[asterisk-commits] lmadsen: tag 1.6.2.17-rc1 r301943 - /tags/1.6.2.17-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jan 14 14:59:56 CST 2011
Author: lmadsen
Date: Fri Jan 14 14:59:51 2011
New Revision: 301943
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=301943
Log:
Importing files for 1.6.2.17-rc1 release.
Added:
tags/1.6.2.17-rc1/.lastclean (with props)
tags/1.6.2.17-rc1/.version (with props)
tags/1.6.2.17-rc1/ChangeLog (with props)
Added: tags/1.6.2.17-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.17-rc1/.lastclean?view=auto&rev=301943
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--- tags/1.6.2.17-rc1/ChangeLog (added)
+++ tags/1.6.2.17-rc1/ChangeLog Fri Jan 14 14:59:51 2011
@@ -1,0 +1,29078 @@
+2011-01-14 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.17-rc1 Released.
+
+2011-01-14 20:03 +0000 [r301842-301848] lathama <lathama at localhost>:
+
+ * funcs/func_base64.c, funcs/func_aes.c: Add relationships to
+ function documentation. Fix amatuer type mistake
+
+ * funcs/func_base64.c, funcs/func_aes.c: Add relationships to
+ function documentation.
+
+2011-01-13 17:01 +0000 [r301730] Leif Madsen <lmadsen at digium.com>
+
+ * configs/phoneprov.conf.sample: 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 21:05 +0000 [r301682] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: 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 18:50 +0000 [r301594] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/manager.c, /: 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 18:11 +0000 [r301503] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, /: 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 ........
+
+2011-01-11 19:14 +0000 [r301310] Paul Belanger <pabelanger at digium.com>
+
+ * configs/extensions.conf.sample: Fix a logic issue when passing
+ context ARG
+
+2011-01-11 18:42 +0000 [r301307] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, main/utils.c: 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 ........
+
+2011-01-09 21:38 +0000 [r301176-301220] Paul Belanger <pabelanger at digium.com>
+
+ * autoconf/ast_ext_lib.m4, configure, configure.ac: 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: 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)
+
+2011-01-07 20:52 +0000 [r301089] Jason Parker <jparker at digium.com>
+
+ * apps/app_meetme.c: 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 19:57 +0000 [r300951-301046] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: 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 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 ........
+
+2011-01-05 18:54 +0000 [r300622] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * res/res_odbc.c, /: 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 16:28 +0000 [r300574] Paul Belanger <pabelanger at digium.com>
+
+ * cdr/cdr_sqlite.c: Change deprecated message to LOG_WARNING Also
+ removed latter part of message Discussed on #asterisk-dev
+
+2011-01-04 21:52 +0000 [r300431-300520] Leif Madsen <lmadsen at digium.com>
+
+ * channels/chan_iax2.c, main/xmldoc.c, channels/chan_sip.c,
+ channels/chan_agent.c: 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: 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 20:59 +0000 [r300429] Russell Bryant <russell at digium.com>
+
+ * contrib/scripts/autosupport, /, contrib/scripts/autosupport.8:
+ 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 17:37 +0000 [r300298] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: 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/
+ ........
+
+2011-01-03 23:02 +0000 [r300165] Richard Mudgett <rmudgett at digium.com>
+
+ * main/features.c: Use correct variable for atxfercallbackretries
+ config option. * Misc formatting changes.
+
+2010-12-28 18:51 +0000 [r299864] Paul Belanger <pabelanger at digium.com>
+
+ * apps/app_chanspy.c: Documentation typo
+
+2010-12-25 10:05 +0000 [r299625] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * channels/chan_local.c, /: 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 ........
+
+2010-12-23 03:02 +0000 [r299530-299533] Moises Silva <moises.silva at gmail.com>
+
+ * channels/chan_dahdi.c: do not use progress which is for PRI and
+ SS7, add mfcr2_progress member
+
+ * channels/chan_dahdi.c: 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 20:03 +0000 [r299448] 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:
+ 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
+
+2010-12-20 21:25 +0000 [r299242] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_sip.c: 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 ........
