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authorlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2011-01-14 20:59:51 +0000
committerlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2011-01-14 20:59:51 +0000
commite50226af4738c652a329efed1a46f68e4b8b8e70 (patch)
tree606c91fd97d61d63a871ae17768a98bfb4a52337
parentd1b093cb21f1e5955a42aa4d9f1a6872ad66e518 (diff)
Importing files for 1.6.2.17-rc1 release.
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.17-rc1@301943 f38db490-d61c-443f-a65b-d21fe96a405b
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-rw-r--r--.version1
-rw-r--r--ChangeLog29078
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+1.6.2.17-rc1
diff --git a/ChangeLog b/ChangeLog
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@@ -0,0 +1,29078 @@
+2011-01-14 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.17-rc1 Released.
+
+2011-01-14 20:03 +0000 [r301842-301848] lathama <lathama@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@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@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@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@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@digium.com>
+
+ * configs/extensions.conf.sample: Fix a logic issue when passing
+ context ARG
+
+2011-01-11 18:42 +0000 [r301307] Matthew Nicholson <mnicholson@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@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@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@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@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@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@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@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@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@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@digium.com>
+
+ * apps/app_chanspy.c: Documentation typo
+
+2010-12-25 10:05 +0000 [r299625] Tilghman Lesher <tilghman@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@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@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@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@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@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@xorcom.com>
+
+ * channels/chan_sip.c: Typos: recieved => received
+
+2010-12-18 00:08 +0000 [r298817-298962] Tilghman Lesher <tilghman@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@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@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@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@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@digium.com>
+
+ * Asterisk 1.6.2.16 Released.
+
+2011-01-12 Leif Madsen <lmadsen@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@digium.com>
+
+ * Asterisk 1.6.2.16-rc1 Released.
+
+2010-12-10 16:24 +0000 [r298050] Tilghman Lesher <tlesher@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 295628 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010)
+ | 8 lines Discard responses with more than one Via This is not a
+ perfect solution as headers that are joined via commas are not
+ detected. This is a parsing issue that to fix "correctly" would
+ necessitate a new SIP parser. Review:
+ https://reviewboard.asterisk.org/r/1019/ ........
+
+2010-11-18 17:51 +0000 [r295440] Paul Belanger <pabelanger@digium.com>
+
+ * res/res_jabber.c, include/asterisk/jabber.h: Fix compiler
+ warnings when using openssl-dev 1.0.0+ Review:
+ https://reviewboard.asterisk.org/r/1016/
+
+2010-11-16 22:57 +0000 [r295281] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Merged revisions 295280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010)
+ | 1 line Dead code elimination in channel.c:ast_channel_bridge()
+ variable who. ........
+
+2010-12-02 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.15 Released.
+
+2010-11-15 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.15-rc1
+
+2010-11-15 18:24 +0000 [r294988-295062] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_expr.c (added), /: Merged revisions 295026 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010)
+ | 2 lines Create test verifying results of expression parser
+ ........
+
+ * funcs/func_curl.c: It is possible to crash Asterisk by feeding
+ the curl engine invalid data. (closes issue #18161) Reported by:
+ wdoekes Patches: 20101029__issue18161.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman
+
+2010-11-12 21:14 +0000 [r294904-294910] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c: Return correct error code if lock path
+ fails. The recent changes to open_mailbox actually caused it to
+ be fixed, but let's be consistent. Reported by alecdavis in
+ asterisk-dev.
+
+ * apps/app_voicemail.c, /: Merged revisions 294903 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12
+ Nov 2010) | 16 lines Fix regression causing abort in voicemail
+ after opening a mailbox with no mesgs. In order to be more safe,
+ some error handling code was changed to respect more error
+ conditions including the potential memory allocation failure for
+ deleted and heard message tracking introduced in 293004. However,
+ last_message_index returns -1 for zero messages (perhaps as
+ expected) and was triggering the stricter error checking. Because
+ last_message_index is only called directly in one place, just
+ return 0 from open_mailbox (for file based storage) when no
+ messages are detected unless a real error has occurred. (closes
+ issue #18240) Reported by: leobrown Patches:
+ bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
+ Tested by: pabelanger ........
+
+2010-11-12 02:44 +0000 [r294822] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 294821 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11
+ Nov 2010) | 11 lines Asterisk is getting a "No D-channels
+ available!" warning message every 4 seconds. Asterisk is just
+ whining too much with this message: "No D-channels available!
+ Using Primary channel XXX as D-channel anyway!". Filtered the
+ message so it only comes out once if there is no D channel
+ available without an intervening D channel available period.
+ (closes issue #17270) Reported by: jmls ........
+
+2010-11-11 21:57 +0000 [r294639-294733] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 294688 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010)
+ | 18 lines Fix problem with qualify option packets for realtime
+ peers never stopping. The option packets not only never stopped,
+ but if a realtime peer was not in the peer list multiple options
+ dialogs could accumulate over time. This scenario has the
+ potential to progress to the point of saturating a link just from
+ options packets. The fix was to ensure that the poke scheduler
+ checks to see if a peer is in the peer list before continuing to
+ poke. The reason a peer must be in the peer list to be able to
+ properly manage an options dialog is because otherwise the call
+ pointer is lost when the peer is regenerated from the database,
+ which is how existing qualify dialogs are detected. (closes issue
+ #16382) (closes issue #17779) Reported by: lftsy Patches:
+ bug16382-3.patch uploaded by jpeeler (license 325) Tested by:
+ zerohalo ........
+
+ * main/asterisk.c, include/asterisk.h, main/pbx.c, /: Merged
+ revisions 294384 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010)
+ | 47 lines Fix a deadlock in device state change processing.
+ Copied from some notes from the original author (Russell):
+ Deadlock scenario: Thread 1: device state change thread Holds -
+ rdlock on contexts Holds - hints lock Waiting on channels
+ container lock Thread 2: SIP monitor thread Holds the "iflock"
+ Holds a sip_pvt lock Holds channel container lock Waiting for a
+ channel lock Thread 3: A channel thread (chan_local in this case)
+ Holds 2 channel locks acquired within app_dial Holds a 3rd
+ channel lock it got inside of chan_local Holds a local_pvt lock
+ Waiting on a rdlock of the contexts lock A bunch of other threads
+ waiting on a wrlock of the contexts lock To address this
+ deadlock, some locking order rules must be put in place and
+ enforced. Existing relevant rules: 1) channel lock before a pvt
+ lock 2) contexts lock before hints lock 3) channels container
+ before a channel What's missing is some enforcement of the order
+ when you involve more than any two. To fix this problem, I put in
+ some code that ensures that (at least in the code paths involved
+ in this bug) the locks in (3) come before the locks in (2). To
+ change the operation of thread 1 to comply, I converted the
+ storage of hints to an astobj2 container. This allows processing
+ of hints without holding the hints container lock. So, in the
+ code path that led to thread 1's state, it no longer holds either
+ the contexts or hints lock while it attempts to lock the channels
+ container. (closes issue #18165) Reported by: antonio ABE-2583
+ ........
+
+2010-11-10 23:16 +0000 [r294571] Tilghman Lesher <tlesher@digium.com>
+
+ * main/features.c: Actually pay attention to documented settings in
+ features.conf. (closes issue #16757) Reported by: voxter Patches:
+ 20101012__issue16757.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/994/
+
+2010-11-10 12:41 +0000 [r294500] Russell Bryant <russell@digium.com>
+
+ * main/devicestate.c: Improve a debug message to be more readable
+ and consistent. (closes issue #18282) Reported by: klaus3000
+ Patches: ast_devstate2str-patch.txt uploaded by klaus3000
+ (license 65)
+
+2010-11-09 20:27 +0000 [r294429] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Detect GMime properly on systems where
+ gmime flags and libs are configured with pkg-config. (closes
+ issue #16155) Reported by: jcollie Patches:
+ 20100917__issue16155.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+2010-11-08 22:30 +0000 [r294277-294312] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_timing_timerfd.c: add missing unlock not present in
+ 294277
+
+ * main/timing.c, main/channel.c, res/res_timing_timerfd.c,
+ include/asterisk/timing.h: Fix playback failure when using IAX
+ with the timerfd module. To fix this issue the alert pipe will
+ now be used when the timerfd module is in use. There appeared to
+ be a race that was not solved by adding locking in the timerfd
+ module, but needed to be there anyway. The race was between the
+ timer being put in non-continuous mode in ast_read on the channel
+ thread and the IAX frame scheduler queuing a frame which would
+ enable continuous mode before the non-continuous mode event was
+ read. This race for now is simply avoided. (closes issue #18110)
+ Reported by: tpanton Tested by: tpanton I put tested by tpanton
+ because it was tested on his hardware. Thanks for the remote
+ access to debug this issue!
+
+2010-11-08 20:50 +0000 [r294242] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Go off hold when we get an empty reinvite
+ telling us to. (closes issue 0014448) Reported by: frawd (closes
+ issue #17878) Reported by: frawd
+
+2010-11-05 00:06 +0000 [r293969] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c, /: Merged revisions 293968 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04
+ Nov 2010) | 17 lines codecs/codec_dahdi: Prevent "choppy" audio
+ when receiving unexpected frame sizes. dahdi-linux 2.4.0
+ (specifically commit 9034) added the capability for the wctc4xxp
+ to return more than a single packet of data in response to a
+ read. However, when decoding packets, codec_dahdi was still
+ assuming that the default number of samples was in each read. In
+ other words, each packet your provider sent you, regardless of
+ size, would result in 20 ms of decoded data (30 ms if decoding
+ G723). If your provider was sending 60 ms packets then
+ codec_dahdi would end up stripping 40 ms of data from each
+ transcoded frame resulting in "choppy" audio. This would only
+ affect systems where G729 packets are arriving in sizes greater
+ than 20ms or G723 packets arriving in sizes greater than 30ms.
+ DAHDI-744. ........
+
+2010-11-03 18:31 +0000 [r293806] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 293805 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03
+ Nov 2010) | 20 lines Party A in an analog 3-way call would
+ continue to hear ringback after party C answers. All parties are
+ analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A
+ flash hooks to bring C into 3-way call before C answers. (A and B
+ hear ringback) 4) C answers 5) A continues to hear ringback
+ during the 3-way call. (All parties can hear each other.) * Fixed
+ use of wrong variable in dahdi_bridge() that stopped ringback on
+ the wrong subchannel. * Made several debug messages have more
+ information. A similar issue happens if B and C are SIP channels.
+ B continues to hear ringback. For some reason this only affects
+ v1.8 and trunk. * Don't start ringback on the real and 3-way
+ subchannels when creating the 3-way conference. Removing this
+ code is benign on v1.6.2 and earlier. ........
+
+2010-11-02 23:07 +0000 [r293723] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 293722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010)
+ | 8 lines Add enabled/disabled information for rtautoclear sip
+ show settings output. When setting to zero/"no", the numeric
+ default was shown making it not obvious the disabled setting was
+ respected. (closes issue #18123) Reported by: zerohalo ........
+
+2010-11-02 21:26 +0000 [r293647] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 293639 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02
+ Nov 2010) | 6 lines Make warning message have more useful
+ information in it. Change "Unable to get index, and nullok is not
+ asserted" to "Unable to get index for '<channel-name>' on channel
+ <number> (<function>(), line <number>)". ........
+
+2010-10-30 01:49 +0000 [r293340-293417] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 293416 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29
+ Oct 2010) | 1 line Remove some more code that serves no purpose.
+ ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 293339 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29
+ Oct 2010) | 1 line Remove some code that serves no purpose.
+ ........
+
+2010-10-28 19:54 +0000 [r293195-293196] Tilghman Lesher <tlesher@digium.com>
+
+ * main/ast_expr2.c, main/ast_expr2.h: Merged revisions 293194 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+ | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+ Reported (though the reporter did not understand he was reporting
+ a bug) on the asterisk-users list:
+ http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+ ........
+
+ * /, res/ael/ael.tab.c, main/ast_expr2.y, res/ael/ael_lex.c,
+ res/ael/ael.tab.h: Merged revisions 293194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+ | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+ Reported (though the reporter did not understand he was reporting
+ a bug) on the asterisk-users list:
+ http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+ ........
+
+2010-10-28 16:09 +0000 [r293158] Jeff Peeler <jpeeler@digium.com>
+
+ * funcs/func_strings.c: Fix infinite loop in FILTER(). Specifically
+ when you're using characters above \x7f or invalid character
+ escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes
+ Patches: issue18060_func_strings_filter_infinite_loop.patch
+ uploaded by wdoekes (license 717) Tested by: wdoekes
+
+2010-10-26 18:33 +0000 [r293118] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 293004 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25
+ Oct 2010) | 29 lines Fix inprocess_container in voicemail to
+ correctly restrict max messages. The comparison function logic
+ was off, so the number of sessions for a given mailbox were not
+ being incremented properly. This problem caused the maximum
+ number of messages per folder to not be respected when
+ simultaneously leaving multiple voicemails just below the
+ threshold. These problems should be fixed by the above, but just
+ in case: Fixed resequence_mailbox to rely on the actual number of
+ detected number of files in a directory rather than just assuming
+ only 10 messages more than the maximum had been left. Also if
+ more messages than the maximum are deleted they are actually
+ removed now. The second purpose of this commit should have been
+ separated out probably, but is related to the above. Again, if
+ the number of messages in a given voicemail folder exceeds the
+ maximum set limit make sure to allocate enough space for the
+ deleted and heard index tracking array. A few random fixes: There
+ was a forgotten decrement of the inprocess count in
+ imap_store_file. When using IMAP storage, do not look in the
+ directory where file based storage messages may still reside and
+ influence the message count. Ensure to use only the first format
+ in sendmail. ABE-2516 ........
+
+2010-10-25 19:06 +0000 [r292867] David Vossel <dvossel@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 292866 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25
+ Oct 2010) | 27 lines This patch turns chan_local pvts into
+ astobj2 objects. chan_local does some dangerous things involving
+ deadlock avoidance. tech_pvt functions like hangup and
+ queue_frame are provided with a locked channel upon entry. Those
+ functions are completely safe as long as you don't attempt to
+ give up that channel lock, but that is impossible to guarantee
+ due to the required deadlock avoidance necessary to lock both the
+ tech_pvt and both channels involved. In the past, we have tried
+ to account for this by doing things like setting a "glare" flag
+ that indicates what function should destroy the pvt. This was
+ used in local_hangup and local_queue_frame to decided who should
+ destroy the pvt if they collided in separate threads. I have
+ removed the need to do this by converting all chan_local
+ tech_pvts to astobj2. This means we can ref a pvt before deadlock
+ avoidance and not have to worry about that pvt possibly getting
+ destroyed under us. It also cleans up where we destroy the
+ tech_pvt. The only unlink from the tech_pvt container occurs in
+ local_hangup now, which is where it should occur. Since there
+ still may be thread collisions on some functions like
+ local_hangup after deadlock avoidance, I have added some checks
+ to detect those collisions and exit appropriately. I think this
+ patch is going to solve quite a bit of weirdness we have had with
+ local channels in the past. ........
+
+2010-10-22 21:16 +0000 [r292786] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/asterisk.ldif, channels/chan_sip.c,
+ configs/res_ldap.conf.sample: Update the LDIF file for LDAP. The
+ LDIF file asterisk.ldif was quite a bit out of date from the
+ asterisk.ldap-schema file, so I've now updated that to be in
+ sync. The asterisk.ldif file being out of sync was a problem on
+ my systems where I was doing an ldapadd to import the schema into
+ the LDAP database, and the existing file would cause problems and
+ ERROR messages when registering. Additional documention has been
+ added based on feedback in the issue I'm closing. (closes issue
+ #13861) Reported by: scramatte Patches: ldap-update.txt uploaded
+ by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec,
+ rgenthner
+
+2010-10-21 13:11 +0000 [r292556] Leif Madsen <lmadsen@digium.com>
+
+ * configs/res_ldap.conf.sample: Change res_ldap.sample.conf to
+ match the schema. (closes issue #17376) Reported by: jcovert
+ Patches: res_ldap.conf.sample.patch uploaded by jcovert (license
+ 551)
+
+2010-10-21 00:05 +0000 [r292412] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * apps/app_dial.c, /: Merged revisions 292411 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct
+ 2010) | 10 lines Record priv-recordintro as sln, not gsm This
+ removes the gsm->sln step when transcoding priv-recordintro.
+ (closes issue #18176) Reported by: pabelanger Patches:
+ chan_sip.diff uploaded by pabelanger (license 224) ........
+
+2010-10-18 22:01 +0000 [r292229] Leif Madsen <lmadsen@digium.com>
+
+ * sounds/Makefile: Fix typo in the sounds/Makefile. (Issue #17426)
+
+2010-10-18 21:54 +0000 [r292226] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 292223 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18
+ Oct 2010) | 11 lines Fix improper operator key acceptance and
+ clean up temp recording files. This is a fix for when pressing
+ the operator key after recording an unavailable, busy, name, or
+ temporary message in mailbox options. The operator key should not
+ be accepted here, but should be allowed during the message
+ recording. If the operator key is pressed during ensure the file
+ is saved or deleted as apporopriate. Also, ensure removal of
+ temporary recorded files after an early hang up or when message
+ acceptance confirmation times out. ABE-2518 ........
+
+2010-10-18 21:50 +0000 [r292224] Leif Madsen <lmadsen@digium.com>
+
+ * sounds/Makefile, /, sounds/sounds.xml: Merged revisions 292222
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010)
+ | 9 lines Add support for the new English (Australian Accent)
+ sound files. (closes issue #17426) Reported by: camsown Patches:
+ core-sounds-en_AU.txt uploaded by camsown (license 1050)
+ add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested
+ by: camsown, lmadsen, jtodd, qwell ........
+
+2010-10-16 10:03 +0000 [r292049] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * res/res_musiconhold.c, configs/musiconhold.conf.sample: Base
+ directory for MOH should be ASTDATADIR If the directive
+ 'directory' is relative, make it relative to the datadir, rather
+ than to the varlibdir. In the sample configuration it is relative
+ ('moh'). This has no effect unless you have actively set the
+ datadir explicitly (at build time or at run time). (closes issue
+ #16906) Patches: moh_datadir uploaded by tzafrir (license 46)
+ Review: https://reviewboard.asterisk.org/r/974/
+
+2010-10-15 19:35 +0000 [r291939] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * configs/gtalk.conf.sample, /: Merged revisions 291938 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri, 15 Oct
+ 2010) | 2 lines Clean up formatting. ........
