_pgsql.c, /: Merged revisions 299130 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r299130 | tilghman | 2010-12-20 11:41:24 -0600 (Mon, 20 Dec 2010)
+ | 11 lines If a call was not answered, then the billsec was
+ calculated unusually large. Also, due to a copy and paste error,
+ a request for the answer field would have given the start value,
+ instead. (closes issue #18460) Reported by: joscas Patches:
+ 20101215__issue18460.diff.txt uploaded by tilghman (license 14)
+ Tested by: joscas ........
+
+2010-12-20 16:18 +0000 [r299088] Leif Madsen <lmadsen at digium.com>
+
+ * /, main/features.c: Merged revisions 299087 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r299087 | lmadsen | 2010-12-20 10:18:03 -0600 (Mon, 20 Dec 2010)
+ | 5 lines Note that Park() timeout is milliseconds. (closes issue
+ #15758) Reported by: mmurdock Tested by: mmurdock, seanbright
+ ........
+
+2010-12-20 09:14 +0000 [r299004] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * main/aoc.c, channels/sig_pri.h, channels/chan_sip.c: Typos:
+ recieved => received
+
+2010-12-18 00:09 +0000 [r298818-298963] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * /, main/say.c: Merged revisions 298962 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r298962 | tilghman | 2010-12-17 18:08:57 -0600 (Fri, 17 Dec 2010)
+ | 2 lines Remove backtrace used for testing merge process
+ ........
+
+ * main/utils.c, main/astobj2.c, utils/conf2ael.c,
+ include/asterisk/logger.h, configure,
+ build_tools/menuselect-deps.in, main/logger.c, utils/ael_main.c,
+ utils/hashtest2.c, makeopts.in, utils/check_expr.c,
+ utils/refcounter.c, include/asterisk/utils.h,
+ build_tools/cflags-devmode.xml, /,
+ include/asterisk/autoconfig.h.in, main/Makefile, main/say.c,
+ configure.ac, utils/hashtest.c: Merged revisions 298957 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298957 | tilghman | 2010-12-17 17:30:55 -0600
+ (Fri, 17 Dec 2010) | 13 lines Merged revisions 298905 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010)
+ | 6 lines Let Asterisk find better backtrace information with
+ libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will
+ use libbfd to search for better symbol information within both
+ the Asterisk binary, as well as loaded modules, to assist when
+ using inline backtraces to track down problems. ........
+ ................
+
+ * contrib/init.d/rc.debian.asterisk: -v implies -f, so override
+ with -F. (closes issue #18446) Reported by: lathama Patches:
+ rc.debian.asterisk.diff uploaded by lathama (license 1028) Tested
+ by: lathama
+
+ * /, configure, configure.ac: Merged revisions 298817 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r298817 | tilghman | 2010-12-17 15:03:06 -0600 (Fri, 17
+ Dec 2010) | 8 lines Also include PTHREAD_LIBS and PTHREAD_CFLAGS
+ for SQLite 3, as it's needed on some platforms. (closes issue
+ #18493) Reported by: pprindeville Patches:
+ asterisk-1.8-sqlite3.patch uploaded by pprindeville (license 347)
+ Tested by: pprindeville ........
+
+2010-12-17 17:26 +0000 [r298773] Brad Watkins <Marquis42 at gmail.com>
+
+ * configs/sip.conf.sample, channels/chan_sip.c: Fix parsing of mwi
+ => lines in sip.conf Reworking parsing of mwi => lines to resolve
+ a segfault. Also add a set of unit tests for the function that
+ does the parsing. (closes issue #18350) Reported by: gbour Tested
+ by: Marquis, gbour Review:
+ https://reviewboard.asterisk.org/r/1053/
+
+2010-12-16 23:31 +0000 [r298598-298685] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 298684 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298684 | jpeeler | 2010-12-16 17:30:59 -0600
+ (Thu, 16 Dec 2010) | 9 lines Merged revisions 298683 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16
+ Dec 2010) | 2 lines After recording only silence for a voicemail
+ prepending, restore backup files. ........ ................
+
+ * apps/app_queue.c, /: Merged revisions 298597 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298597 | jpeeler | 2010-12-16 14:49:33 -0600
+ (Thu, 16 Dec 2010) | 14 lines Merged revisions 298596 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010)
+ | 7 lines Fix improper hangup when doing an attended transfer to
+ queue. Had to indicate ringing in wait_for_answer so the attended
+ transfer code would not try and hang up the local channel it
+ created, which would kill the call. ABE-2624 ........
+ ................
+
+2010-12-16 09:28 +0000 [r298394-298539] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * channels/chan_sip.c: Ensure the ipaddr field in realtime is large
+ enough to handle IPv6 addresses. (closes issue #18464) Reported
+ by: IgorG Patches: realtime_ipv6store.diff uploaded by IgorG
+ (license 20) (plus a few additional lines by tilghman)
+
+ * res/res_config_odbc.c, /: Merged revisions 298481 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298481 | tilghman | 2010-12-16 03:04:38 -0600
+ (Thu, 16 Dec 2010) | 21 lines Merged revisions 298480 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298480 | tilghman | 2010-12-16 03:03:40 -0600 (Thu, 16 Dec 2010)
+ | 14 lines Only increment the pointer once per loop, otherwise we
+ corrupt the value. (closes issue #18251) Reported by: bcnit
+ Patches: 20101110__issue18251.diff.txt uploaded by tilghman
+ (license 14) Tested by: trev, jthurman, elguero (closes issue
+ #18279) Reported by: zerohalo Patches:
+ 20101109__issue18279.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo ........ ................
+
+ * /, funcs/func_dialgroup.c: Merged revisions 298477 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r298477 | tilghman | 2010-12-16 02:54:23 -0600 (Thu, 16
+ Dec 2010) | 8 lines Eliminate duplicates from container. (closes
+ issue #18091) Reported by: bunny Patches:
+ 20101006__issue18091.diff.txt uploaded by tilghman (license 14)
+ Tested by: bunny ........
+
+ * /, cdr/cdr_sqlite.c: Merged revisions 298393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298393 | tilghman | 2010-12-15 18:29:10 -0600
+ (Wed, 15 Dec 2010) | 15 lines Merged revisions 298392 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298392 | tilghman | 2010-12-15 18:28:04 -0600 (Wed, 15 Dec 2010)
+ | 8 lines Unregister before shutting down the connection, to
+ avoid a race. (closes issue #18481) Reported by: pabelanger
+ Patches: 20101215__issue18481.diff.txt uploaded by tilghman
+ (license 14) Tested by: pabelanger ........ ................
+
+2010-12-13 17:11 +0000 [r298195] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c, channels/chan_dahdi.c, /: Merged revisions
+ 298194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298194 | rmudgett | 2010-12-13 11:04:41 -0600
+ (Mon, 13 Dec 2010) | 26 lines Merged revisions 298193 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010)
+ | 19 lines Outgoing PRI/BRI calls cannot do DTMF triggered
+ transfers. Outgoing PRI/BRI calls cannot do DTMF triggered
+ transfers if a PROCEEDING message is not received. The debug
+ output shows that the DTMF begin event is seen, but the DTMF end
+ event is missing. When the DTMF begin happens, the call is muted
+ so we now have one way audio (until a DTMF end event is somehow
+ seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER
[... 27023 lines stripped ...]
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