+
+2010-12-20 18:16 +0000 [r299130-299136] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * sample.call: Documentation fix
+
+ * cdr/cdr_pgsql.c: 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 [r299087] Leif Madsen <lmadsen at digium.com>
+
+ * main/features.c: Note that Park() timeout is milliseconds.
+ (closes issue #15758) Reported by: mmurdock Tested by: mmurdock,
+ seanbright
+
+2010-12-20 09:13 +0000 [r299003] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * channels/chan_sip.c: Typos: recieved => received
+
+2010-12-18 00:08 +0000 [r298817-298962] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * main/say.c: Remove backtrace used for testing merge process
+
+ * 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, /, main/Makefile,
+ include/asterisk/autoconfig.h.in, main/say.c, configure.ac,
+ utils/hashtest.c, main/utils.c: 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. ........
+
+ * configure, configure.ac: 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
+
+2010-12-16 23:30 +0000 [r298597-298684] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: 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 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 09:04 +0000 [r298393-298481] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * res/res_config_odbc.c, /: 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: 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 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 ........
+
+2010-12-15 21:31 +0000 [r298346] Sean Bright <sean at malleable.com>
+
+ * main/astobj2.c, /: Merged revisions 298345 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298345 | seanbright | 2010-12-15 16:28:29 -0500 (Wed, 15 Dec
+ 2010) | 6 lines Fix reference and container leaks when running
+ 'astobj2 test.' We need to make sure that ao2_iterator_destroy is
+ called once for each time that ao2_iterator_init is called. Also
+ make sure to unref a newly allocated object that we've linked
+ into a container. ........
+
+2010-12-13 17:04 +0000 [r298194] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: 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 event is received. * Made absorb the DTMF begin
+ and DTMF end events if we are overlap dialing and have not seen a
+ PROCEEDING message. * Added a debug message when absorbing a DTMF
+ event. JIRA SWP-2690 JIRA ABE-2697 ........
+
+2011-01-12 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.16 Released.
+
+2011-01-12 Leif Madsen <lmadsen at digium.com>
+
+ * Merge in changes for configure script to resolve issue for
+ Debian package builders.
+
+ ------------------------------------------------------------------------
+ r301220 | pabelanger | 2011-01-09 15:38:25 -0600 (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 0018589)
+ Reported by: pabelanger
+ Patches:
+ issue18589.patch uploaded by pabelanger (license 224)
+ Tested by: pabelanger
+
+ Review: https://reviewboard.asterisk.org/r/1074/ [^]
+
+ ------------------------------------------------------------------------
+
+2010-12-13 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.16-rc1 Released.
+
+2010-12-10 16:24 +0000 [r298050] Tilghman Lesher <tlesher at digium.com>
+
+ * main/netsock.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Portability issue on OpenSolaris. Also detect the
+ required structure element, because OpenSolaris defines
+ SIOCGIFHWADDR, but without support for IP sockets. (closes issue
+ #18442) Reported by: ranjtech Patches:
+ 20101209__issue18442.diff.txt uploaded by tilghman (license 14)
+ Tested by: ranjtech
+
+2010-12-09 22:10 +0000 [r297960] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297959 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010)
+ | 14 lines Ignore spurious REGISTER requests If a REGISTER
+ request with a Call-ID matching an existing transaction is
+ received it was possible that the REGISTER request would
+ overwrite the initreq of the private structure. This info is used
+ to generate messages for other responses in the transaction. This
+ patch ignores REGISTER requests that match non-REGISTER
+ transactions. (closes issue #18051) Reported by: eeman Tested by:
+ twilson Review: https://reviewboard.asterisk.org/r/1050/ ........