+
+2010-10-15 16:16 +0000 [r291904] Terry Wilson <twilson@digium.com>
+
+ * res/res_jabber.c: Don't crash or deadlock on module unload We
+ can't hold the lock while pthread_join is called since
+ aji_log_hook will attempt to lock from the other therad. We
+ reorder the pthread_join and ast_aji_disconnect so that we don't
+ do an SSL_read() while SSL_shutdown is running, causing a crash.
+
+2010-10-13 23:36 +0000 [r291655] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 291643 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13
+ Oct 2010) | 20 lines Deadlock between dahdi_exception() and
+ dahdi_indicate(). There is a deadlock between dahdi_exception()
+ and dahdi_indicate() for analog ports. The call-waiting and
+ three-way-calling feature can experience deadlock if these
+ features are trying to do something and an event from the bridged
+ channel happens at the same time. Deadlock avoidance code added
+ to obtain necessary channel locks before attemting an operation
+ with call-waiting and three-way-calling. (closes issue #16847)
+ Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch
+ uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+ Review: https://reviewboard.asterisk.org/r/971/ ........
+
+2010-10-13 22:58 +0000 [r291580] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Merged revisions 291577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010)
+ | 21 lines Don't ignore frames that have been queued when
+ softhangup'd When an outgoing call is answered and hung up by the
+ far end *very* quickly, we may not read any frames and therefor
+ end up with a call that displays the wrong
+ disposition/DIALSTATUS. The reason is because ast_queue_hangup()
+ immediately sets the _softhangup flag on the channel and then
+ queues the HANGUP control frame, but __ast_read refuses to read
+ any frames if ast_check_hangup() indicates that a hangup request
+ has been made (which it will if _softhangup is set). So, we end
+ up losing control frames. This change makes __ast_read continue
+ to read frames even if a soft hangup has been requested. It
+ queues a hangup frame to make sure that __ast_read() will still
+ eventually return NULL. Much thanks to David Vossel for all of
+ the reviews, discussion, and help! (closes issue #16946) Reported
+ by: davidw Review: https://reviewboard.asterisk.org/r/740/
+ ........
+
+2010-10-13 15:29 +0000 [r291393] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 291392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010)
+ | 6 lines Lock pvt so pvt->owner can't disappear when queueing up
+ a frame. This fixes a crash due to a hangup race condition.
+ ABE-2601 ........
+
+2010-10-12 17:20 +0000 [r291280] Leif Madsen <lmadsen@digium.com>
+
+ * configs/phoneprov.conf.sample: Add undocumented variables to
+ phoneprov.conf.sample (closes issue #18107) Reported by: lathama
+ Patches: phoneprov.conf.sample.diff uploaded by lathama (license
+ 1028)
+
+2010-10-12 17:05 +0000 [r291264] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/acl.c: Merged revisions 291263 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 Oct 2010)
+ | 2 lines Oops, incorrect range (although unallocated at ARIN)
+ ........
+
+2010-10-12 16:07 +0000 [r291229] Leif Madsen <lmadsen@digium.com>
+
+ * configs/manager.conf.sample: Add documention that mentions
+ options are defined but not used. (Issue #18101)
+
+2010-10-11 18:39 +0000 [r291073-291111] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Make exit from handle_request_do()
+ consistent.
+
+ * /, channels/chan_sip.c: Merged revisions 291109 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010)
+ | 1 line Add missing unlock to an exception condition in
+ reload_config(). ........
+
+ * main/cli.c: Fixed infinite loop in verbose/debug message output.
+ Setting the module/filename specific message level and then
+ changing it resulted in the linked list being looped on itself.
+ Traversing this linked list is an infinite loop if what you are
+ looking for is not in the list. Also plugged some CLI parsing
+ holes in the associated CLI command: * Removing a nonexistent
+ module from the list actually added it with a level of zero. *
+ Setting the non-module specific level to zero is now equivalent
+ to setting it to "off" as documented.
+
+2010-10-08 02:45 +0000 [r290863] Jeff Peeler <jpeeler@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 290862 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010)
+ | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed
+ at control console. A recent change was made to avoid a race
+ condition on shutdown which only called the end functions from
+ the console thread. However, when pressing Ctrl-C the quit
+ handler is called from the signal handler thread. (closes issue
+ #17698) Reported by: jmls ........
+
+2010-10-07 20:57 +0000 [r290751] Jason Parker <jparker@digium.com>
+
+ * autoconf/ast_ext_lib.m4, /, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 290750 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) |
+ 9 lines Allow PRI to build properly when using --with-pri. Use
+ the directories found for the parent when using lib dependencies.
+ (closes issue #17314) Reported by: tzafrir Patches:
+ 17314-withdeps.diff uploaded by qwell (license 4) ........
+
+2010-10-07 10:53 +0000 [r290712] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Don't crash when Set() is called without a value.
+ Review: https://reviewboard.asterisk.org/r/949/
+
+2010-10-06 13:48 +0000 [r290396-290575] Tilghman Lesher <tlesher@digium.com>
+
+ * main/file.c: Allow streaming audio from a pipe. (closes issue
+ #18001) Reported by: jamicque Patches:
+ 20100926__issue18001.diff.txt uploaded by tilghman (license 14)
+ Tested by: jamicque
+
+ * res/res_jabber.c, /: Merged revisions 290392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010)
+ | 8 lines Fix a crash by ensuring that we don't alter memory
+ after it's freed. (closes issue #17387) Reported by: jmls
+ Patches: 20100726__issue17387.diff.txt uploaded by tilghman
+ (license 14) Tested by: jmls ........
+
+2010-10-05 19:54 +0000 [r290375] David Vossel <dvossel@digium.com>
+
+ * apps/app_directed_pickup.c: Fixes PickupChan() not working with
+ full channel name. (closes issue #18011) Reported by: schern
+ Patches: app_directed_pickup.c.2.patch uploaded by schern
+ (license 995) app_directed_pickup.c.trunk.patch uploaded by
+ schern (license 995) Tested by: schern, dvossel
+
+2010-10-05 17:42 +0000 [r290324] Richard Mudgett <rmudgett@digium.com>
+
+ * contrib/valgrind.supp, /: Merged revisions 290323 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r290323 | rmudgett | 2010-10-05 12:41:18 -0500
+ (Tue, 05 Oct 2010) | 11 lines Merged revision 258974 from
+ https://origsvn.digium.com/svn/asterisk/trunk .......... r258974
+ | diruggles | 2010-04-26 14:05:47 -0500 (Mon, 26 Apr 2010) | 4
+ lines Line 24 missed in compatibility fix in revision 233577
+ added a "fun:" prefix line 24 .......... ................
+
+2010-10-04 23:14 +0000 [r290101-290254] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, main/pbx.c,
+ pbx/ael/ael-test/ref.ael-vtest17,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+ pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
+ pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5:
+ Change new pattern matcher to regard dashes the same as the old
+ pattern matcher -- as visual candy to be ignored. Also change the
+ AEL parser to not generate dashes within extensions, as those
+ dashes would be ignored. Update the AEL tests to match this
+ behavior. (closes issue #17366) Reported by: murf Patches:
+ 20100727__issue17366.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+ * /, configure, configure.ac: Merged revisions 290177 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04
+ Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........
+
+ * /, configure, configure.ac: Merged revisions 290100 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03
+ Oct 2010) | 2 lines Automatically re-run configure test for
+ menuselect, when the relevant makeopts settings change. ........
+
+2010-10-02 08:52 +0000 [r289950] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c, /: Merged revisions 289949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2
+ lines Add documentation for undocumented option to AMI action
+ originate ........
+
+2010-10-02 04:45 +0000 [r289874] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 289873 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01
+ Oct 2010) | 8 lines When forwarding a message, a prepend means
+ that the filesystem will always have a better copy. (closes issue
+ #17803) Reported by: dpetersen Patches:
+ 20100923__issue17803.diff.txt uploaded by tilghman (license 14)
+ Tested by: dpetersen ........
+
+2010-10-01 23:01 +0000 [r289798] Jeff Peeler <jpeeler@digium.com>
+
+ * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h:
+ Merged revisions 289797 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010)
+ | 15 lines Change RFC2833 DTMF event duration on end to report
+ actual elapsed time. The scenario here is with a non P2P early
+ media session. The reported time length of DTMF presses are
+ coming up short when sending to the remote side. Currently the
+ event duration is a running total that is incremented when
+ sending continuation packets. These continuation packets are only
+ triggered upon incoming media from the remote side, which means
+ that the running total probably is not going to end up matching
+ the actual length of time Asterisk received DTMF. This patch
+ changes the end event duration to be lengthened if it is detected
+ that the end event is going to come up short. Review:
+ https://reviewboard.asterisk.org/r/957/ ABE-2476 ........
+
+2010-10-01 17:09 +0000 [r289704] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * res/res_jabber.c, /, configs/jabber.conf.sample: Merged revisions
+ 289703 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct
+ 2010) | 6 lines Disable debugging by default and reformat .config
+ file. Review: https://reviewboard.asterisk.org/r/929/ ........
+
+2010-10-01 16:21 +0000 [r289700] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 289699 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010)
+ | 14 lines Ensure user portion of SIP URI matches dialplan when
+ using encoded characters. This commit takes a simliar approach to
+ 288112 and checks the dialplan to determine the proper action for
+ an incoming contact header as to whether or not it should be
+ decoded or not. sip_new was blindly always decoding the
+ extension, which also caused the outgoing contact header to be
+ incorrect as well as failing to match the encoded extension in
+ the dialplan. (closes issue #17892) Reported by: wdoekes Patches:
+ bug17892-1.patch uploaded by jpeeler (license 325) Tested by:
+ wdoekes ........
+
+2010-10-01 09:42 +0000 [r289622] schmitds <schmitds@localhost>:
+
+ * channels/chan_sip.c: don't iterate through all dialogs to find
+ and delete old subscribes On every incoming subscribe there is a
+ iteration through all dialogs to find old subscribes and delete
+ them. This is slow and not RFC conform. This was only needed in
+ 1.2 cause a subscribe was not deleted when a dialog was
+ destroyed, after 1.4 a subscribe get removed when its dialog is
+ destroyed. (closes issue #17950) Reported by: schmidts Tested by:
+ schmidts Review: https://reviewboard.asterisk.org/r/901/
+
+2010-09-30 19:51 +0000 [r289553] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Properly handle channel allocation failures
+ duing invites with replaces. ABE-2588
+
+2010-09-30 17:09 +0000 [r289501] Brett Bryant <bbryant@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 289500 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289500 | bbryant | 2010-09-30 13:08:20 -0400 (Thu, 30 Sep 2010)
+ | 11 lines res_agi.c:handle_getvariablefull() could recursively
+ lock a channel and not release it if an argument is the current
+ channel's name. (closes issue #17970) Reported by: mdu113
+ Patches: res_agi.c.diff3 uploaded by mdu113 (license 582) Tested
+ by: mdu113 Review: https://reviewboard.asterisk.org/r/947/
+ ........
+
+2010-09-30 15:37 +0000 [r289425] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_sms.c: Merged revisions 289424 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010)
+ | 8 lines Fix a crash in app_sms. Since the data being passed to
+ the generator callback is on the stack of the SMS() application,
+ we must ensure that the generator is stopped before the
+ application exits. ABE-2587 ........
+
+2010-09-29 21:03 +0000 [r289339] Jason Parker <jparker@digium.com>
+
+ * main/channel.c, /, main/features.c: Merged revisions 289338 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) |
+ 8 lines Allow a manager originate to succeed on forwarded
+ devices. The timeout to wait for an answer was being set to 0
+ when a device forwarded to another extension. We don't always
+ need the timeout set like this, so make it an optional parameter,
+ and don't use it in this case. ABE-2544 ........
+
+2010-09-29 20:24 +0000 [r289334] Leif Madsen <lmadsen@digium.com>
+
+ * configs/res_ldap.conf.sample: Update sample documentation to note
+ md5secret requirements.
+
+2010-09-29 20:15 +0000 [r289332] Russell Bryant <russell@digium.com>
+
+ * res/res_config_ldap.c: Don't completely ignore md5secret from
+ LDAP if the value does not begin with {md5}. This fixes a problem
+ that lmadsen ran in to where md5secret was not working for him.
+
+2010-09-29 15:04 +0000 [r289178] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 289177 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep
+ 2010) | 8 lines Set the caller id on CDRs when it is set on the
+ parent channel. (closes issue #17569) Reported by: tbelder
+ Patches: 17569.diff uploaded by tbelder (license 618) Tested by:
+ tbelder ........
+
+2010-09-28 18:14 +0000 [r289095] Brett Bryant <bbryant@digium.com>
+
+ * main/channel.c, /: Merged revisions 289094 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010)
+ | 14 lines Fixes an issue with the Newchannel AMI event during
+ the Masquerading process. Fixes an issue with the Newchannel AMI
+ event during the Masquerading process, where no Newchannel AMI
+ event was generated for the psuedo channel used during the
+ masquerading process. (closes issue #17987) Reported by:
+ RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish
+ (license 1122) Tested by: RadicAlish Review:
+ https://reviewboard.asterisk.org/r/937/ ........
+
+2010-09-24 15:37 +0000 [r288747] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 288746 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24
+ Sep 2010) | 5 lines Don't fail a masquerade if it is already
+ being hung up This avoids noise on some Local channel situations
+ where we don't use /n. Thanks to Alec Davis for the suggestion.
+ ........
+
+2010-09-24 13:53 +0000 [r288637-288712] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_strings.c: Solaris won't printf a NULL. (closes issue
+ #18041) Reported by: asgaroth
+
+ * cdr/cdr_pgsql.c, /, configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, configure.ac,
+ include/asterisk/channel.h: Merged revisions 288636 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23
+ Sep 2010) | 2 lines Solaris compatibility fixes ........
+
+2010-09-22 23:10 +0000 [r288500] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 288499 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22
+ Sep 2010) | 8 lines Don't let a Local channel get bridged to
+ itself If a local channel gets bridged to itself, it becomes
+ orphaned with no devices left to actually tell it to hang up.
+ This patch modifies local_fixup() to detect this case and deny
+ it. Review: https://reviewboard.asterisk.org/r/934 ........
+
+2010-09-22 17:49 +0000 [r288344-288417] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288416 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010)
+ | 5 lines RFC3261 section 12.2 explicitly says out of order
+ requests are responded with a 500 Server Internal Error response.
+ ABE-2458 ........
+
+ * /, channels/chan_sip.c: Merged revisions 288343 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010)
+ | 2 lines During check_pendings, if the dialog is terminated with
+ a CANCEL, change the invitestate to INV_CANCEL like in
+ sip_hangup. ........
+
+2010-09-22 16:44 +0000 [r288340] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 288339 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010)
+ | 11 lines Fix a 100% CPU consumption problem when setting
+ console=yes in asterisk.conf. The handling of -c and console=yes
+ should be the same, but they were not. When you specify -c, it
+ sets both a flag for console module and for asterisk not to
+ fork() off into the background. The handling of console=yes only
+ set console mode, so you would end up with a background process()
+ trying to run the Asterisk console and freaking out since it
+ didn't have anything to read input from. Thanks to beagles for
+ reporting and helping debug the problem! ........
+
+2010-09-22 15:11 +0000 [r288267] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample, /, UPGRADE.txt:
+ Merged revisions 288265-288266 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010)
+ | 9 lines Allow the encoding to be set, in case local charset
+ does not agree with database. (closes issue #16940) Reported by:
+ jamicque Patches: 20100827__issue16940.diff.txt uploaded by
+ tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt
+ uploaded by tilghman (license 14) Tested by: jamicque ........
+ r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010)
+ | 5 lines Document addition of encoding parameter. (issue #16940)
+ Reported by: jamicque ........
+
+2010-09-22 00:03 +0000 [r288193] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 288192 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21
+ Sep 2010) | 26 lines In chan_iax2.c:schedule_delivery() calls
+ ast_bridged_channel() on an unlocked channel. Near the beginning
+ of schedule_delivery(), ast_bridged_channel() is called on
+ iaxs[fr->callno]->owner. However, the channel is not locked,
+ which can result in ast_bridged_channel() crashing should
+ owner->tech change to a technology that doesn't implement
+ bridged_channel. I also fixed the other calls to
+ ast_bridged_channel() in chan_iax2.c since the owner lock was not
+ held there either. Converted the existing channel deadlock
+ avoidance to use iax2_lock_owner(). Using the new function
+ simplified some awkward code. In the process of fixing the
+ locking on ast_bridged_channel(), I also found a memory leak in
+ socket_process() for v1.6.2 and v1.8. The local struct variable
+ ies.vars is not freed on early/abnormal function exits. (closes
+ issue #17919) Reported by: rain Patches: issue17919_v1.4.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch
+ uploaded by rmudgett (license 664) Review:
+ https://reviewboard.asterisk.org/r/926/ ........
+
+2010-09-21 22:22 +0000 [r288147] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_iax2.c: Setup timer before set_config(). (closes
+ issue #18019) Reported by: Netview Patches: issue_0018019.patch
+ uploaded by pabelanger (license 224) Tested by: Netview
+
+2010-09-21 21:59 +0000 [r288113] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288112 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010)
+ | 15 lines Try both the encoded and unencoded subscription URI
+ for a match in hints. When a phone sends an encoded URI for a
+ subscription, the URI is not matched with the actual hint that is
+ in decoded format. For example, if we have an extension with a
+ hint that is named: "#5601" or "*5601", the subscription will
+ work fine if the phone subscribes with an already decoded URI,
+ but when it's decoded like "%255601" or "%2A5601", Asterisk is
+ unable to match it with the correct hint. (closes issue #17785)
+ Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt
+ uploaded by tilghman (license 14) Tested by: ramonpeek ........
+
+2010-09-21 19:46 +0000 [r288006] Brett Bryant <bbryant@digium.com>
+
+ * main/channel.c, /: Merged revisions 288005 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010)
+ | 8 lines Add a check to fix a rare segmentation fault you'd get
+ if ast_frdup couldn't allocate memory on the first frame being
+ queued in ast_queue_frame. (closes issue #17882) Reported by:
+ seanbright Tested by: seanbright ........
+
+2010-09-21 19:07 +0000 [r287934] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 287933 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21 Sep 2010)
+ | 2 lines Less than zero is an error, not any non-zero value.
+ ........