+
+2010-12-08 18:04 +0000 [r297908] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/extensions.conf.sample: Use inheritance to get correct
+ results for SIPFROMDOMAIN. (from an internal Digium discussion)
+
+2010-12-07 22:58 +0000 [r297824] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, /: Merged revisions 297823 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010)
+ | 12 lines Revert code that changed SSRC for DTMF. Some previous
+ behavior was attempted to be restored, but mistakingly I did not
+ realize that the previous behavior was incorrect. This fixes DTMF
+ not being detected since DTMF shouldn't cause the SSRC to change.
+ (related to issue #17404) (closes issue #18189) (closes issue
+ #18352) Reported by: marcbou Tested by: cmbaker82 ........
+
+2010-12-07 22:40 +0000 [r297713-297819] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/init.d/org.asterisk.muted.plist (added), Makefile,
+ utils/muted.c, /: Merged revisions 297818 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010)
+ | 4 lines Use non-deprecated APIs for CoreAudio Review:
+ https://reviewboard.asterisk.org/r/1040/ ........
+
+ * apps/app_followme.c, /: Merged revisions 297689 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010)
+ | 8 lines Don't create a Local channel if the target extension
+ does not exist. (closes issue #18126) Reported by: junky Patches:
+ followme.diff uploaded by junky (license 177) (partially
+ restructured by me to avoid a possible memory leak) ........
+
+2010-12-06 22:03 +0000 [r297605] Jeff Peeler <jpeeler at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297603 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010)
+ | 12 lines Improve handling of REGISTER requests with multiple
+ contact headers. The changes here attempt to more strictly follow
+ RFC 3261 section 10.3. Basically the following will now cause a
+ 400 Bad Response to be returned, if: - multiple Contact headers
+ are present with one set to expire all bindings ("*") - wildcard
+ parameter is specified for Contact without Expires header or
+ Expires header is not set to zero. ABE-2442 ABE-2443 ........
+
+2010-12-03 17:40 +0000 [r297534] Sean Bright <sean at malleable.com>
+
+ * channels/chan_console.c: The CLI command should not contain
+ <placeholder>s, these are for descriptions.
+
+2010-12-02 20:06 +0000 [r297405] Paul Belanger <pabelanger at digium.com>
+
+ * Makefile, /: Merged revisions 297404 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec
+ 2010) | 7 lines Resolve compile error under FreeBSD We now set
+ _ASTCFLAGS+=-march=i686 for i386 processors, still allowing
+ ASTCFLAGS to override the setting. Review:
+ https://reviewboard.asterisk.org/r/1043/ ........
+
+2010-12-02 18:07 +0000 [r297311] Terry Wilson <twilson at digium.com>
+
+ * /, main/abstract_jb.c: Merged revisions 297310 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010)
+ | 12 lines Initialize offset for adaptive jitter buffer When the
+ adaptive jitter buffer is enabled in sip.conf, the first frame
+ placed in the jitter buffer fails with something like:
+ jb_warning_output: Resyncing the jb. last_delay 0, this delay
+ -215886466, threshold 1000, new offset 215886466 This happens
+ because the offset is not initialized before calling jb_put().
+ This patch modifies jb_put_first_adaptive() to set the offset to
+ the frame's timestamp. Review:
+ https://reviewboard.asterisk.org/r/1041/ ........
+
+2010-12-02 13:16 +0000 [r297229] Russell Bryant <russell at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 297228 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010)
+ | 6 lines Add "DAHDI" to a couple of app_meetme error messages.
+ This is in response to some questions on IRC. To the user, there
+ was nothing that made it obvious that this error had anything to
+ do with DAHDI not being loaded. ........
+
+2010-12-02 08:55 +0000 [r297186] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 297185 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297185 | oej | 2010-12-02 09:37:17 +0100 (Tor, 02 Dec 2010) | 5
+ lines If we get a NOTIFY from a non-existing subscription we
+ should answer with 481, not bad event. If we answer 481 the
+ subscription that we don't want will be cancelled. ........
+
+2010-12-01 17:52 +0000 [r297073] Jeff Peeler <jpeeler at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297072 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010)
+ | 23 lines Fix not stopping MOH when transfered local channel
+ queue member is answered. The problem here is only present when
+ local channels are used with the MOH passthru option as well as
+ no optimization (/nm). I will describe the slightly bizarre
+ scenario that was used to test, where phones B and C are queue
+ members: Phone A dials into a queue with two members using local
+ channels and the above options. Phone B answers. Phone A blind
+ transfers phone B into the same queue. Phone A hangs up. Phone C
+ answers, but phone B didn't stop playing MOH. In this scenario,
+ the unhold frame that should have gotten to phone B never arrived
+ due to the masquerade from the blind transfer. This is usually
+ fine since app_queue manages the starting and stopping of MOH.
+ However, with the passthrough option enabled when app_queue
+ attempts to stop MOH it tries to do so on the local channel
+ rather than the real channel. The easiest solution was to just
+ make sure to send an unhold frame during the transfer since it
+ wouldn't make sense to have MOH playing after a transfer anyway.
+ This only modifies SIP transfers, but the other transfers did not
+ seem to be a problem. If DTMF based transfers were a problem it
+ might be okay to add ast_moh_stop to finishup, but I didn't want
+ to have to add that unless required. ABE-2624 ........
+
+2010-12-01 17:01 +0000 [r296950-296991] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/frame.h, /: Merged revisions 296990 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01
+ Dec 2010) | 5 lines Clarify documentation on how we store codec
+ preference lists. (closes issue #18397) Reported by: birgita
+ ........
+
+ * channels/chan_iax2.c: Missed initializations caused startup
+ errors on Mac OS X (and possibly others, too).
+
+2010-12-01 00:24 +0000 [r296869] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 296868 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30
+ Nov 2010) | 4 lines Properly restore backup information file when
+ hanging up during message prepending. ABE-2654 ........
+
+2010-11-29 22:54 +0000 [r296671] Paul Belanger <pabelanger at digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 296670 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon,
+ 29 Nov 2010) | 5 lines Make sure nothing else is needed before
+ destroying the scheduler. (closes issue #18398) Reported by:
+ pabelanger ........
+
+2010-11-29 07:27 +0000 [r296533] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: I love standards. There are so many to choose from.
+ Except when there isn't one. Linux and *BSD disagree on the
+ elements within the ucred structure. Detect which one is in use
+ on the system. (closes issue #18384) Reported by: bjm Patches:
+ cred-diffs uploaded by bjm (license 473)
+ 20101127__issue18384__1.6.2.diff.txt uploaded by tilghman
+ (license 14) 20101127__issue18384__1.8.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman, bjm
+
+2010-11-27 10:39 +0000 [r296466] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_meetme.c: 18 characters is too short for most date/times
+ (20 is the usual, but we add more in case of greater precision).
+ (closes issue #18369) Reported by: tnakonz
+
+2010-11-26 12:23 +0000 [r296351] Olle Johansson <oej at edvina.net>
+
+ * /, main/say.c: Merged revisions 296309 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11
+ lines Fix bugs in saying numbers using the Swedish language
+ syntax (closes issue #18355) Reported by: oej Patch by: oej Much
+ help from Peter Lindahl. Testing by the ClearIT team during a
+ coffee break. Review: https://reviewboard.asterisk.org/r/1033/
+ ........
+
+2010-11-24 23:28 +0000 [r296221] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, /: Merged revisions 296213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010)
+ | 6 lines Make Asterisk less crashy. Since we might not put a new
+ translation path on the channel, go ahead and set it to NULL
+ right after destroying the old one to ensure we don't try to free
+ an invalid translation path later on. ........