+
+2010-09-20 23:58 +0000 [r287759] Brett Bryant <bbryant@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 287758 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010)
+ | 16 lines Fix misvalidation of meetme pins in conjunction with
+ the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a
+ user and admin pin setup for your conference, using the user pin
+ would gain you admin priviledges. Also, when no user pin was set,
+ an admin pin was, the 'a' MeetMe flag wasn't used, and the user
+ tried to enter a conference then they were still prompted for a
+ pin and forced to hit #. (closes issue #17908) Reported by: kuj
+ Patches: pins_2.patch uploaded by kuj (license 1111) Tested by:
+ kuj Review: [full review board URL with trailing slash] ........
+
+2010-09-20 23:16 +0000 [r287685] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/channel.c: ast_channel_masquerade: Avoid recursive
+ masquerades. Check all 4 combinations of (original/clonechan) *
+ (masq/masqr). Initially original->masq and clonechan->masqr were
+ only checked. It's possible with multiple masq's planned - and
+ not yet executed, that the 'original' chan could already have
+ another masq'd into it - thus original->masqr would be set, that
+ masqr would lost. Likewise for the clonechan->masq. (closes issue
+ #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches:
+ based on bug16057.diff4.txt uploaded by alecdavis (license 585)
+ Tested by: ramonpeek, davidw, alecdavis
+
+2010-09-20 21:28 +0000 [r287642] Jason Parker <jparker@digium.com>
+
+ * channels/chan_skinny.c: Don't crash when parking a non-bridged
+ call. (closes issue #17680) Reported by: jmhunter Patches:
+ chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by:
+ jmhunter, DEA
+
+2010-11-02 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.14 Released.
+
+2010-09-20 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.14-rc1 Released.
+
+2010-09-20 15:56 +0000 [r287556-287558] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Use ast_str when processing hint state changes
+ Merged revisions 287555 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
+ 2010) | 5 lines Use ast_dynamic_str when processing hint state
+ changes (related to issue #17928) Reported by: mdu113 ........
+
+ * /: Revert r287556.
+
+ * /: Use ast_str when processing hint state changes Merged
+ revisions 287555 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
+ 2010) | 5 lines Use ast_dynamic_str when processing hint state
+ changes (related to issue #17928) Reported by: mdu113 ........
+
+2010-09-19 16:06 +0000 [r287470] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c, /: Merged revisions 287469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7
+ lines Make sure we always free variables properly in manager
+ originate. (closes issue #17891) reported, solved and tested by
+ oej Review: https://reviewboard.asterisk.org/r/869/ ........
+
+2010-09-17 21:08 +0000 [r287387] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 287386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010)
+ | 7 lines Blank columns should get set on reload, not ignored.
+ (closes issue #16893) Reported by: haakon Patches:
+ 20100818__issue16893.diff.txt uploaded by tilghman (license 14)
+ ........
+
+2010-09-17 13:36 +0000 [r287308] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 287307 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep
+ 2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while
+ processing in ast_hint_state_changed(). (related to issue #17928)
+ Reported by: mdu113 ........
+
+2010-09-16 22:12 +0000 [r287198] Jason Parker <jparker@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 287197 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287197 | qwell | 2010-09-16 17:12:30 -0500 (Thu, 16 Sep 2010) |
+ 7 lines Add LSB headers for Debian init script, since Debian will
+ complain if it isn't there. Headers were taken from trunk.
+ (closes issue #17958) Reported by: javyer ........
+
+2010-09-16 20:06 +0000 [r287115-287119] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 287118 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep
+ 2010) | 8 lines Don't limit hint processing in
+ ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
+ (closes issue #17928) Reported by: mdu113 Patches:
+ 20100831__issue17928.diff.txt uploaded by tilghman (license 14)
+ Tested by: mdu113 ........
+
+ * main/cdr.c, /: Merged revisions 287114 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep
+ 2010) | 8 lines Don't stop printing cdr variables if we encounter
+ one with a blank name or value. (closes issue #17900) Reported
+ by: under Patches: core-show-channel-cdr-fix1.diff uploaded by
+ mnicholson (license 96) Tested by: mnicholson ........
+
+2010-09-15 20:28 +0000 [r286998] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 286941 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15
+ Sep 2010) | 7 lines Ensure mailbox is not filled to capacity
+ before doing message forwarding. Specifically, before prompting
+ to record a prepended message the capacity is checked first. If
+ the mailbox is full the extension will be reprompted. ABE-2517
+ ........
+
+2010-09-14 19:27 +0000 [r286681-286757] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 286756 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep
+ 2010) | 13 lines Don't clear the username from a realtime
+ database when a registration expires. Non-realtime chan_sip does
+ not clear the username from memory when a registration expiries
+ so realtime probably shouldn't either. (closes issue #17551)
+ Reported by: ricardolandim Patches:
+ reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license
+ 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson
+ (license 96) reg-expiry-username-1.8-fix1.diff uploaded by
+ mnicholson (license 96) reg-expiry-username-trunk-fix1.diff
+ uploaded by mnicholson (license 96) Tested by: ricardolandim,
+ mnicholson ........
+
+ * main/channel.c, /: Merged revisions 286679 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep
+ 2010) | 7 lines Only drop duplicate answer frames if the channel
+ is bridged. Back in r3710 ast_read() was modified to drop answer
+ frames on channels that were in the UP state. This modification
+ prevented bridges that were up before the answer from being
+ broken and reestablished by an ANSWER control frame. That change
+ also prevents pickup of channels called from the ast_dial
+ framework from working properly. The ast_dial framework expects
+ to see an ANSWER frame after dialing and the pickup code queues
+ one but ast_read() drops it. This new change only drops ANSWER
+ frames when the channel is bridged, allowing the answer queued by
+ the pickup code to properly pass through ast_read() on to the
+ ast_dial framework. ABE-2473 (related to issue #2342) ........
+
+2010-09-14 05:06 +0000 [r286527-286587] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/realtime/mysql/voicemail_messages.sql (added),
+ contrib/realtime/mysql/voicemail_data.sql (added): Add
+ documentation on missing backend tables for Voicemail
+
+ * main/features.c: C precedence got me
+
+ * main/features.c: Refactor conversion to ast_poll() to fix
+ callparking regression.
+
+2010-09-13 19:38 +0000 [r286456] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Remove "Internal IP" from sip show settings,
+ as it's not at all useful to display. (closes issue #17840)
+ Reported by: oej
+
+2010-09-11 17:05 +0000 [r286268] Olle Johansson <oej@edvina.net>
+
+ * /, main/file.c: Merged revisions 286267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4
+ lines Handle error response when we can't make file compatible
+ Review: https://reviewboard.asterisk.org/r/911/ ........
+
+2010-09-10 22:56 +0000 [r286223] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 286222 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r286222 | twilson | 2010-09-10 17:54:23 -0500 (Fri, 10
+ Sep 2010) | 1 line Return -1 if chan_local doesn't support an
+ option ........
+
+2010-09-10 20:55 +0000 [r286117] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 286114 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri,
+ 10 Sep 2010) | 4 lines Load iax.conf before registering any
+ functions/applications/actions. Review:
+ https://reviewboard.asterisk.org/r/914/ ........
+
+2010-09-10 20:42 +0000 [r286116] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 286113 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10
+ Sep 2010) | 11 lines An outgoing call may not get hung up if a
+ pre-connect incoming ISDN call is disconnected. If the ISDN link
+ a pre-connect incoming call is using fails or is reset, the
+ outgoing leg may not hang up or be delayed in hanging up.
+ (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER,
+ PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
+ PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the
+ incoming call leg hangs up before connecting for any reason. It
+ makes no sense to send a BUSY or CONGESTION control frame to the
+ outgoing call leg under these circumstances. ........
+
+2010-09-10 20:35 +0000 [r286115] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/pbx.h, include/asterisk/frame.h,
+ channels/chan_local.c, /, funcs/func_channel.c,
+ include/asterisk/channel.h: Merged revisions 286059 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10
+ Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a
+ Local channel Having Local (/n) channels as queue members and
+ setting the language in the extension with
+ Set(CHANNEL(language)=fr) sets the language on the Local/...,2
+ channel. Hold time report playbacks happen on the Local/...,1
+ channel and therefor do not play in the specified language. This
+ patch modifies func_channel_write to call the setoption callback
+ and pass the CHANNEL() write info to the callback. chan_local
+ uses this information to look up the other side of the channel
+ and apply the same changes to it. (closes issue #17673) Reported
+ by: Guggemand Review: https://reviewboard.asterisk.org/r/903/
+ ........
+
+2010-09-10 18:30 +0000 [r285930-286024] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_heap.c, /, main/test.c: Merged revisions 286023 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010)
+ | 2 lines Missing newline ........
+
+ * include/asterisk/select.h: Another fix for Mac OS X. While trying
+ to fix this the "right" way, I wandered into dependency hell. Two
+ hours later, I backed out, and just removed the offending code.
+ ast_inline_api only goes one level deep and then it breaks. Ouch.
+
+ * tests/test_poll.c, include/asterisk/select.h, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 285889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010)
+ | 7 lines Fix Mac OS X build. This also fixes a rather grievous
+ calculation error for the offset of ast_fdset, which was masked
+ on Linux and FreeBSD, because these platforms check the first 256
+ FDs regardless of the bitmask setting (due to backwards
+ compatibility). ........
+
+2010-09-09 22:49 +0000 [r285818] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, codecs/gsm/Makefile: Merged revisions 285817 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep
+ 2010) | 8 lines GCC 4.2.x optimizations result in improper
+ behavior of GSM codec (closes issue #17688) Reported by:
+ pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by
+ pprindeville (license 347) Tested by: mkeuter, pprindeville
+ ........
+
+2010-09-09 20:09 +0000 [r285744] Jason Parker <jparker@digium.com>
+
+ * main/channel.c, /: Merged revisions 285742 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) |
+ 9 lines Transmit silence when reading DTMF in ast_readstring.
+ Otherwise, you could get issues with DTMF timeouts causing
+ hangups. (closes issue #17370) Reported by: makoto Patches:
+ channel-readstring-silence-generator.patch uploaded by makoto
+ (license 38) ........
+
+2010-09-09 18:50 +0000 [r285639-285710] Brett Bryant <bbryant@digium.com>
+
+ * main/pbx.c: Fixes an issue with dialplan pattern matching where
+ the specificity for pattern ranges and pattern special characters
+ was inconsistent. (closes issue #16903) Reported by: Nick_Lewis
+ Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license
+ 657) Tested by: Nick_Lewis
+
+ * res/res_musiconhold.c, /: Merged revisions 285638 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09
+ Sep 2010) | 7 lines Fixes an issue with MOH where it doesn't
+ recover cleanly when it can't play a file and would just stop,
+ instead of continuing to find the next playable file in the MOH
+ class. (closes issue #17807) Reported by: kshumard Review:
+ https://reviewboard.asterisk.org/r/910/ ........
+
+2010-09-08 22:11 +0000 [r285563-285567] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 285566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010)
+ | 2 lines In retrans_pkt, do not unlock pvt until the end of the
+ function on a transmit failure. ........
+
+ * channels/chan_sip.c: Fixes interoperability problems with session
+ timer behavior in Asterisk. CHANGES: 1. Never put "timer" in
+ "Require" header. This is not to our benefit and RFC 4028 section
+ 7.1 even warns against it. It is possible for one endpoint to
+ perform session-timer refreshes while the other endpoint does not
+ support them. If in this case the end point performing the
+ refreshing puts "timer" in the Require field during a refresh,
+ the dialog will likely get terminated by the other end. 2. Change
+ the behavior of 'session-timer=accept' in sip.conf (which is the
+ default behavior of Asterisk with no session timer configuration
+ specified) to only run session-timers as result of an incoming
+ INVITE request if the INVITE contains an "Session-Expires"
+ header... Asterisk is currently treating having the "timer"
+ option in the "Supported" header as a request for session timers
+ by the UAC. I do not agree with this. Session timers should only
+ be negotiated in "accept" mode when the incoming INVITE supplies
+ a "Session-Expires" header, otherwise RFC 4028 says we should
+ treat a request containing no "Session-Expires" header as a
+ session with no expiration. Below I have outlined some situations
+ and what Asterisk's behavior is. The table reflects the behavior
+ changes implemented by this patch. SITUATIONS: -Asterisk as UAS
+ 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS
+ "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO
+ "Session-Expires". 200 Ok Response HAS "Session-Expires" header
+ 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO
+ "Session-Expires" header 5. Outgoing INVITE: HAS
+ "Session-Expires". Active - Asterisk will have an active refresh
+ timer regardless if the other endpoint does. Inactive - Asterisk
+ does not have an active refresh timer regardless if the other
+ endpoint does. XXXXXXX - Not possible for mode.
+ ______________________________________ |SITUATIONS |
+ 'session-timer' MODES | |___________|________________________| |
+ | originate | accept | |-----------|------------|-----------| |1.
+ | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX |
+ Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX |
+ -------------------------------------- (closes issue #17005)
+ Reported by: alexrecarey
+
+2010-09-08 20:56 +0000 [r285532] Brett Bryant <bbryant@digium.com>
+
+ * apps/app_meetme.c: Fixes a bug with MeetMe where after announcing
+ the amount of time left in a conference, if music on hold was
+ playing, it doesn't restart. (closes issue #17408) Reported by:
+ sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by
+ sysreq (license 1009) Tested by: sysreq
+
+2010-09-08 20:42 +0000 [r285526-285529] Jason Parker <jparker@digium.com>
+
+ * res/res_musiconhold.c, include/asterisk/astobj2.h: Follow coding
+ guidelines in moh rescan fix. Also fix the documentation that got
+ me in trouble.
+
+ * res/res_musiconhold.c: Fixes issue where moh files were no longer
+ rescanned during a reload. (closes issue #16744) Reported by: pj
+ Patches: 16744-reload.diff uploaded by qwell (license 4) Tested
+ by: qwell
+
+2010-09-07 20:31 +0000 [r285267-285366] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 285365 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010)
+ | 9 lines Catch invalid extensions at the parser, instead of
+ making the core deal with them. (closes issue #17794) Reported
+ by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded
+ by tilghman (license 14) 20100820__issue17794__1.4.diff.txt
+ uploaded by tilghman (license 14) Tested by: PavelL ........
+
+ * main/poll.c, /: Merged revisions 285266 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010)
+ | 4 lines Use poll, if indicated to do so, in the ast_poll2
+ implementation. This fixes the unit tests on FreeBSD 8.0.
+ ........
+
+2010-09-07 17:49 +0000 [r285196] Brett Bryant <bbryant@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 285194 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07
+ Sep 2010) | 10 lines Fixes voicemail.conf issues where mailboxes
+ with passwords that don't precede a comma would throw unnecessary
+ error messages. (closes issue #15726) Reported by: 298 Patches:
+ M15726.diff uploaded by junky (license 177) Tested by: junky
+ Review: [full review board URL with trailing slash] ........
+
+2010-09-06 06:55 +0000 [r285089] Tilghman Lesher <tlesher@digium.com>
+
+ * makeopts.in, /, BSDmakefile (added): Merged revisions 285088 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 Sep 2010)
+ | 2 lines Silly convenience script for BSD platforms. ........
+
+2010-09-03 18:15 +0000 [r284958] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_iax2.c: This is a patch provided for issue #17935
+ to add the ActionID to the IAXregistry AMI response. (closes
+ issue #17935) Reported by: alexkuklin Patches: iaxshowreg
+ uploaded by alexkuklin (license 1115) Tested by: alexkuklin
+
+2010-09-03 16:20 +0000 [r284897] Terry Wilson <twilson@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 284881 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010)
+ | 5 lines Properly detect when a sound file doesn't exist
+ ast_fileexists returns -1 for error and 0 for a non-existant
+ file. The existing code treated missing files as though they
+ existed. ........
+
+2010-09-02 20:54 +0000 [r284778] Brett Bryant <bbryant@digium.com>
+
+ * main/manager.c, /: Merged revisions 284777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010)
+ | 7 lines Fixes a bug in manager.c where the default
+ configuration values weren't reset when the manager configuration
+ was reloaded. (closes issue #17917) Reported by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/883/ ........
+
+2010-09-02 16:48 +0000 [r284704] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 284703 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010)
+ | 7 lines Removed relatedpeer code from sip_autodestruct Handling
+ of the relatedpeer structure associated with a sip_pvt should be
+ done during the final sip_destruction function, not in
+ sip_autodestruct. ........
+
+2010-09-02 16:07 +0000 [r284399-284665] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_usbradio.c: Fixing build.
+
+ * apps/app_queue.c: Don't reset queue stats on a module reload.
+ (closes issue #17535) Reported by: raarts Patches:
+ 20100819__issue17535.diff.txt uploaded by tilghman (license 14)
+
+ * configure, include/asterisk/autoconfig.h.in: Failed to rerun
+ bootstrap.sh after last commit
+
+ * res/res_jabber.c, main/rtp.c, main/poll.c,
+ include/asterisk/select.h (added), channels/chan_usbradio.c,
+ channels/chan_phone.c, channels/chan_misdn.c, main/features.c,
+ include/asterisk/poll-compat.h, tests/test_poll.c (added),
+ main/asterisk.c, utils/clicompat.c, res/res_ais.c, /,
+ configure.ac, channels/console_video.c,
+ include/asterisk/channel.h: Merged revisions 284478 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01
+ Sep 2010) | 11 lines Ensure that all areas that previously used
+ select(2) now use poll(2), with implementations that need poll(2)
+ implemented with select(2) safe against 1024-bit overflows. This
+ is a followup to the fix for the pthread timer in 1.6.2 and
+ beyond, fixing a potential crash bug in all supported releases.
+ (closes issue #17678) Reported by: russell Branch:
+ https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
+ Review: https://reviewboard.asterisk.org/r/824/ ........
+
+ * res/res_config_pgsql.c: Don't warn on floats and timestamps
+ (closes issue #17082) Reported by: coolmig
+
+ * /, channels/chan_sip.c: Merged revisions 284393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010)
+ | 7 lines Don't send a devstate change on poke_noanswer if the
+ state did not change. (closes issue #17741) Reported by: schmidts
+ Patches: chan_sip.c.patch uploaded by schmidts (license 1077)
+ ........
+
+2010-08-31 18:59 +0000 [r284317] Leif Madsen <lmadsen@digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 284316 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31
+ Aug 2010) | 7 lines Update say.conf.sample to match the rules in
+ say.c (closes issue #17835) Reported by: RoadKill Patches:
+ say.conf.sample.patch.rules uploaded by RoadKill (license 933)
+ Tested by: RoadKill ........
+
+2010-08-30 22:27 +0000 [r284280] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_festival.c: Fix 3 coding errors: 1) After we close FD,
+ we should not be trying to write to it. 2) Call _exit(0), not
+ exit(0), to avoid running shutdown routines in a child. 3) Use
+ endian, not processor, detection to ensure bytes are written in
+ the correct order. (closes issue #15706) Reported by: modelnine
+ Patches: asterisk-1.6.1.1-festival-debug.patch uploaded by
+ modelnine (license 865) Tested by: gmartinez
+
+2010-08-27 22:27 +0000 [r284002] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010)
+ | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests.
+ (closes issue #17758) Reported by: ibc Patches:
+ multiple_accept_headers_1.4.diff uploaded by dvossel (license
+ 671) ........
+
+2010-08-27 20:30 +0000 [r283881] Jason Parker <jparker@digium.com>
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
+ revisions 283880 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) |
+ 8 lines Fix issue with decoding ^-escaped characters in realtime.
+ (closes issue #17790) Reported by: denzs Patches:
+ 17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell,
+ denzs ........
+
+2010-08-26 15:24 +0000 [r283381-283691] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283690 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010)
+ | 19 lines Fixed how Asterisk destroys a dialog on channel hangup
+ before invite receives a response. If an ast_channel with a SIP
+ tech pvt hangs up before the sip dialog gets a response to its
+ outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is
+ not rfc compliant and results in confusion at the other endpoint.
+ sip_pretend_ack will ack and remove all the packets in the
+ retransmit queue. This means that the INVITE will stop
+ retransmitting, and that any response to that INVITE that comes
+ after the pretend_ack occurs will be ignored. Instead of faking
+ any sort of acknowledgement for an outgoing INVITE during an
+ internal hangup, we should let the protocol stack process the
+ INVITE transaction and terminate the dialog properly. This is
+ achieved by setting the PENDING_BYE flag. When this flag is used,
+ once the dialog proceeds to an escapable state the transaction
+ will either be canceled with a SIP_CANCEL or completed followed
+ immediately by a BYE. Attempting to do this any other way is
+ incorrect. If the endpoint is not responding to the INVITE
+ request, the INVITE must continue to be retransmitted until it
+ times out which will result in the dialog being destroyed.
+ ........
+
+ * channels/chan_sip.c: Add to and from tags to NOTIFY dialog-info
+ xml body so pickup can occur. When pedantic mode is used, the
+ dialog-info xml generated during a ringing event must contain the
+ to and from tag values. Otherwise if a pickup occurs using INVITE
+ with replaces, Astrisk will not be able to locate the
+ subscription.
+
+ * channels/chan_sip.c: Asterisk will not advertise session timers
+ are supported when 'session-timers=refuse' is used. Asterisk now
+ dynamically builds the "Supported" header depending on what is
+ enabled/disabled in sip.conf. Session timers used to always be
+ advertised as being supported even when they were disabled in the
+ configuration. This caused problems with some end points. (issue
+ #17005)
+
+ * /, channels/chan_sip.c: Merged revisions 283380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010)
+ | 11 lines This fix makes sure the ast_channel hangs up correctly
+ when the dialog's PENDING_BYE flag is set. When the pending bye
+ flag is used, it is possible that the dialog will terminate and
+ leave the sip_pvt->owner channel up. This is because we never
+ hangup the ast_channel after sending the SIP_BYE request. When we
+ receive the response for the SIP_BYE we set need_destroy which we
+ would expect to destroy the dialog on the next do_monitor loop,
+ but this is not the case. The dialog will only be destroyed once
+ the owner is hungup even with the need_destroy flag set. This
+ patch sets the softhangup flag on the ast_channel when a SIP_BYE
+ request is sent as a result of the pending bye flag. ........
+
+2010-08-23 21:32 +0000 [r283318] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_odbc.c, cdr/cdr_adaptive_odbc.c: CDR drivers depend upon
+ res_odbc, not directly on the ODBC libraries
+
+2010-09-15 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.13 released.
+
+ * Incorrect .version and ChangeLog files updated. Re-release
+ of Asterisk 1.6.2.12 with corrections and version
+ number bump.
+
+2010-09-15 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.12 released.
+
+2010-08-23 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.12-rc1 Released.
+
+2010-08-20 16:48 +0000 [r283049-283124] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 283123 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500
+ (Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from
+ https://origsvn.digium.com/svn/asterisk/trunk .......... r278274
+ | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
+ line Reference correct struct member for unlikely event
+ PRI_EVENT_CONFIG_ERR. .......... ................
+
+ * channels/chan_dahdi.c, /: Merged revisions 283048 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20
+ Aug 2010) | 22 lines Q931 - Sending PROGRESS after sending
+ ALERTING is a protocol error The PRI layer in chan_dadhi will
+ check if a PROGRESS message has already been sent, and not allow
+ sending another (although that is technically allowed by the Q931
+ spec), however it does not protect against sending an ALERTING
+ and then sending a PROGRESS message, which is a violation of the
+ specification. Most switches don't seem to care too deeply about
+ this, but some do, and will disconnect the call when receiving
+ this invalid sequence. Protocol specification reference:
+ T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
+ protocol control (network side) point-point (sheet 3 of 8)"
+ (closes issue #17874) Reported by: nic_bellamy Patches:
+ asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
+ nic bellamy (license 299)
+ asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299)
+ asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299) ........
+
+2010-08-19 21:05 +0000 [r282890-282894] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282893 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010)
+ | 11 lines tos_sip option was not being set correctly When
+ tos_sip is used, the tos of the sip socket is only set correctly
+ if the socket binding changes on a reload. If the binding stays
+ the same but the TOS changes, the new tos value would not take
+ into effect. This patch fixes that. (closes issue #17712)
+ Reported by: nickb ........
+
+ * channels/chan_sip.c: fixes sip peer memory leaks in the
+ peer_by_ip table (issue #17798)
+
+2010-08-19 19:44 +0000 [r282859] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Merged revisions 277944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul
+ 2010) | 16 lines Regression with T.38 negotiation Prior to
+ 1.4.26.3 T.38 negotiation worked properly, in the case of the
+ reporter. (issue #16852) Reported by: cfc (closes issue #16705)
+ Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
+ by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
+ samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........
+
+2010-08-19 02:14 +0000 [r282730] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample, /: Merged revisions 282729 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18
+ Aug 2010) | 2 lines Add some documentation about codec
+ negotiation to sip.conf ........
+
+2010-08-18 14:28 +0000 [r282668] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes crash with notifycid (closes issue
+ #17868) Reported by: francesco_r Patches: issue_17868.diff
+ uploaded by dvossel (license 671) Tested by: francesco_r
+
+2010-08-18 07:43 +0000 [r282607] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Don't warn on callerid when completely
+ text, instead of numeric with localdialplan prefixes. (closes
+ issue #16770) Reported by: jamicque Patches:
+ 20100413__issue16770.diff.txt uploaded by tilghman (license 14)
+ 20100811__issue16770.diff.txt uploaded by tilghman (license 14)
+ Tested by: jamicque
+
+2010-08-17 21:35 +0000 [r282576] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes no default transport for temp peer
+ creation in chan_sip (closes issue #17829) Reported by: falves11
+ Patches: issue_17829.rev1.txt uploaded by russell (license 2)
+ issue_17829.diff uploaded by dvossel (license 671) Tested by:
+ falves11
+
+2010-08-16 18:00 +0000 [r282469] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/asterisk.tex, doc/tex/sounds.tex (added): Add information
+ about creating sounds files using the sounds tools publically
+ available so that others can create their own sounds prompts
+ using the same tools we use to generate sounds releases. This
+ allows people creating their own prompts to sound consistent with
+ the prompts available from the open source project. SWP-595
+
+2010-08-16 17:32 +0000 [r282467] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Merged revisions 282430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010)
+ | 16 lines Send a SRCCHANGE indication when we masquerade
+ Masquerading a channel means that the src of the audio is
+ potentially changing, so send a SRCCHANGE so that RTP-based media
+ streams can get a new SSRC generated to reflect the change.
+ Original patch by addix (along with lots of testing--thanks!).
+ (closes issue #17007) Reported by: addix Patches:
+ 1001-reset-SSRC-original-channel.diff uploaded by addix (license
+ 1006) srcchange.diff uploaded by twilson (license 396) Tested by:
+ addix, twilson Review: https://reviewboard.asterisk.org/r/862/
+ ........
+
+2010-08-13 18:54 +0000 [r282235] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: only do magic pickup when notifycid is
+ enabled A new way of doing BLF pickup was introduced into 1.6.2.
+ This feature adds a call-id value into the XML of a SIP_NOTIFY
+ message sent to alert a subscriber that a device is ringing. This
+ option should only be enabled when the new 'notifycid' option is
+ set... but this was not the case. Instead the call-id value was
+ included for every RINGING Notify message, which caused a
+ regression for people who used other methods for call pickup.
+ (closes issue #17633) Reported by: urosh Patches: chan_sip.txt
+ uploaded by urosh (license ) blf_cid_issue.diff uploaded by
+ dvossel (license 671) Tested by: dvossel, urosh, okrief,
+ alecdavis
+
+2010-08-12 22:50 +0000 [r282130] Jason Parker <jparker@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 282129 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) |
+ 1 line Register CLI commands before parsing config, in case there
+ is a config error. ........
+
+2010-08-12 03:01 +0000 [r281912] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 281911 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010)
+ | 20 lines Ensure SSRC is changed when media source is changed to
+ resolve audio delay. This change causes the SSRC to change right
+ before the channels are bridged, which is what used to happen. It
+ seems that fixes were made to attempt limiting SSRC changes,
+ targeted mainly at sending DTMF. DTMF is not affecting the SSRC
+ with this change. There are two other control frames sent in
+ ast_channel_bridge that probably should also be changed to
+ AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change
+ up to the discretion of resolving issue #17007. For reference -
+ old review implementing new control frame SRCCHANGE:
+ https://reviewboard.asterisk.org/r/540 (closes issue #17404)
+ Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler
+ (license 325) Tested by: sdolloff ........
+
+2010-08-11 21:09 +0000 [r281763-281873] Leif Madsen <lmadsen@digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 281819 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11
+ Aug 2010) | 6 lines Add Danish support to say.conf.sample (closes
+ issue #17836) Reported by: RoadKill Patches:
+ say.conf.sample.patch.dk uploaded by RoadKill (license 933)
+ ........
+
+ * configs/say.conf.sample, /: Merged revisions 281762 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11
+ Aug 2010) | 6 lines Allow say.conf to handle large numbers ending
+ with multiple zeros. (closes issue #17833) Reported by: RoadKill
+ Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill
+ (license 933) ........
+
+2010-08-11 15:17 +0000 [r281722] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_readexten.c: Only set status TIMEOUT, if we have no
+ digits. (closes issue #15188) Reported by: jcovert Patches:
+ app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license
+ 551)
+
+2010-08-10 18:04 +0000 [r281567-281574] Russell Bryant <russell@digium.com>
+
+ * main/sched.c: Don't move the time threshold for running scheduled
+ events on every iteration. Instead, only calculate the time
+ threshold each time ast_sched_runq() is called. (closes issue
+ #17742) Reported by: schmidts Patches: sched.c.patch uploaded by
+ schmidts (license 1077)
+
+ * apps/app_dial.c, /: Merged revisions 281566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
+ | 8 lines Reset visible indication after answer. (closes issue
+ #17641) Reported by: klaus3000 Patches:
+ ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
+ klaus3000 (license 65) Tested by: schmidts ........
+
+2010-08-09 20:46 +0000 [r281430] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes SIP peers memory leak We zeroed out
+ the peer's addr before it was removed from the peers_by_ip
+ container. This made it impossible to be removed from the
+ container as the addr is the key used by the container to find
+ the peer. (closes issue #17774) Reported by: kkm Patches:
+ 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
+ 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
+
+2010-08-09 20:07 +0000 [r281391] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 281390 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09
+ Aug 2010) | 13 lines Prevent loss of Caller ID information set on
+ local channel after masquerade. Caller ID set on the channel
+ before a masquerade occurs when using a local channel would cause
+ the information to be lost. The problem was that the information
+ was set on a channel destined to be hung up. The somewhat
+ confusing fix is to detect if any Caller ID has been set on the
+ channel and if so preswap the Caller ID data so that basically
+ the masquerade puts the data back. (closes issue #17138) Reported
+ by: kobaz Review: https://reviewboard.asterisk.org/r/847/
+ ........
+
+2010-08-05 13:11 +0000 [r281051] Russell Bryant <russell@digium.com>
+
+ * main/cdr.c: Cleanup default option value handling for cdr.conf
+ [general]. The default values would differ depending on whether
+ or not cdr.conf exists. That is no longer the case. Apply a
+ default value to the unanswered option. Define all default values
+ as named constants.
+
+2010-08-05 07:40 +0000 [r280983] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280982
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010)
+ | 8 lines Change context lock back to a mutex, because
+ functionality depends upon the lock being recursive. (closes
+ issue #17643) Reported by: zerohalo Patches:
+ 20100726__issue17643.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo ........
+
+2010-08-03 20:52 +0000 [r280671-280812] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_callerid.c, channels/chan_dahdi.c, /: Merged revisions
+ 280811 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010)
+ | 9 lines Prevent DAHDI channels from overriding the callerid,
+ once it's been set by the user. (closes issue #16661) Reported
+ by: jstapleton Patches: 20100414__issue16661.diff.txt uploaded by
+ tilghman (license 14) 20100415__issue16661__1.6.2.diff.txt
+ uploaded by tilghman (license 14) Tested by: jstapleton ........
+
+ * doc/asterisk.sgml, doc/asterisk.8, doc/Makefile (added): Document
+ -B and -W flags and regenerate manpage from sgml
+
+ * apps/app_voicemail.c: Allow the pipe, but also allow the comma
+
+2010-08-02 21:14 +0000 [r280669] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: Change SIP NOTIFY requests to expect a
+ response so authentication will work. This changes the request to
+ be sent with the transmit type XMIT_RELIABLE so that sip_ack
+ doesn't return false and cause the 401 to be ignored in cases
+ where authentication is required. (closes issue #14255) Reported
+ by: zktech
+
+2010-07-29 21:07 +0000 [r280556] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_curl.c: Off-by-one error (closes issue #17590)
+ Reported by: atis Patches: 20100729__issue17590.diff.txt uploaded
+ by tilghman (license 14)
+
+2010-07-29 20:42 +0000 [r280449-280551] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes wrong SRV query for TLS connection
+ (closes issue #17612) Reported by: marcelloceschia Patches:
+ chan-sip_srvQuery.patch uploaded by marcelloceschia (license
+ 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
+ chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia
+ (license 1079) Tested by: marcelloceschia, st, pabelanger
+
+ * main/channel.c, /: Merged revisions 280448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010)
+ | 12 lines fixes issue with translator frame not getting freed A
+ translator frame even if it local storage so the translation path
+ can be freed. This issue prevented g729 licenses from being freed
+ up. (closes issue #17630) Reported by: manvirr Patches:
+ encoder_fix.diff uploaded by dvossel (license 671) Tested by:
+ manvirr, dvossel ........
+
+2010-07-29 16:01 +0000 [r280345] Jean Galarneau <jgalarneau@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 280341 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) |
+ 2 lines Fix a dsp structure leak occuring when a local channel is
+ put into a meetme conference, then masquaraded away. ABE-2422
+ ........
+
+2010-07-29 13:45 +0000 [r280306] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_local.c: Implement support for
+ ast_channel_queryoption on local channels. Currently only
+ AST_OPTION_T38_STATE is supported. ABE-2229 Review:
+ https://reviewboard.asterisk.org/r/813/
+
+2010-07-28 20:02 +0000 [r280231] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile: Work around some silly behavior on BSD. A
+ non-zero exit from a subshell should make the build fail. (closes
+ issue #17621)
+
+2010-07-28 19:57 +0000 [r280229] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Add missing enum value "unknown" to the
+ SS7 called_nai and calling_nai config options.
+
+2010-07-28 19:54 +0000 [r280193-280227] Jason Parker <jparker@digium.com>
+
+ * build_tools/sha1sum-sh (added): Add sha1sum-sh in case there is
+ no util on the system.
+
+ * sounds/Makefile: Remove unnecessary subshells. Attempt to make
+ checksumming work. Also improves readability. (issue #17621)
+ Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/
+
+2010-07-28 16:51 +0000 [r280160] Sean Bright <sean@malleable.com>
+
+ * apps/app_queue.c: Plug a reference leak in app_queue when adding
+ members dynamically. (closes issue #17738) Reported by:
+ bobwienholt Patches: issue17738.patch uploaded by bobwienholt
+ (license 950) Tested by: bobwienholt, seanbright
+
+2010-07-28 13:51 +0000 [r280089] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/live_ast, /: Merged revisions 280088 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28
+ Jul 2010) | 1 line Update help text to be less confusing.
+ ........
+
+2010-07-27 20:54 +0000 [r279946] David Vossel <dvossel@digium.com>
+
+ * main/audiohook.c, main/channel.c, /,
+ include/asterisk/audiohook.h: Merged revisions 279945 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010)
+ | 19 lines remove empty audiohook write list on channel If a
+ channel has an audiohook write list created on it, that list
+ stays on the channel until the channel is destroyed. There is no
+ reason to keep that list on the channel if it becomes empty. If
+ it is empty that just means we are doing needless translating for
+ every ast_read and ast_write. This patch removes the audiohook
+ list from the channel once it is detected to be empty on either a
+ read or write. If a audiohook is added back to the channel after
+ this list is destroyed, the list just gets recreated as if it
+ never existed to begin with. (closes issue #17630) Reported by:
+ manvirr Review: https://reviewboard.asterisk.org/r/799/ ........
+
+2010-07-27 17:54 +0000 [r279849-279883] Jason Parker <jparker@digium.com>
+
+ * makeopts.in, configure, configure.ac: Add SHA1SUM to configure,
+ since we require it for sounds/
+
+ * sounds/Makefile: Remove aptly-named EMPTY and BS vars, since they
+ aren't used anymore.
+
+ * sounds/Makefile: Simply sounds/Makefile some more.
+
+2010-07-27 15:13 +0000 [r279784] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix bad behavior of dynamic_exclude_static
+ option in sip.conf. We were attempting to create a contactdeny
+ rule based on the peer's IP address before the peer's IP address
+ had been set. By moving the processing further down in the
+ function, we can ensure stuff works as we expect for it to.