+
+2010-11-24 22:42 +0000 [r296166] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 296165 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24
+ Nov 2010) | 43 lines Oneway audio to SIP phone from FXS port
+ after FXS port gets a CallWaiting pip. The FXS connected phone
+ has to have CW/CID support to fail, as it will send back a DTMF
+ 'A' or 'D' when it's ready to receive CallerID. A normal phone
+ with no CID never fails. Also the SIP phone does not hear MOH
+ when the CW call is answered. The DTMF end frame is suppressed
+ when the phone acknowledges the CW signal for CID. The problem is
+ the DTMF begin frame needs to be suppressed as well. The DTMF
+ begin frame is causing SIP to start sending the DTMF RTP frames.
+ Since the DTMF end frame is suppressed, SIP will not stop sending
+ those DTMF RTP packets. * Suppress the DTMF begin and end frames
+ when the channel driver is looking for DTMF digits. * Fixed a
+ couple issues caused by not cleaning up the CID spill if you
+ answer the CW call while it is sending the CID spill. * Fixed not
+ sending CW/CID spill to the phone when the call is natively
+ bridged. (Fixed by not using native bridge if CW/CID is
+ possible.) * Suppress received audio when sending CW/CID spills.
+ The other parties involved do not need to hear the CW/CID spills
+ and may be confused if the CW call is for them. (closes issue
+ #18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch
+ uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+ NOTE: * v1.4 does not have the main problem fixed by suppressing
+ the DTMF start frames. The other three items fixed are relevant.
+ * If you really must restore native bridging between analog
+ ports, you need to disable CW/CID either by configuring
+ chan_dahdi.conf callwaitingcallerid=no or dialing *70 before
+ dialing the number to temporarily disable CW. ........
+
+2010-11-24 20:23 +0000 [r296001-296083] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, /: Merged revisions 296082 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010)
+ | 12 lines Fix false reporting of an error by set_format(). In
+ the case that the native format was able to be changed to match
+ the new requested format, the code proceeded to attempt to build
+ a translation path, anyway. The result would be NULL, since no
+ translation path is necessary and resulted in this function
+ thinking an error has occurred. This case is now specifically
+ caught and no attempt to build a translation path is attempted.
+ Thanks to our automated tests and bamboo.asterisk.org for
+ catching this problem and making a whole lot of noise when things
+ started failing. :-) ........
+
+ * apps/app_dial.c, main/channel.c, /: Merged revisions 296000 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010)
+ | 38 lines Handle failures building translation paths more
+ effectively. The problem scenario occurred on a heavily loaded
+ system that was using the codec_dahdi module and exceeded the
+ hardware transcoding capacity. The failure mode at that point was
+ not good. The report came in to us as an Asterisk lock-up. The
+ "core show locks" shows a ton of threads locked up (but no
+ obvious deadlock). Upon deeper investigation, when the system is
+ in this state, the CPU was maxed out. The CPU was being consumed
+ by the Asterisk logger spewing messages on every audio frame for
+ calls set up after transcoder capacity was reached. The purpose
+ of this patch is to make Asterisk handle failures to create a
+ translation path in a more graceful manner. If we can't
+ translate, then the call just needs to be dropped, as it's not
+ going to work. These are the changes: 1) In set_format() of
+ channel.c (which is called by set_read_format() and
+ set_write_format()), it was ignoring if
+ ast_translator_build_path() failed and returned NULL. It now pays
+ attention to that case and returns a result reflecting failure.
+ With this change in place, the bridging code will immediately
+ detect a failure and end the bridge instead of proceeding to try
+ to bridge frames that can't be translated and making channel
+ drivers freak out by sending them frames in a format they weren't
+ expecting. 2) In ast_indicate_data() of channel.c, failure of
+ ast_playtones_start() was ignored. It is now reflected in the
+ return value of the function. This didn't turn out to have any
+ affect on the bug, but seemed like a good change to leave in. 3)
+ In app_dial(), when only sending a call to a single endpoint, it
+ will attempt to do some bridging of its own of early audio. It
+ uses make_compatible() when it's going to do this. However, it
+ ignored failure from make compatible. So, even with the fix from
+ #1, if there was early audio going through app_dial, there would
+ still be a period of invalid frames passing through. After
+ detecting failure here, Dial() exits. ABE-2658 ........