+ (closes issue #17717) Reported by: mmichelson Patches:
+ 17717.patch uploaded by mmichelson (license 60) Tested by:
+ DennisD
+
+2010-07-26 22:59 +0000 [r279657] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile (added), sounds/Makefile.380 (removed),
+ configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
+ (removed), configure.ac: Really fix sounds Makefile (and make it
+ readableish). There was a rather large syntax error that should
+ have caused ALL versions of GNU make to fail. I don't know how it
+ worked.
+
+2010-07-26 21:18 +0000 [r279609] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Dunno why this worked on my machine, but
+ it works better this way.
+
+2010-07-26 20:25 +0000 [r279597] Gavin Henry <ghenry@suretecsystems.com>
+
+ * res/res_config_ldap.c: Apply all patches in:
+ https://issues.asterisk.org/view.php?id=13573 (closes issue
+ #13573) Reported by: navkumar Patches:
+ res_config_ldap-category.diff uploaded by navkumar (license 580)
+ res_config_ldap.patch uploaded by bencer (license 961)
+ res_config_ldap uploaded by bencer (license 961) Tested by:
+ suretec
+
+2010-07-26 19:15 +0000 [r279561] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/Makefile (removed), configure, sounds/Makefile.380
+ (added), sounds/Makefile.381 (added), configure.ac: Use a special
+ Makefile for noobs who still have GNU Make 3.80. (Closes issue
+ #17716) Reported by: farisraouf
+
+2010-07-26 15:41 +0000 [r279501] Sean Bright <sean@malleable.com>
+
+ * autoconf/ast_ext_lib.m4: Expand the correct value within
+ AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth
+
+2010-07-24 23:58 +0000 [r279347] Bradley Latus <brad.latus@gmail.com>
+
+ * doc/asterisk.8: Minor update to man page
+
+2010-07-23 22:11 +0000 [r279207] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279206 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010)
+ | 7 lines SIP promiscuous redirect could fail to dial the
+ redirect. The ast_channel was created with one variable to
+ ast_request() but the call to ast_call() that initiates the
+ outgoing call was using a different variable. The two variables
+ are not equivalent if the call_forward string included a channel
+ technology specifier. e.g., SIP/200 ........
+
+2010-07-23 18:29 +0000 [r279112] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Backport sip_uri_params_cmp() fix from trunk
+ to 1.6.2.
+
+2010-07-23 18:22 +0000 [r279072-279088] Russell Bryant <russell@digium.com>
+
+ * /: remove old properties
+
+ * /: Add branch-1.4-merged and branch-1.4-blocked properties to
+ 1.6.2 branch.
+
+2010-07-23 17:06 +0000 [r278983-278986] Tilghman Lesher <tlesher@digium.com>
+
+ * autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged
+ revisions 278985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r278985 | tilghman | 2010-07-23 12:05:16 -0500 (Fri, 23 Jul 2010)
+ | 12 lines Merged revisions 278984 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010)
+ | 5 lines Establish a maximum version for openh323 (i.e. not
+ opal), because chan_h323 will fail to load, even if it links.
+ (issue #17679) Reported by: am ........ ................
+
+ * main/asterisk.c, /: Merged revisions 278982 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r278982 | tilghman | 2010-07-23 11:43:34 -0500 (Fri, 23 Jul 2010)
+ | 15 lines Merged revisions 278981 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010)
+ | 8 lines Avoid race with consolethread on shutdown (on parallel
+ processors). (closes issue #17080) Reported by: sybasesql
+ Patches: 20100721__issue17080.diff.txt uploaded by tilghman
+ (license 14) Tested by: sybasesql ........ ................
+
+2010-07-23 15:23 +0000 [r278934] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c: Two more typos to cancell.
+
+2010-07-22 19:52 +0000 [r278709] Jeff Peeler <jpeeler@digium.com>
+
+ * main/xmldoc.c, /: Merged revisions 278708 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r278708 |
+ jpeeler | 2010-07-22 14:45:30 -0500 (Thu, 22 Jul 2010) | 16 lines
+ Add method for finding XML doc files for systems that don't
+ support GLOB_BRACE. In particular, Solaris and perhaps others do
+ not support the above mentioned GNU extension. In this case the
+ paths are simply expanded without the braces and the calls to
+ glob are made separately. Note: I could not explain memory
+ allocation failures that were being reported from within libxml
+ itself when making calls to glob without using GLOB_NOCHECK. This
+ is the only reason why that flag is being used. (closes issue
+ #15402) Reported by: snuffy Patches: bug_xmlpatt-v3.diff uploaded
+ by snuffy (license 35), modified by me ........
+
+2010-07-22 19:32 +0000 [r278703] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: DNID does not get cleard on a new call
+ when using immediate=yes with ISDN signaling. When you are using
+ chan_dahdi ISDN signaling with immediate=yes and a call comes in
+ without a DNID then you get the DNID of a previous call.
+ Chan_dahdi does not touch the DNID field on a new call if it does
+ not have a DNID. Made always copy the DNID from the new call. The
+ patches backport the relevant changes from trunk -r210387.
+ (closes issue #17568) Reported by: wuwu Patches:
+ issue17568_v1.4.patch uploaded by rmudgett (license 664)
+ issue17568_v1.6.2.patch uploaded by rmudgett (license 664)
+
+2010-08-10 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.11 Released.
+
+2010-07-26 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.11-rc2 Released.
+
+2010-07-26 Leif Madsen <lmadsen@digium.com>
+
+ * qwell, asterisk, branch-1.6.2, r279657 ***
+ Really fix sounds Makefile (and make it readableish).
+ There was a rather large syntax error that should have
+ caused ALL versions of GNU make to fail.
+ I don't know how it worked.
+
+ (Closes issue #17716)
+
+2010-07-22 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.11-rc1 Released.
+
+2010-07-22 15:00 +0000 [r278621] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 278620 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r278620 | mmichelson | 2010-07-22 09:58:01 -0500 (Thu, 22 Jul
+ 2010) | 19 lines Merged revisions 278618 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul
+ 2010) | 13 lines Allow PLC to function properly when channels use
+ SLIN for audio. If a channel involved in a bridge was using SLIN
+ audio, then translation paths were not guaranteed to be set up
+ properly since in all likelihood the number of translation steps
+ was only 1. This patch enforces the transcode_via_slin behavior
+ if transcode_via_slin or generic_plc is enabled and one of the
+ formats to make compatible is SLIN. AST-352 ........
+ ................
+
+2010-07-21 18:22 +0000 [r278524] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c: Fix invalid test for rxisoffhook in FXO
+ channels This fixes some cases of no outgoing calls on FXO before
+ an incoming call. Remove an unnecessary testing of an "off-hook"
+ bit from DAHDI for FXO (KS/GS) channels.In some cases the bit
+ would not be initialized properly before the first inbound call
+ and thus prevent an outgoing call. If those tests are actually
+ required by anybody, they should define DAHDI_CHECK_HOOKSTATE in
+ channels/sig_analog.c . (closes issue #14577) Reported by: jkroon
+ Patches: asterisk_chan_dahdi_hookstate_fix.diff uploaded by frawd
+ (license 610) Tested by: frawd Review:
+ https://reviewboard.asterisk.org/r/699/
+
+2010-07-21 16:20 +0000 [r278479] Russell Bryant <russell@digium.com>
+
+ * /, res/res_timing_pthread.c: Merged revisions 278465 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010)
+ | 41 lines Use poll() instead of select() in res_timing_pthread
+ to avoid stack corruption. This code did not properly check
+ FD_SETSIZE to ensure that it did not try to select() on fds that
+ were too large. Switching to poll() removes the limitation on the
+ maximum fd value. (closes issue #15915) Reported by: keiron
+ (closes issue #17187) Reported by: Eddie Edwards (closes issue
+ #16494) Reported by: Hubguru (closes issue #15731) Reported by:
+ flop (closes issue #12917) Reported by: falves11 (closes issue
+ #14920) Reported by: vrban (closes issue #17199) Reported by:
+ aleksey2000 (closes issue #15406) Reported by: kowalma (closes
+ issue #17438) Reported by: dcabot (closes issue #17325) Reported
+ by: glwgoes (closes issue #17118) Reported by: erikje possibly
+ other issues, too ... ........
+
+2010-07-21 15:58 +0000 [r278025-278464] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 278463 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r278463 |
+ tilghman | 2010-07-21 10:56:05 -0500 (Wed, 21 Jul 2010) | 11
+ lines Ensure realtime conferences are treated the same as static
+ conferences when trying to find an empty one. Also, parse the
+ useropts properly, when retrieving from realtime, and add them to
+ the existing flags. (closes issue #17502) Reported by: kenji
+ Patches: 20100720__issue17502.diff.txt uploaded by tilghman
+ (license 14) Tested by: kenji ........
+
+ * apps/app_voicemail.c, /: Merged revisions 278275 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r278275 | tilghman | 2010-07-20 17:40:19 -0500
+ (Tue, 20 Jul 2010) | 14 lines Merged revisions 278261 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010)
+ | 7 lines Delete IMAP messages in reverse order, to ensure
+ reordering after each expunge does not cause deletion of the
+ wrong message. (closes issue #16350) Reported by: noahisaac
+ Patches: 20100623__issue16350.diff.txt uploaded by tilghman
+ (license 14) ........ ................
+
+ * main/autoservice.c, /, main/features.c,
+ include/asterisk/channel.h: Merged revisions 278272 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r278272 | tilghman | 2010-07-20 17:26:23 -0500
+ (Tue, 20 Jul 2010) | 11 lines Merged revisions 278167 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010)
+ | 4 lines Do not queue up DTMF frames while a call is on hold.
+ (Fixes ABE-2110) ........ ................
+
+ * main/manager.c, /: Merged revisions 278024 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r278024 | tilghman | 2010-07-20 11:50:11 -0500 (Tue, 20 Jul 2010)
+ | 14 lines Merged revisions 278023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010)
+ | 7 lines Off-by-one error (closes issue #16506) Reported by:
+ nik600 Patches: 20100629__issue16506.diff.txt uploaded by
+ tilghman (license 14) ........ ................
+
+2010-07-19 21:21 +0000 [r277966] Jean Galarneau <jgalarneau@digium.com>
+
+ * /, main/features.c: Merged revisions 277945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277945 | jeang | 2010-07-19 16:07:08 -0500 (Mon, 19 Jul 2010) |
+ 15 lines Merged revisions 277906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) |
+ 7 lines Avoid trying to pickup a parked extension before the park
+ operation is completed. A crash could occur if the extension is
+ picked up while the parking extension is being announced. Testing
+ pu->notquiteyet while searching for a parked extension resolves
+ this crash. (ABE-2418) ........ ................
+
+2010-07-17 17:52 +0000 [r277774-277777] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c: Merge issues...
+
+ * /, autoconf/ast_func_fork.m4, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 277775 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r277775 | tilghman | 2010-07-17 12:42:32 -0500
+ (Sat, 17 Jul 2010) | 12 lines Merged revisions 277738 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010)
+ | 5 lines Remove uclibc cross-compile triplet, as uclibc has a
+ working fork()... it's only uclinux that does not. (closes issue
+ #17616) Reported by: pprindeville ........ ................
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
+ revisions 277773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277773 | tilghman | 2010-07-17 12:39:28 -0500 (Sat, 17 Jul 2010)
+ | 15 lines Merged revisions 277568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010)
+ | 8 lines Since we split values at the semicolon, we should store
+ values with a semicolon as an encoded value. (closes issue
+ #17369) Reported by: gkservice Patches:
+ 20100625__issue17369.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........ ................
+
+2010-07-16 23:37 +0000 [r277666] Tim Ringenbach <tim.ringenbach@gmail.com>
+
+ * /, main/features.c: Merged revisions 277657 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277657 | tringenbach | 2010-07-16 18:23:15 -0500 (Fri, 16 Jul
+ 2010) | 16 lines Merged revisions 277625 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul
+ 2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on
+ attended transfer. ast_bridge_call() clears
+ AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer,
+ ast_bridge_call() is called for a second bridge on the same
+ channel, and it clears that flag, which still needs to get set
+ for when the original ast_bridge_call() gets control back and
+ checks it. Review: https://reviewboard.asterisk.org/r/741
+ ........ ................
+
+2010-07-16 21:31 +0000 [r277563] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 277530 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277530 | mnicholson | 2010-07-16 16:24:45 -0500 (Fri, 16 Jul
+ 2010) | 11 lines Merged revisions 277497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul
+ 2010) | 4 lines Default to no udptl error correction so that
+ error correction will be disabled in the event that the remote
+ end indicates that they do not support the error correction mode
+ we requested. FAX-128 ........ ................
+
+2010-07-16 21:16 +0000 [r277489] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 277488 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r277488 |
+ jpeeler | 2010-07-16 16:16:08 -0500 (Fri, 16 Jul 2010) | 10 lines
+ Fix reporting estimated queue hold time. Just say the number of
+ seconds (after minutes) rather than doing some incorrect
+ calculation with respect to minutes. (closes issue #17498)
+ Reported by: corruptor Patches: holdesecs_bug.diff uploaded by
+ corruptor (license 253) ........
+
+2010-07-16 20:35 +0000 [r277485] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 277467 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r277467 | rmudgett | 2010-07-16 15:27:51 -0500
+ (Fri, 16 Jul 2010) | 22 lines Merged revisions 277419 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010)
+ | 15 lines priexclusive in chan_dahdi.conf ignored when reloading
+ dahdi module During a reload, the priexclusive and outsignalling
+ parameters are not read in from the config file as intended.
+ Unfortunately, they get set to defaults as a result. This patch
+ makes sure that they do not get set to defaults during a reload.
+ (closes issue #17441) Reported by: mtryfoss Patches:
+ issue17441_v1.4.patch uploaded by rmudgett (license 664)
+ issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
+ issue17441_trunk.patch uploaded by rmudgett (license 664) Tested
+ by: rmudgett ........ ................
+
+2010-07-16 20:30 +0000 [r277478] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql
+ (added), /: Merged revisions 277452 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r277452 |
+ tilghman | 2010-07-16 15:25:11 -0500 (Fri, 16 Jul 2010) | 2 lines
+ Add documentation for MOH realtime fields ........
+
+2010-07-16 19:24 +0000 [r277377] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 277366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r277366 |
+ jpeeler | 2010-07-16 14:22:49 -0500 (Fri, 16 Jul 2010) | 7 lines
+ Add missing handling for ringing state for use with queue empty
+ options. (closes issue #17471) Reported by: jazzy Patches:
+ app_queue.c.diff uploaded by jazzy (license 1056) ........
+
+2010-07-16 18:33 +0000 [r277338] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 277331 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277331 | mnicholson | 2010-07-16 13:31:08 -0500 (Fri, 16 Jul
+ 2010) | 15 lines Merged revisions 277327 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul
+ 2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as
+ extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035)
+ Reported by: francesco_r Patches: pbx.c.patch uploaded by
+ viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar
+ ........ ................
+
+2010-07-16 18:14 +0000 [r277264] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: Merged revisions 277263 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277263 | tilghman | 2010-07-16 13:14:05 -0500 (Fri, 16 Jul 2010)
+ | 12 lines Merged revisions 277261 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010)
+ | 5 lines If variable gotten is not set, will segfault on
+ Solaris. (closes issue #17636) Reported by: bklang ........
+ ................
+
+2010-07-16 17:31 +0000 [r277256] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 277250 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277250 | mnicholson | 2010-07-16 12:30:39 -0500 (Fri, 16 Jul
+ 2010) | 11 lines Merged revisions 277247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul
+ 2010) | 4 lines For pass through DTMF tones, measure the actual
+ duration between the begin and end packets on the wire. If it is
+ detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf
+ emulation. AST-362 ........ ................
+
+2010-07-16 17:18 +0000 [r277188] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, apps/app_amd.c: Merged revisions 277183 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277183 | pabelanger | 2010-07-16 13:13:46 -0400 (Fri, 16 Jul
+ 2010) | 15 lines Merged revisions 277182 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul
+ 2010) | 8 lines Total analysis time error with SIP and silence
+ suppression When using app_amd with SIP providers that have
+ silence suppression on, the iTotalTime count increases
+ exponentially. (closes issue #17656) Reported by: juls ........
+ ................
+
+2010-07-16 15:21 +0000 [r277144] Sean Bright <sean@malleable.com>
+
+ * /, main/translate.c: Merged revisions 277143 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r277143 |
+ seanbright | 2010-07-16 11:20:40 -0400 (Fri, 16 Jul 2010) | 8
+ lines Avoid crashing when installing a duplicate translation path
+ with a lower cost. (closes issue #17092) Reported by: moy
+ Patches: translate.rev254273.patch uploaded by moy (license 222)
+ Tested by: moy ........
+
+2010-07-15 20:42 +0000 [r276572-276809] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 276788 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r276788 |
+ jpeeler | 2010-07-15 15:21:03 -0500 (Thu, 15 Jul 2010) | 6 lines
+ Correct not setting the bindport before attempting to open the
+ socket. Related to changes from 276571, I was accidentally
+ testing with a port set in my configuration causing me to miss
+ this. Also moved the TCP handling as well to occur before
+ build_peer is called. ........
+
+ * main/channel.c, /: Merged revisions 276653 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r276653 | jpeeler | 2010-07-15 08:51:11 -0500 (Thu, 15 Jul 2010)
+ | 9 lines Merged revisions 276652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010)
+ | 2 lines In a perfect world, the frame source would never be
+ NULL. In the meantime, don't crash when it is. ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 276571 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r276571 |
+ jpeeler | 2010-07-14 17:58:24 -0500 (Wed, 14 Jul 2010) | 21 lines
+ Fix MWI notification transmission problems over SIP. MWI updates
+ were not being sent if no messages were found in the event cache.
+ This was corrected since a phone may need to clear its MWI status
+ configured previously from another mailbox. Upon module or sip
+ reload, MWI updates could not be sent due to the sipsock socket
+ not being set early enough in reload_config. The code handling
+ the descriptor assignment and such has simply been moved before
+ the call to build_peer. Issuing a sip reload cleared the IP
+ address of the peer, but skipped checking the database for
+ registration information. The database is now checked both for
+ sip reload and actually reloading the module. If a transmission
+ occurs before the do_monitor thread has started, do not attempt
+ to send a signal to it. (closes issue #17398) Reported by: ip-rob
+ ........
+
+2010-07-14 20:16 +0000 [r276442] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/loader.c, /: Merged revisions 276441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r276441 |
+ kpfleming | 2010-07-14 15:15:48 -0500 (Wed, 14 Jul 2010) | 4
+ lines Don't try to call an embedded module's backup_globals()
+ function until after confirming it exists. ........
+
+2010-07-14 11:52 +0000 [r276269] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 276268 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r276268 | lmadsen | 2010-07-14 06:51:48 -0500
+ (Wed, 14 Jul 2010) | 9 lines Merged revisions 276267 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14
+ Jul 2010) | 1 line Update documentation for voicemail.conf
+ externpass option. ........ ................
+
+2010-07-13 19:11 +0000 [r276125] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 276124 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r276124 | russell | 2010-07-13 14:09:42 -0500 (Tue, 13 Jul 2010)
+ | 9 lines Merged revisions 276123 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010)
+ | 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr
+ instead of peer_cdr in the last commit). ........
+ ................
+
+2010-07-13 19:01 +0000 [r276121] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 276074 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r276074 | jpeeler | 2010-07-13 12:37:40 -0500 (Tue, 13 Jul 2010)
+ | 19 lines Merged revisions 275773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010)
+ | 12 lines Make user removals and traversals thread safe in
+ meetme. Race conditions present in meetme involving the user list
+ where a lack of locking has the potential for a user to be
+ removed during a traversal or as in the case of the reporter
+ after checking if the list is empty could cause a crash. Fixing
+ this was done by convering the userlist to an ao2 container.
+ (closes issue #17390) Reported by: Vince Review:
+ https://reviewboard.asterisk.org/r/746/ ........ ................
+
+2010-07-13 16:55 +0000 [r275996] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 275995 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r275995 | russell | 2010-07-13 11:53:44 -0500 (Tue, 13 Jul 2010)
+ | 21 lines Merged revisions 275994 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010)
+ | 14 lines Access peer->cdr directly instead of through a saved
+ off reference. At this point in the code, it is possible that
+ peer_cdr may be invalid. Specifically, in the blind transfer
+ code, CDRs are swapped between channels. So, peer_cdr is no
+ longer == peer->cdr. The scenario that exposed a crash in this
+ code was a blind transfer that hit the system call limit, causing
+ the transferee channel to get destroyed after the transfer
+ attempt failed. Even if it succeeds and this code doesn't crash,
+ this code was still trying to reset a CDR on a channel that was
+ now owned by a different thread, which is a BadThing(tm).
+ (ABE-2417) ........ ................
+
+2010-07-13 14:49 +0000 [r275911] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/realtime/mysql, contrib/realtime/oracle,
+ contrib/scripts/sip-friends.sql (removed),
+ contrib/realtime/mysql/sipfriends.sql,
+ contrib/realtime/mysql/voicemail.sql, contrib/scripts/vmdb.sql
+ (removed), contrib/realtime/mysql/meetme.sql,
+ contrib/realtime/sqlserver, contrib/scripts/realtime_pgsql.sql
+ (removed), contrib/scripts/iax-friends.sql (removed), /,
+ contrib/realtime/mysql/iaxfriends.sql, contrib/scripts/meetme.sql
+ (removed), contrib/realtime (added), contrib/realtime/postgresql,
+ contrib/realtime/postgresql/realtime.sql: Merged revisions 275910
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r275910 | tilghman | 2010-07-13 09:48:40 -0500
+ (Tue, 13 Jul 2010) | 9 lines Merged revisions 275909 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13
+ Jul 2010) | 2 lines Move SQL scripts into their own
+ database-specific directories. ........ ................
+
+2010-07-12 17:26 +0000 [r275706] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 275682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r275682 | jpeeler | 2010-07-12 12:21:01 -0500 (Mon, 12 Jul 2010)
+ | 18 lines Merged revisions 275665 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010)
+ | 11 lines Change ast_write to not stop generator when called
+ from ast_prod. For SIP channels configured with the
+ progressinband option on, the ringback was being immediately
+ stopped. This problem was due to ast_prod being moved for a
+ deadlock fix in 259858. Prodding the channel after setting up the
+ generator triggered the check in ast_write to stop the generator.
+ The fix here should write the frame the same as was done before
+ the call to ast_prod was moved. (closes issue #17372) Reported
+ by: tech_admin ........ ................
+
+2010-07-12 15:38 +0000 [r275627] Leif Madsen <lmadsen@digium.com>
+
+ * cdr/cdr_pgsql.c, /: Merged revisions 275626 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r275626 |
+ lmadsen | 2010-07-12 10:37:01 -0500 (Mon, 12 Jul 2010) | 11 lines
+ cdr_pgsql does not detect when a table is found. This change adds
+ an ERROR message to let you know when a failure exists to get the
+ columns from the pgsql database, which typically means that the
+ table does not exist. (closes issue #17478) Reported by: kobaz
+ Patches: cdr_pgsql.patch uploaded by kobaz (license 834) Tested
+ by: kobaz, russell, lmadsen ........
+
+2010-07-10 15:11 +0000 [r275311-275469] Russell Bryant <russell@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 245192 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r245192 |
+ mmichelson | 2010-02-06 08:43:03 -0600 (Sat, 06 Feb 2010) | 21
+ lines Remove useless sip options related to hash table size.
+ First off, these options weren't actually doing anything. By the
+ time the options were parsed, the peer and dialog containers had
+ already been allocated with their default values. Second, hash
+ table size is something that doesn't really make sense to change
+ in a config file. If a user is that interested in changing the
+ hashtable size, he can modify the source itself. I have removed
+ the parsing of the hash_peer, hash_user, and hash_dialog options.
+ I have removed the hash_user_size variable altogether since it is
+ not used at all. I also changed hash_peer_size and
+ hash_dialog_size to be constant, and have changed the symbols to
+ be in all caps as constants typically are. I have also removed
+ the entire section in sip.conf.sample regarding configurable
+ hashtable sizes. ........ (merge to 1.6.2 inspired by issue
+ #17553)
+
+ * /: unblock a rev
+
+ * configs/features.conf.sample, /, main/features.c: Merged
+ revisions 275424 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r275424 |
+ russell | 2010-07-09 16:57:21 -0500 (Fri, 09 Jul 2010) | 27 lines
+ Fix some issues related to dynamic feature groups in
+ features.conf. The bridge handling code did not properly consider
+ feature groups when setting parameters that would affect whether
+ or not a native bridge would be attempted. If DYNAMIC_FEATURES
+ only include a feature group, a native bridge would occur that
+ may prevent features from working. Fix a bug in verbose output
+ that would show the key mapping as empty if it was using the
+ default mapping and not a custom mapping in the feature group.
+ Add feature groups to the output of "features show". Adjust the
+ feature execution logic to match that of the logic when executing
+ a feature that was not configured through a feature group. Update
+ features.conf.sample to show that an '=' is still required if
+ using the default key mapping from [applicationmap]. Finally,
+ clean up a little bit of formatting to better coform to coding
+ guidelines while in the area. (closes issue #17589) Reported by:
+ lmadsen Patches: issue_17589.rev4.txt uploaded by russell
+ (license 2) Tested by: russell, lmadsen ........
+
+ * /, main/features.c: Merged revisions 275310 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r275310 |
+ russell | 2010-07-09 14:58:06 -0500 (Fri, 09 Jul 2010) | 2 lines
+ Add missing ao2_iterator_destroy(). ........
+
+2010-07-09 19:23 +0000 [r275260] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_sip.c: Merged revisions 275249 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r275249 | pabelanger | 2010-07-09 15:21:27 -0400 (Fri, 09 Jul
+ 2010) | 15 lines Merged revisions 275241 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul
+ 2010) | 8 lines Fix logging message for stale nonce. (closes
+ issue #17582) Reported by: kenner Patches: chan_sip.c.diff
+ uploaded by kenner (license 1040) Tested by: lmadsen ........
+ ................
+
+2010-07-09 18:24 +0000 [r275191] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/loader.c, /: Merged revisions 275186 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r275186 | mnicholson | 2010-07-09 13:24:03 -0500 (Fri, 09 Jul
+ 2010) | 9 lines Merged revisions 275182 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul
+ 2010) | 2 lines give a better error message when attempting to
+ unload a module that is not loaded ........ ................
+
+2010-07-09 18:11 +0000 [r275148] Russell Bryant <russell@digium.com>
+
+ * configs/features.conf.sample, /: Merged revisions 275147 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r275147 | russell | 2010-07-09 13:11:13 -0500 (Fri, 09
+ Jul 2010) | 2 lines Move parking lot sample config out from the
+ middle of dynamic features sample config. ........
+
+2010-07-09 17:51 +0000 [r275029-275145] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/loader.c, /: Merged revisions 275144 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r275144 | mnicholson | 2010-07-09 12:50:45 -0500 (Fri, 09 Jul
+ 2010) | 9 lines Merged revisions 275143 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul
+ 2010) | 2 lines don't unload modules that returned
+ AST_MODULE_LOAD_DECLINE when they were loaded ........
+ ................
+
+ * apps/app_dial.c, /: Merged revisions 275028 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r275028 | mnicholson | 2010-07-09 11:05:58 -0500 (Fri, 09 Jul
+ 2010) | 15 lines Merged revisions 275027 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul
+ 2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels
+ going into the pbx via the G option in app_dial (closes issue
+ #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff
+ uploaded by mnicholson (license 96) Tested by: jamicque,
+ mnicholson ........ ................
+
+2010-07-09 15:39 +0000 [r275023] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/test.h, /, main/test.c: Merged revisions 275022
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r275022 | russell | 2010-07-09 10:35:53 -0500
+ (Fri, 09 Jul 2010) | 11 lines Merged revisions 275021 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010)
+ | 4 lines Document that a leading and trailing slash is expected
+ for test categories. Also, emit a warning if a test is registered
+ without one of these. ........ ................
+
+2010-07-07 18:34 +0000 [r274627-274640] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 274639 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r274639 | rmudgett | 2010-07-07 13:32:35 -0500 (Wed, 07 Jul 2010)
+ | 1 line Add missing conditional around chan_dahdi
+ mfcr2_skip_category config parameter. ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 274595 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r274595 | rmudgett | 2010-07-07 13:20:00 -0500
+ (Wed, 07 Jul 2010) | 9 lines Merged revisions 274579 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07
+ Jul 2010) | 1 line Close the DAHDI FD on error when processing
+ chan_dahdi toneduration config parameter. ........
+ ................
+
+2010-07-07 06:16 +0000 [r274419] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 274418 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r274418 | tilghman | 2010-07-07 01:15:43 -0500
+ (Wed, 07 Jul 2010) | 15 lines Merged revisions 274417 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 Jul 2010)
+ | 8 lines Correct how 100, 200, 300, etc. is said. Also add the
+ crazy British numbers. (closes issue #16102) Reported by: Delvar
+ Patches: say.conf.fix.patch uploaded by Delvar (license 908)
+ (plus a few additional fixes and simplifications by me) ........
+ ................
+
+2010-07-06 23:06 +0000 [r274360] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample, channels/chan_sip.c: Merged revisions
+ 274284 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r274284 | twilson | 2010-07-06 17:15:27 -0500 (Tue, 06 Jul 2010)
+ | 18 lines Merged revisions 274280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010)
+ | 9 lines Add option to not do a call forward on 482 Loop
+ Detected Asterisk has always set up a forwarded call when
+ receiving a 482 Loop Detected. This prevents handling the call
+ failure by just continuing on in the dialplan. Since this would
+ be a change in behavior, the new option to disable this behavior
+ is forwardloopdetected which defaults to 'yes'. Review:
+ https://reviewboard.asterisk.org/r/764/ ........ ................
+
+2010-07-06 22:30 +0000 [r274347] Jeff Peeler <jpeeler@digium.com>
+
+ * configs/sip.conf.sample, /: Merged revisions 274316 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r274316 | jpeeler | 2010-07-06 17:23:35 -0500
+ (Tue, 06 Jul 2010) | 14 lines Merged revisions 274283 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010)
+ | 7 lines Correct sip.conf.sample comments for prematuremedia
+ option. (closes issue #17513) Reported by: festr Patches: patch
+ uploaded by festr (license 443) ........ ................
+
+2010-07-06 22:10 +0000 [r274282] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 274281 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r274281 | tilghman | 2010-07-06 17:09:23 -0500 (Tue, 06 Jul 2010)
+ | 2 lines Status shows all non-CRC4 lines as "yellow", even if
+ "yellow" was not in the bitfield. ........
+
+2010-07-06 14:33 +0000 [r274168] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, /: Merged revisions 274164 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r274164 | mmichelson | 2010-07-06 09:31:13 -0500 (Tue, 06 Jul
+ 2010) | 22 lines Merged revisions 274157 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul
+ 2010) | 16 lines Fix problem with RFC 2833 DTMF not being
+ accepted. A recent check was added to ensure that we did not
+ erroneously detect duplicate DTMF when we received packets out of
+ order. The problem was that the check did not account for the
+ fact that the seqno of an RTP stream will roll over back to 0
+ after hitting 65535. Now, we have a secondary check that will
+ ensure that the seqno rolling over will not cause us to stop
+ accepting DTMF. (closes issue #17571) Reported by: mdeneen
+ Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license
+ 60) Tested by: richardf, maxochoa, JJCinAZ ........
+ ................
+
+2010-07-05 13:55 +0000 [r273888] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/config.c, /: Merged revisions 273886 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273886 | pabelanger | 2010-07-05 09:53:44 -0400 (Mon, 05 Jul
+ 2010) | 15 lines Merged revisions 273884 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul
+ 2010) | 8 lines Remove extra line breaks from 'core show config
+ mappings' (closes issue #17583) Reported by: pabelanger Patches:
+ issue17583.patch uploaded by pabelanger (license 224) Tested by:
+ lmadsen ........ ................
+
+2010-07-03 02:43 +0000 [r273716-273831] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /, channels/chan_agent.c,
+ channels/chan_h323.c, include/asterisk/lock.h: Merged revisions
+ 273830 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273830 | tilghman | 2010-07-02 21:36:31 -0500 (Fri, 02 Jul 2010)
+ | 16 lines Merged revisions 273793 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010)
+ | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock
+ fails, to help catch potentially large software bugs. (closes
+ issue #17407) Reported by: pdf Patches:
+ 20100527__issue17407.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/751/ ........
+ ................
+
+ * main/autoservice.c, /: Merged revisions 273718 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273718 | tilghman | 2010-07-02 12:10:59 -0500 (Fri, 02 Jul 2010)
+ | 15 lines Merged revisions 273717 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010)
+ | 8 lines Autoservice loop optimization causes a busy loop, when
+ channels are serviced while in hangup. (closes issue #17564)
+ Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt
+ uploaded by tilghman (license 14) Tested by: ramonpeek ........
+ ................
+
+ * apps/app_queue.c, /: Merged revisions 273714 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273714 |
+ tilghman | 2010-07-02 11:57:28 -0500 (Fri, 02 Jul 2010) | 2 lines
+ The switch fallthrough could create some errorneous situations,
+ so best to force directly to the default case. ........
+
+2010-07-02 15:59 +0000 [r273642] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_iax2.c, apps/app_voicemail.c,
+ channels/chan_dahdi.c, channels/chan_sip.c, res/res_agi.c: Fix
+ typos reported by Lintian
+
+2010-07-01 22:17 +0000 [r273571] Russell Bryant <russell@digium.com>
+
+ * main/datastore.c, /: Merged revisions 273566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273566 | russell | 2010-07-01 17:16:23 -0500 (Thu, 01 Jul 2010)
+ | 14 lines Merged revisions 273565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010)
+ | 7 lines Don't return a partially initialized datastore. If
+ memory allocation fails in ast_strdup(), don't return a partially
+ initialized datastore. Bad things may happen. (related to
+ ABE-2415) ........ ................
+
+2010-07-01 20:29 +0000 [r273356-273529] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 273522 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273522 | jpeeler | 2010-07-01 15:28:15 -0500 (Thu, 01 Jul 2010)
+ | 21 lines Merged revisions 273474 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010)
+ | 14 lines Allow admin user to join conference without using
+ admin mode and no user pin. Configuring the conference in
+ meetme.conf like the following: conf => 2345,,6666 did not prompt
+ for pin when used without admin mode. This meant that the
+ conference could not be joined as an admin even if the user knew
+ the correct pin. The original bug report was submitted claiming
+ that the blank user pin should deny entry into the conference. I
+ think a better way to handle this would be with a feature
+ enhancement that used the following syntax: conf => 2345,X,6666 -
+ where X denotes no acceptable pin allowed (closes issue #15704)
+ Reported by: modelnine ........ ................
+
+ * /, apps/app_meetme.c: Merged revisions 273355 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273355 | jpeeler | 2010-07-01 10:12:31 -0500 (Thu, 01 Jul 2010)
+ | 19 lines Merged revisions 273354 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010)
+ | 12 lines Ensure channel placed in meetme in ringing state is
+ properly hung up. An outgoing channel placed in meetme while
+ still ringing which was then hung up would not exit meetme and
+ the channel was not properly destroyed. Specifically checking for
+ this scenario by looking at the appropriate control frames
+ resolves the issue. (closes issue #15871) Reported by: Ivan
+ Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan
+ (license 229) ........ ................
+
+2010-07-01 14:39 +0000 [r273271-273353] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /: Merged revisions 273352 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273352 |
+ mnicholson | 2010-07-01 09:37:37 -0500 (Thu, 01 Jul 2010) | 2
+ lines Fixed whitespace problems ........
+
+ * main/manager.c, /: Merged revisions 273350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273350 |
+ mnicholson | 2010-07-01 09:34:31 -0500 (Thu, 01 Jul 2010) | 2
+ lines Altered my comment about TCP_NODELAY ........
+
+ * main/manager.c, /: Merged revisions 273270 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273270 |
+ mnicholson | 2010-06-30 13:48:21 -0500 (Wed, 30 Jun 2010) | 2
+ lines Set TCP_NODELAY on manager TCP sockets to prevent delays on
+ outgoing packets. This regression was introduced in r48338.
+ AST-359 ........
+
+2010-06-30 17:32 +0000 [r273193-273234] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/rtp.c, /: Merged revisions 273233 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273233 |
+ pabelanger | 2010-06-30 13:28:04 -0400 (Wed, 30 Jun 2010) | 11
+ lines Fix rt(c)p set debug ip taking wrong argument Also clean up
+ some coding errors. (closes issue #17469) Reported by: wdoekes
+ Patches: astsvn-rtp-set-debug-ip.patch uploaded by wdoekes
+ (license 717) Tested by: wdoekes, pabelanger ........
+
+ * /: Revert previous commit; res_rtp_asterisk.c does not exist.