+
+2010-11-23 09:36 +0000 [r295907] Olle Johansson <oej at edvina.net>
+
+ * /, main/say.c: Merged revisions 295906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8
+ lines Fix support of saynumber(1,n) in the Swedish language
+ (closes issue #18353) Reported by: oej Review:
+ https://reviewboard.asterisk.org/r/1031/ ........
+
+2010-11-22 20:02 +0000 [r295868] Sean Bright <sean at malleable.com>
+
+ * configs/chan_dahdi.conf.sample: Change some documentation to
+ suggest dahdi_monitor instead of ztmonitor.
+
+2010-11-22 19:28 +0000 [r295843] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/frame.h, main/channel.c, main/pbx.c, /,
+ apps/app_macro.c, include/asterisk/channel.h: Merged revisions
+ 295790 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010)
+ | 46 lines The channel redirect function (CLI or AMI) hangs up
+ the call instead of redirecting the call. To recreate the
+ problem: 1) Party A calls Party B 2) Invoke CLI "channel
+ redirect" command to redirect channel call leg associated with A.
+ 3) All associated channels are hung up. Note that if the CLI
+ command were done on the channel call leg associated with B it
+ works. This regression was a result of the fix for issue #16946
+ (https://reviewboard.asterisk.org/r/740/). The regression affects
+ all features that use an async goto to execute the dialplan
+ because of an external event: Channel redirect, AMI redirect, SIP
+ REFER, and FAX detection. The struct ast_channel._softhangup code
+ is a mess. The variable is used for several purposes that do not
+ necessarily result in the call being hung up. I have added
+ doxygen comments to describe how the various _softhangup bits are
+ used. I have corrected all the places where the variable was
+ tested in a non-bit oriented manner. The primary fix is the new
+ AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so
+ the soft hangup requests that do not normally result in a hangup
+ do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171)
+ Reported by: SantaFox (closes issue #18185) Reported by:
+ kwemheuer (closes issue #18211) Reported by: zahir_koradia
+ (closes issue #18230) Reported by: vmarrone (closes issue #18299)
+ Reported by: mbrevda (closes issue #18322) Reported by: nerbos
+ Review: https://reviewboard.asterisk.org/r/1013/ ........
+
+2010-11-20 00:45 +0000 [r295710] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/event.h, main/event.c: Fix cache of device state
+ changes for multiple servers. This patch addresses a regression
+ where device states across multiple servers were not being
+ processing completely correctly. The code works to determine the
+ overall state by looking at the last known state of a device on
+ each server. However, there was a regression due to some invasive
+ rewrites of how the cache works that led to the cache only
+ storing the last device state change for a device, regardless of
+ which server it was on. The code is set up to cache device state
+ change events by ensuring that each event in the cache has a
+ unique device name + entity ID (server ID). The code that was
+ responsible for comparing raw information elements (which EID is)
+ always returned a match due to a memcmp() with a length of 0.
+ There isn't much code to fix the actual bug. This patch also
+ introduces a new CLI command that was very useful for debugging
+ this problem. The command allows you to dump the contents of the
+ event cache. (closes issue #18284) Reported by: klaus3000
+ Patches: issue18284.rev1.txt uploaded by russell (license 2)
+ Tested by: russell, klaus3000 (closes issue #18280) Reported by:
+ klaus3000 Review: https://reviewboard.asterisk.org/r/1012/
+
+2010-11-19 21:55 +0000 [r295672] Terry Wilson <twilson at digium.com>
+
[... 28356 lines stripped ...]
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