+
+ * /: Unblock revisions 218107 ........ r218107 | mvanbaak |
+ 2009-09-12 15:08:16 +0200 (Sat, 12 Sep 2009) | 8 lines use the
+ actual given ip address for 'rtp set debug ip <foo>' instead of
+ the word 'ip' (closes issue 0015711) Reported by: davidw Patches:
+ 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7)
+ Tested by: davidw ........
+
+2010-06-30 01:07 +0000 [r273056-273145] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: Merged revisions 273144 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273144 |
+ tilghman | 2010-06-29 20:07:02 -0500 (Tue, 29 Jun 2010) | 8 lines
+ Permission checking for the system application is backwards.
+ (closes issue #17550) Reported by: kenner Patches: manager.c.diff
+ uploaded by kenner (license 1040) Tested by: kenner ........
+
+ * main/config.c, /: Merged revisions 273142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273142 |
+ tilghman | 2010-06-29 20:01:14 -0500 (Tue, 29 Jun 2010) | 5 lines
+ Don't attempt to proceed if our internal parser indicates an
+ invalid file. (closes issue #17560) Reported by: Nick_Lewis
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 273078 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273078 | tilghman | 2010-06-29 18:20:40 -0500 (Tue, 29 Jun 2010)
+ | 17 lines Merged revisions 273060 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010)
+ | 10 lines Allow the "useragent" value to be restored into memory
+ from the realtime backend. This value is purely informational. It
+ does not alter configuration at all. (closes issue #16029)
+ Reported by: Guggemand Patches: realtime-useragent.patch uploaded
+ by Guggemand (license 897) Tested by: Guggemand ........
+ ................
+
+ * main/channel.c, /: Merged revisions 273058 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273058 | tilghman | 2010-06-29 17:59:51 -0500 (Tue, 29 Jun 2010)
+ | 11 lines Recorded merge of revisions 273057 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010)
+ | 4 lines _Really_ skip the channel... don't just retry for
+ another 200 cycles. (Closes issue SWP-1652, ABE-2240) ........
+ ................
+
+ * main/pbx.c, /: Merged revisions 273054 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273054 |
+ tilghman | 2010-06-29 17:39:22 -0500 (Tue, 29 Jun 2010) | 11
+ lines Send DialPlanComplete as a response, not as a separate
+ event. Otherwise, it goes to all manager sessions and may exclude
+ the current session, if the Events mask excludes it. (closes
+ issue #17504) Reported by: rrb3942 Patches:
+ showdialplan_patch.diff uploaded by rrb3942 (license 1003) Tested
+ by: rrb3942 ........
+
+2010-06-29 16:43 +0000 [r272972] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 253357 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253357 |
+ russell | 2010-03-18 13:18:43 -0500 (Thu, 18 Mar 2010) | 8 lines
+ Increase CLI command output timeout for asterisk -rx to 60
+ seconds. (closes issue #17049) Reported by: russell Tested by:
+ russell Review: https://reviewboard.asterisk.org/r/573/ ........
+
+2010-07-22 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.10
+
+ * Included a fix for res_timing_pthread per the description below:
+
+ r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010) | 41 lines
+
+ Use poll() instead of select() in res_timing_pthread to avoid stack corruption.
+ This code did not properly check FD_SETSIZE to ensure that it did not try to
+ select() on fds that were too large. Switching to poll() removes the limitation
+ on the maximum fd value.
+
+2010-07-07 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.10-rc2
+
+ * Fix problem with RFC 2833 DTMF not being accepted.
+
+ A recent check was added to ensure that we did not erroneously
+ detect duplicate DTMF when we received packets out of order.
+ The problem was that the check did not account for the fact that
+ the seqno of an RTP stream will roll over back to 0 after hitting
+ 65535. Now, we have a secondary check that will ensure that the
+ seqno rolling over will not cause us to stop accepting DTMF.
+
+ (closes issue 0017571)
+ Reported by: mdeneen
+ Patches:
+ rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
+ Tested by: richardf, maxochoa, JJCinAZ
+
+ * Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx
+ via the G option in app_dial
+
+ (closes issue 0017592)
+ Reported by: jamicque
+ Patches:
+ G-flag-cdr-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: jamicque, mnicholson
+
+2010-06-29 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.10-rc1
+
+2010-06-28 21:51 +0000 [r272924-272927] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 272926 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272926 | tilghman | 2010-06-28 16:50:57 -0500 (Mon, 28 Jun 2010)
+ | 15 lines Merged revisions 272925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010)
+ | 8 lines Don't change ownership/group/permissions on run
+ directory, if it already exists. (closes issue #17076) Reported
+ by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by
+ tilghman (license 14) Tested by: stuarth ........
+ ................
+
+ * main/config.c, /: Merged revisions 272923 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272923 | tilghman | 2010-06-28 16:42:52 -0500 (Mon, 28 Jun 2010)
+ | 19 lines Merged revisions 272921-272922 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010)
+ | 8 lines Change the way that we read include files, to
+ accommodate for changes in GCC 4.4. (closes issue #17472)
+ Reported by: seandarcy Patches: config2.patch uploaded by nivan
+ (license 1066) Tested by: nivan ........ r272922 | tilghman |
+ 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines Also trim
+ trailing blanks on #includes ........ ................
+
+2010-06-28 18:50 +0000 [r272882] Russell Bryant <russell@digium.com>
+
+ * tests/test_astobj2.c (added): Backport applicable parts of
+ test_astobj2 from trunk.
+
+2010-06-28 17:37 +0000 [r272806] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 272805 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272805 | mmichelson | 2010-06-28 12:33:12 -0500 (Mon, 28 Jun
+ 2010) | 11 lines Merged revisions 272804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun
+ 2010) | 5 lines Decode URI in contact header of 302 response.
+ ABE-2352 ........ ................
+
+2010-06-28 15:36 +0000 [r272685-272686] Russell Bryant <russell@digium.com>
+
+ * doc/tex/chan-mobile.tex (removed): remove accidentally added
+ file.
+
+ * doc/tex/cdrdriver.tex, doc/tex/asterisk.tex, /,
+ doc/tex/chan-mobile.tex (added): Merged revisions 272684 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r272684 | russell | 2010-06-28 10:33:32 -0500 (Mon, 28
+ Jun 2010) | 2 lines Use the underscore package so that
+ underscores do not need to be escaped. ........
+
+2010-06-25 20:20 +0000 [r272556-272577] Tilghman Lesher <tlesher@digium.com>
+
+ * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 272568 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r272568 | tilghman | 2010-06-25 15:18:47 -0500
+ (Fri, 25 Jun 2010) | 12 lines Merged revisions 272562 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010)
+ | 5 lines Make the structure of the table specified before match
+ the queries and results. (closes issue #17557) Reported by: cmaj
+ ........ ................
+
+ * sounds/Makefile, /: Merged revisions 272533 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272533 |
+ tilghman | 2010-06-25 14:17:16 -0500 (Fri, 25 Jun 2010) | 2 lines
+ Symlink sounds files, to save disk space, when multiple
+ tarballs/checkouts are on the same system. ........
+
+2010-06-25 18:58 +0000 [r272531] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/_private.h, tests/test_sched.c, main/asterisk.c,
+ include/asterisk/test.h (added), build_tools/cflags-devmode.xml,
+ tests/test_heap.c, tests/test_skel.c, main/Makefile, main/test.c
+ (added): Backport unit test API from trunk. Also, update existing
+ test modules that were already in this branch but had been
+ converted to the unit test API in trunk. Review:
+ https://reviewboard.asterisk.org/r/748/
+
+2010-06-24 22:19 +0000 [r272459] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 272447 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r272447 | rmudgett | 2010-06-24 17:11:26 -0500
+ (Thu, 24 Jun 2010) | 17 lines Merged revisions 272446 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010)
+ | 10 lines ss_thread calls pri_grab without lock during overlap
+ dial Recent changes to chan_dahdi with relation to overlap
+ dialing call pri_grab without first obtaining a lock. (closes
+ issue #17414) Reported by: pdf Patches: bug17414.patch uploaded
+ by jpeeler (license 325) ........ ................
+
+2010-06-23 23:40 +0000 [r272440] Terry Wilson <twilson@digium.com>
+
+ * autoconf/ast_ext_tool_check.m4, /, configure: Merged revisions
+ 272254,272256 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272254 |
+ twilson | 2010-06-23 15:53:48 -0500 (Wed, 23 Jun 2010) | 10 lines
+ Honor the --with-${library}=path for AST_EXT_TOOL_CHECK (closes
+ issue #16991) Reported by: pprindeville Patches:
+ with_netsnmp.patch.txt uploaded by twilson (license 396) Tested
+ by: twilson Review: https://reviewboard.asterisk.org/r/739/
+ ........ r272256 | twilson | 2010-06-23 15:59:17 -0500 (Wed, 23
+ Jun 2010) | 2 lines Update configure when changing autconf m4
+ files... ........
+
+2010-06-23 23:14 +0000 [r272371] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 272370 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r272370 | russell | 2010-06-23 18:09:28 -0500 (Wed, 23 Jun 2010)
+ | 23 lines Resolve some errors produced during module unload of
+ chan_iax2. The external test suite stops Asterisk using the "core
+ stop gracefully" command. The logs from the tests show that there
+ are a number of problems with Asterisk trying to cleanly shut
+ down. This patch addresses the following type of error that comes
+ from chan_iax2: [Jun 22 16:58:11] ERROR[29884]: lock.c:129
+ __ast_pthread_mutex_destroy: chan_iax2.c line 11371
+ (iax2_process_thread_cleanup): Error destroying mutex
+ &thread->lock: Device or resource busy For an example in the
+ context of a build, see:
+ http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary
+ purpose of this patch is to change the thread pool shutdown
+ procedure to be more explicit to ensure that the thread exits
+ from a point where it is not holding a lock. While testing that,
+ I encountered various crashes due to the order of operations in
+ unload_module() being problematic. I reordered some things there,
+ as well. Review: https://reviewboard.asterisk.org/r/736/ ........
+
+2010-06-23 22:37 +0000 [r272369] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 272368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272368 | mnicholson | 2010-06-23 17:36:49 -0500 (Wed, 23 Jun
+ 2010) | 16 lines Merged revisions 272367 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 This version
+ of the patch only adds AgentComplete for attended transfers. It
+ was already present for blind transfers. ........ r272367 |
+ mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8
+ lines Send AgentComplete manager events in the event of blind and
+ attended transfers. (closes issue #16819) Reported by: elbriga
+ Patches: app_queue.diff uploaded by elbriga (license 482)
+ ........ ................
+
+2010-06-23 21:54 +0000 [r272333] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 272332 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r272332 | tilghman | 2010-06-23 16:53:49 -0500 (Wed, 23 Jun 2010)
+ | 8 lines If there is realtime configuration, it does not get
+ re-read on reload unless the config file also changes. (closes
+ issue #16982) Reported by: dmitri Patches: res_musiconhold.patch
+ uploaded by dmitri (license 1001) Tested by: atis ........
+
+2010-06-23 21:15 +0000 [r272263] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * apps/app_meetme.c: Revert previous commit, ast_test_flag64 does
+ not exist in 1.6.2
+
+2010-06-23 21:09 +0000 [r272262] Tilghman Lesher <tlesher@digium.com>
+
+ * res/ael/ael.flex, /, res/ael/ael.tab.c, res/ael/ael.y,
+ res/ael/ael_lex.c: Merged revisions 272260 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272260 |
+ tilghman | 2010-06-23 16:06:40 -0500 (Wed, 23 Jun 2010) | 8 lines
+ Ensure a NULL file while debugging cannot crash AEL. (closes
+ issue #17215) Reported by: vazir Patches:
+ 20100518__issue17215.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........
+
+2010-06-23 21:07 +0000 [r272253-272261] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, apps/app_meetme.c: Merged revisions 272259 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272259 |
+ pabelanger | 2010-06-23 17:06:15 -0400 (Wed, 23 Jun 2010) | 2
+ lines Fix previous merge. ast_test_flag != ast_test_flag64
+ ........
+
+ * /, apps/app_meetme.c: Merged revisions 272257 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272257 | pabelanger | 2010-06-23 17:00:00 -0400 (Wed, 23 Jun
+ 2010) | 19 lines Merged revisions 272255 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun
+ 2010) | 12 lines First caller into a dynamic conference now enter
+ pin once. If MeetMe is configured to use dynamic conference
+ numbers, then the first caller (which creates the conference) had
+ to enter the PIN number twice. (closes issue #15878) Reported by:
+ shawkris Patches: issue15878.patch uploaded by pabelanger
+ (license 224) Tested by: pabelanger ........ ................
+
+ * main/manager.c, /: Merged revisions 272252 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272252 |
+ pabelanger | 2010-06-23 16:35:45 -0400 (Wed, 23 Jun 2010) | 8
+ lines Correct manager variable 'EventList' case. (closes issue
+ #17520) Reported by: kobaz Patches: manager.patch uploaded by
+ kobaz (license 834) Tested by: lmadsen ........
+
+2010-06-23 18:41 +0000 [r272124-272149] Terry Wilson <twilson@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 272146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272146 |
+ twilson | 2010-06-23 13:39:20 -0500 (Wed, 23 Jun 2010) | 2 lines
+ Don't start the sla thread unless we realy need it ........
+
+ * /, apps/app_meetme.c: Merged revisions 272109 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272109 |
+ twilson | 2010-06-23 12:21:40 -0500 (Wed, 23 Jun 2010) | 12 lines
+ Make sure reload updates SLA config Even if there are no stations
+ or trunks defined, we need to start the sla thread to make sure
+ we get the reload event. Also, when doing a reload we need to
+ remove the existing trunks and stations or they end up hanging
+ around. (closes issue #16818) Reported by: mbonin Patches:
+ sla_reload.patch uploaded by twilson (license 396) Tested by:
+ twilson ........
+
+2010-06-22 22:14 +0000 [r272015] David Vossel <dvossel@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 272014 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272014 |
+ dvossel | 2010-06-22 17:11:50 -0500 (Tue, 22 Jun 2010) | 5 lines
+ fixes issue with 'dialplan remove extension blah' segfaulting
+ with tab completion (closes issue #17440) Reported by: kobaz
+ ........
+
+2010-06-22 17:37 +0000 [r271904] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 271903 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271903 | mnicholson | 2010-06-22 12:35:17 -0500 (Tue, 22 Jun
+ 2010) | 15 lines Merged revisions 271902 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun
+ 2010) | 8 lines Decrease the module ref count in sip_hangup when
+ SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the
+ ref count correct. (closes issue #16815) Reported by: rain
+ Patches: chan_sip-unref-fix.diff uploaded by rain (license 327)
+ (modified) Tested by: rain ........ ................
+
+2010-06-22 16:30 +0000 [r271869] Russell Bryant <russell@digium.com>
+
+ * /, res/ais/clm.c, res/ais/evt.c: Merged revisions 271867 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r271867 | russell | 2010-06-22 11:28:03 -0500 (Tue, 22
+ Jun 2010) | 7 lines Resolve some errors that occur on a graceful
+ shutdown. Don't Finalize() if Initialize() did not succeed. This
+ resulted in an error about trying to Finalize() an invalid
+ handle. Also trim some trailing whitespace while in the area.
+ ........
+
+2010-06-22 15:49 +0000 [r271832] David Vossel <dvossel@digium.com>
+
+ * /, main/features.c: Merged revisions 271831 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r271831 |
+ dvossel | 2010-06-22 10:46:22 -0500 (Tue, 22 Jun 2010) | 10 lines
+ fixes attended transfer behavior when both transferee and
+ transferer hung up If both the transferer and transferee of a
+ attended transfer hangup before the new channel picks up, the new
+ channel should be hung up as well as it has no endpoint to talk
+ to. This mirrors the expected behavior used in 1.4. (closes issue
+ #17444) Reported by: corruptor ........
+
+2010-06-22 15:00 +0000 [r271691-271763] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/dundi.conf.sample, /, pbx/pbx_dundi.c: Merged revisions
+ 271762 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271762 | mnicholson | 2010-06-22 09:54:58 -0500 (Tue, 22 Jun
+ 2010) | 15 lines Merged revisions 271761 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun
+ 2010) | 9 lines Allow users to specify a port for dundi peers.
+ (closes issue #17056) Reported by: klaus3000 Patches:
+ dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
+ Tested by: klaus3000 ........ ................
+
+ * include/asterisk/strings.h, configs/sip_notify.conf.sample, /,
+ channels/chan_sip.c: Merged revisions 271690 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271690 | mnicholson | 2010-06-22 07:58:28 -0500 (Tue, 22 Jun
+ 2010) | 18 lines Merged revisions 271689 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun
+ 2010) | 8 lines Modify chan_sip's packet generation api to
+ automatically calculate the Content-Length. This is done by
+ storing packet content in a buffer until it is actually time to
+ send the packet, at which time the size of the packet is
+ calculated. This change was made to ensure that the
+ Content-Length is always correct. (closes issue #17326) Reported
+ by: kenner Tested by: mnicholson, kenner Review:
+ https://reviewboard.asterisk.org/r/693/ ........ This change also
+ adds an ast_str_copy_string() function (similar to
+ ast_copy_string), that copies one ast_str into another, properly
+ handling embedded nulls. ................
+
+2010-06-21 20:48 +0000 [r271555] Jeff Peeler <jpeeler@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 271554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271554 | jpeeler | 2010-06-21 15:46:53 -0500 (Mon, 21 Jun 2010)
+ | 14 lines Merged revisions 271552 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010)
+ | 7 lines Do not use sizeof to calculate size of a heap allocated
+ character array. Change left out from 271399. (closes issue
+ #16053) Reported by: diLLec ........ ................
+
+2010-06-18 21:33 +0000 [r271338-271484] Jeff Peeler <jpeeler@digium.com>
+
+ * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c: Merged
+ revisions 271483 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271483 | jpeeler | 2010-06-18 16:32:09 -0500 (Fri, 18 Jun 2010)
+ | 18 lines Merged revisions 271399 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010)
+ | 11 lines Fix crash when parsing some heavily nested statements
+ in AEL on reload. Due to the recursion used when compiling AEL in
+ gen_prios, all the stack space was being consumed when parsing
+ some AEL that contained nesting 13 levels deep. Changing a few
+ large buffers to be heap allocated fixed the crash, although I
+ did not test how many more levels can now be safely used. (closes
+ issue #16053) Reported by: diLLec Tested by: jpeeler ........
+ ................
+
+ * channels/chan_dahdi.c, /: Merged revisions 269307 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r269307 | rmudgett | 2010-06-09 11:54:38 -0500 (Wed, 09 Jun 2010)
+ | 12 lines Eliminate deadlock potential in dahdi_fixup(). Calling
+ dahdi_indicate() within dahdi_fixup() while the owner pointers
+ are in a potentially inconsistent state is a potentially bad
+ thing in principle. However, calling dahdi_indicate() when the
+ channel private lock is already held can cause a deadlock if the
+ PRI lock is needed because dahdi_indicate() will also get the
+ channel private lock. The pri_grab() function assumes that the
+ channel private lock is held once to avoid deadlock. ........
+
+2010-06-17 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.9 Released.
+
+2010-06-10 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.9-rc3 Released.
+
+2010-06-10 Tilghman Lesher <tlesher@digium.com>
+
+ * Ensure signals are not blocked inside other signal handlers.
+
+ This eliminates the annoying <beep> on the console.
+
+ (closes issue 0017477)
+ Reported by: jvandal
+ Patches:
+ 20100610__issue17477.diff.txt uploaded by tilghman (license 14
+
+2010-06-09 Paul Belanger <paul.belanger@polybeacon.com>
+
+ * Fix Debian init script to not use -c.
+
+ When using the init script as-is currently, it could cause issues on Debian
+ such as high CPU usage. This fix has worked for several people so I'm
+ implementing the change. We now handle color displays properly.
+
+ (closes issue 0016784)
+ Reported by: pabelanger
+ Patches:
+ 20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
+ Tested by: pabelanger, tilghman
+
+2010-06-07 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.9-rc2 Released.
+
+2010-06-07 Tilghman Lesher <tlesher@digium.com>
+
+ * Fix crash in DTMF detection.
+
+ What I did not originally see in my previous commit was that even
+ though the next digit could be detected before the previous was
+ considered ended, the detection of the next digit effectively ends
+ the detection of the previous. Therefore, the length moves in
+ lockstep with the digit, and no separate counter is needed for the
+ length alone.
+
+ (closes issue 0017371)
+ Reported by: alecdavis
+
+ (closes issue 0017474)
+ Reported by: kenner
+
+2010-06-01 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.9-rc1 Released.
+
+2010-06-01 15:20 +0000 [r266598] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 266592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r266592 | tilghman | 2010-06-01 10:18:59 -0500 (Tue, 01 Jun 2010)
+ | 18 lines Merged revisions 266585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010)
+ | 11 lines Prevent CLI prompt from distorting output of lines
+ shorter than the prompt. Uses the VT100 method of clearing the
+ line from the cursor position to the end of the line: Esc-0K
+ (closes issue #17160) Reported by: coolmig Patches:
+ 20100531__issue17160.diff.txt uploaded by tilghman (license 14)
+ Tested by: coolmig ........ ................
+
+2010-05-31 16:07 +0000 [r266570] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * res/res_agi.c: Fix typo in documentation (closes issue #17395)
+ Reported by: pabelanger Patches: res_agi.c.patch uploaded by
+ pabelanger (license 224)
+
+2010-05-30 04:45 +0000 [r266439] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 266438 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r266438 | tilghman | 2010-05-29 23:44:28 -0500
+ (Sat, 29 May 2010) | 9 lines Merged revisions 266437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29
+ May 2010) | 2 lines Reverting patch and reopening issue #16784,
+ as patch breaks color display. ........ ................
+
+2010-05-28 20:55 +0000 [r266338] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 266337 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r266337 |
+ tilghman | 2010-05-28 15:53:04 -0500 (Fri, 28 May 2010) | 1 line
+ Only report swap on platforms which can examine those statistics
+ ........
+
+2010-05-28 17:57 +0000 [r266293] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 266292 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 |
+ dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines
+ fixes crash when creation of UDPTL fails (closes issue #17264)
+ Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff
+ uploaded by dvossel (license 671)
+ issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel
+ (license 671) Tested by: falves11 ........
+
+2010-05-26 21:19 +0000 [r266154] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/extconf.c, main/asterisk.c, /, main/logger.c: Merged
+ revisions 266146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r266146 | tilghman | 2010-05-26 16:17:46 -0500 (Wed, 26 May 2010)
+ | 21 lines Merged revisions 266142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010)
+ | 14 lines Use sigaction for signals which should persist past
+ the initial trigger, not signal. If you call signal() in a
+ Solaris signal handler, instead of just resetting the signal
+ handler, it causes the signal to refire, because the signal is
+ not marked as handled prior to the signal handler being called.
+ This effectively causes Solaris to immediately exceed the
+ threadstack in recursive signal handlers and crash. (closes issue
+ #17000) Reported by: rmcgilvr Patches:
+ 20100526__issue17000.diff.txt uploaded by tilghman (license 14)
+ Tested by: rmcgilvr ........ ................
+
+2010-05-26 18:37 +0000 [r266007] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 266006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 |
+ dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines
+ fixes failed SIP Directed pickup resulting in dead channel
+ (closes issue #17339) Reported by: one47 Patches:
+ sip_magic_pickup2 uploaded by one47 (license 23) Tested by:
+ one47, dvossel ........
+
+2010-05-26 16:31 +0000 [r265895-265959] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c, /: Merged revisions 265923 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r265923 | tilghman | 2010-05-26 11:23:28 -0500
+ (Wed, 26 May 2010) | 14 lines Merged revisions 265910 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010)
+ | 7 lines Not finding rows in the DB does not rise to the level
+ of a warning. (closes issue #17062) Reported by: drookie Patches:
+ 20100525__issue17062.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+ * configs/res_pgsql.conf.sample, res/res_config_pgsql.c, /: Merged
+ revisions 265894 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265894 |
+ tilghman | 2010-05-26 11:14:48 -0500 (Wed, 26 May 2010) | 8 lines
+ Construct socket name, according to the Postgres docs, and
+ document as such. (closes issue #17392) Reported by: dps Patches:
+ 20100525__issue17392.diff.txt uploaded by tilghman (license 14)
+ Tested by: dps ........
+
+2010-05-26 15:52 +0000 [r265890] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Recorded merge of revisions 265842 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed,
+ 26 May 2010) | 9 lines Re-enable "always" option for videosupport
+ option in sip.conf. (closes issue #17016) Reported by: twilson
+ Patches: 17016.patch uploaded by mmichelson (license 60) Tested
+ by: devmod ........
+
+2010-05-26 00:33 +0000 [r265748] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/pbx_lua.c: Merged revisions 265747 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265747 |
+ tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines
+ Use configure to determine the prefixes and include directories
+ properly. This ensures cross-platform compatibility, even among
+ Linux distributions, which don't always put headers in the same
+ place. (closes issue #17391) Reported by: loloski ........
+
+2010-05-25 21:05 +0000 [r265699] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 265698 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 |
+ mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12
+ lines Properly use peer's outboundproxy for outbound REGISTERs.
+ The logic used in transmit_register to get the outboundproxy for
+ a peer was flawed since this value would be overridden shortly
+ afterwards when create_addr was called. In addition, this also
+ fixes some logic used when parsing users.conf so that the peer
+ name is placed in the internally-generated register string so
+ that an outboundproxy set in the Asterisk GUI will be used for
+ outbound REGISTERs. ........
+
+2010-05-25 17:15 +0000 [r265615] David Vossel <dvossel@digium.com>
+
+ * channels/chan_dahdi.c: fixes build issue with zaptel (closes
+ issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff
+ uploaded by dvossel (license 671) Tested by: aragon
+
+2010-05-25 17:06 +0000 [r265612] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 265611 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May
+ 2010) | 15 lines Merged revisions 265610 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May
+ 2010) | 8 lines Don't mark the cdr records of unanswered queue
+ calls with "NOANSWER". This restores the behavior prior to
+ r258670. (closes issue #17334) Reported by: jvandal Patches:
+ queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
+ by: aragon, jvandal ........ ................
+
+2010-05-24 23:52 +0000 [r265521] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/options.h, main/asterisk.c, Makefile,
+ doc/manager_1_1.txt, doc/tex/manager.tex, main/manager.c: Merged
+ revisions 265320,265467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 |
+ twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines
+ Add the FullyBooted AMI event It is possible to connect to the
+ manager interface before all Asterisk modules are loaded. To
+ ensure that an application does not send AMI actions that might
+ require a module that has not yet loaded, the application can
+ listen for the FullyBooted manager event. It will be sent upon
+ connection if all modules have been loaded, or as soon as loading
+ is complete. The event: Event: FullyBooted Privilege: system,all
+ Status: Fully Booted Review:
+ https://reviewboard.asterisk.org/r/639/ ........ r265467 |
+ twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
+ Merge the rest of the FullyBooted patch ........
+
+2010-05-24 22:07 +0000 [r265450-265452] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/h323/ast_h323.cxx: Merged revisions 265451 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r265451 | mmichelson | 2010-05-24 17:05:15 -0500 (Mon,
+ 24 May 2010) | 8 lines Print openh323 log to the Asterisk
+ console. (closes issue #17109) Reported by: under Patches:
+ logstream.diff uploaded by under (license 914) ........
+
+ * /, channels/chan_sip.c: Merged revisions 265449 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 |
+ mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11
+ lines Allow type=user SIP endpoints to be loaded properly from
+ realtime. (closes issue #16021) Reported by: Guggemand Patches:
+ realtime-type-fix.patch uploaded by Guggemand (license 897)
+ (altered by me slightly to avoid ref leaks) Tested by: Guggemand
+ ........
+
+2010-05-24 19:30 +0000 [r265364] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 265273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265273 |
+ dvossel | 2010-05-24 11:10:09 -0500 (Mon, 24 May 2010) | 2 lines
+ fixes segfault when using generic plc ........
+
+2010-05-24 18:30 +0000 [r265318] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 265316 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265316 |
+ tilghman | 2010-05-24 13:19:08 -0500 (Mon, 24 May 2010) | 7 lines
+ On systems with a LOT of RAM, a signed integer sometimes printed
+ negative. (closes issue #16837) Reported by: jlpedrosa Patches:
+ 20100504__issue16837.diff.txt uploaded by tilghman (license 14)
+ ........
+
+2010-05-21 21:57 +0000 [r264998-265172] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix memory hogging behavior of app_queue. From
+ reviewboard: This review request is for the patch on issue 17081.
+ A user reported that he saw increasing numbers of allocations
+ stemming from app_queue.c when he would run the "queue show" CLI
+ command. The user reported that he was using approximately 40
+ realtime queues and as he ran the CLI command more and more, the
+ memory usage would shoot up. As it turns out, there was a memory
+ leak and a separate usage of memory that, while not really a
+ leak, was very irresponsible. Both memory problems can be
+ attributed to the function init_queue(). When the "queue show"
+ command is run, all realtime queues have the init_queue()
+ function called on the in-memory queue. The idea is to place the
+ queue in its default state and then overwrite options specified
+ in the realtime backend as we read them. The first problem, the
+ memory leak, had to do with the fact that the string field for
+ the name of the first periodic announcement file was being
+ re-created every time init_queue was called. This patch corrects
+ the behavior by only calling ast_str_create if the memory has not
+ already been allocated. The other problem is a bit more
+ complicated. The majority of the strings in the call_queue
+ structure were changed to use the ast_string_fields API for 1.6.0
+ and beyond. init_queue resets all string fields on the queue to
+ their default values. Then, later in the realtime queue loading
+ process, these string fields are set to their configured values.
+ For those unfamiliar with string fields, frequent resizing of a
+ string like this is not what the string fields API is designed
+ for. The result of this constant resizing is that as the queue
+ gets loaded, eventually space for the string runs out and so a
+ new memory pool, at twice the size of the previously allocated
+ one, is created for the string fields. The reporter of issue
+ 17081 wrote a script that ran the "queue show" CLI command 2100
+ times. By the end, each of his 40 queues was taking about a
+ megabyte of memory apiece just for their string fields. My fix
+ for this problem is to revert the call_queue structure from using
+ string fields. In my patch here, I have moved the queue back to
+ using fixed-sized buffers. I ran the script provided by the
+ reporter of 17081 and determined that I no longer saw the
+ steadily-increasing memory usage that I had seen before applying
+ the patch. (closes issue #17081) Reported by: wliegel Patches:
+ 17081v2.patch uploaded by mmichelson (license 60) Tested by:
+ wliegel, mmichelson Review:
+ https://reviewboard.asterisk.org/r/651/
+
+ * apps/app_queue.c, include/asterisk/file.h, /: Merged revisions
+ 265090 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May
+ 2010) | 15 lines Merged revisions 265089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May
+ 2010) | 8 lines Don't hang up on a queue caller if the file we
+ attempt to play does not exist. This also fixes a documentation
+ mistake in file.h that made my original attempt to correct this
+ problem not work correctly. (closes issue #17061) Reported by:
+ RoadKill ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 265087 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 |
+ mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7
+ lines Be sure to set the sin_family on the proxy when allocating.
+ (closes issue #17157) Reported by: stuarth ........
+
+ * /, include/asterisk/channel.h: Merged revisions 265000 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r265000 | mmichelson | 2010-05-21 11:54:21 -0500
+ (Fri, 21 May 2010) | 9 lines Merged revisions 264999 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri,
+ 21 May 2010) | 3 lines Fix grammatical error in comment. ........
+ ................
+
+ * main/channel.c, main/autoservice.c, /,
+ include/asterisk/channel.h: Merged revisions 264997 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r264997 | mmichelson | 2010-05-21 11:44:27 -0500
+ (Fri, 21 May 2010) | 38 lines Merged revisions 264996 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May
+ 2010) | 32 lines Allow ast_safe_sleep to defer specific frames
+ until after the sleep has concluded. From reviewboard Background:
+ A Digium customer discovered a somewhat odd bug. The setup is
+ that parties A and B are bridged, and party A places party B on
+ hold. While party B is listening to hold music, he mashes a bunch
+ of DTMF. Party A takes party B off hold while this is happening,
+ but party B continues to hear hold music. I could reproduce this
+ about 1 in 5 times. The issue: When DTMF features are enabled and
+ a user presses keys, the channel that the DTMF is streamed to is
+ placed in an ast_safe_sleep for 100 ms, the duration of the
+ emulated tone. If an AST_CONTROL_UNHOLD frame is read from the
+ channel during the sleep, the frame is dropped. Thus the unhold
+ indication is never made to the channel that was originally
+ placed on hold. The fix: Originally, I discussed with Kevin
+ possible ways of fixing the specific problem reported. However,
+ we determined that the same type of problem could happen in other
+ situations where ast_safe_sleep() is used. Using autoservice as a
+ model, I modified ast_safe_sleep_conditional() to defer specific
+ frame types so they can be re-queued once the sleep has finished.
+ I made a common function for determining if a frame should be
+ deferred so that there are not two identical switch blocks to
+ maintain. Review: https://reviewboard.asterisk.org/r/674/
+ ........ ................
+
+2010-05-20 23:34 +0000 [r264829] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/callerid.c: Merged revisions 264828 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r264828 | rmudgett | 2010-05-20 18:29:43 -0500 (Thu, 20 May 2010)
+ | 13 lines Merged revisions 264820 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010)
+ | 6 lines ast_callerid_parse() had a path that left name
+ uninitialized. Several callers of ast_callerid_parse() do not
+ initialize the name parameter before calling thus there is the
+ potential to use an uninitialized pointer. ........
+ ................
+
+2010-05-20 22:24 +0000 [r264753-264783] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 264779 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264779 |
+ tilghman | 2010-05-20 17:23:32 -0500 (Thu, 20 May 2010) | 8 lines
+ Let ExtensionState resolve dynamic hints. (closes issue #16623)
+ Reported by: tilghman Patches: 20100116__issue16623.diff.txt
+ uploaded by tilghman (license 14) Tested by: lmadsen ........
+
+ * apps/app_stack.c, /: Merged revisions 264752 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264752 |
+ tilghman | 2010-05-20 16:28:53 -0500 (Thu, 20 May 2010) | 7 lines
+ Error message fix. (closes issue #17356) Reported by: kenner
+ Patches: app_stack.c.diff uploaded by kenner (license 1040)
+ ........
+
+2010-05-19 22:10 +0000 [r264453] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/_private.h, include/asterisk/options.h,
+ main/asterisk.c, main/loader.c, main/channel.c, /,
+ channels/chan_sip.c: Merged revisions 264452 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 |
+ mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86
+ lines Fix transcode_via_sln option with SIP calls and improve PLC
+ usage. From reviewboard: The problem here is a bit complex, so
+ try to bear with me... It was noticed by a Digium customer that
+ generic PLC (as configured in codecs.conf) did not appear to
+ actually be having any sort of benefit when packet loss was
+ introduced on an RTP stream. I reproduced this issue myself by
+ streaming a file across an RTP stream and dropping approx. 5% of
+ the RTP packets. I saw no real difference between when PLC was
+ enabled or disabled when using wireshark to analyze the RTP
+ streams. After analyzing what was going on, it became clear that
+ one of the problems faced was that when running my tests, the
+ translation paths were being set up in such a way that PLC could
+ not possibly work as expected. To illustrate, if packets are lost
+ on channel A's read stream, then we expect that PLC will be
+ applied to channel B's write stream. The problem is that generic
+ PLC can only be done when there is a translation path that moves
+ from some codec to SLINEAR. When I would run my tests, I found
+ that every single time, read and write translation paths would be
+ set up on channel A instead of channel B. There appeared to be no
+ real way to predict which channel the translation paths would be
+ set up on. This is where Kevin swooped in to let me know about
+ the transcode_via_sln option in asterisk.conf. It is supposed to
+ work by placing a read translation path on both channels from the
+ channel's rawreadformat to SLINEAR. It also will place a write
+ translation path on both channels from SLINEAR to the channel's
+ rawwriteformat. Using this option allows one to predictably set
+ up translation paths on all channels. There are two problems with
+ this, though. First and foremost, the transcode_via_sln option
+ did not appear to be working properly when I was placing a SIP
+ call between two endpoints which did not share any common
+ formats. Second, even if this option were to work, for PLC to be
+ applied, there had to be a write translation path that would go
+ from some format to SLINEAR. It would not work properly if the
+ starting format of translation was SLINEAR. The one-line change
+ presented in this review request in chan_sip.c fixed the first
+ issue for me. The problem was that in sip_request_call, the
+ jointcapability of the outbound channel was being set to the
+ format passed to sip_request_call. This is nativeformats of the
+ inbound channel. Because of this, when
+ ast_channel_make_compatible was called by app_dial, both channels
+ already had compatibly read and write formats. Thus, no
+ translation path was set up at the time. My change is to set the
+ jointcapability of the sip_pvt created during sip_request_call to
+ the intersection of the inbound channel's nativeformats and the
+ configured peer capability that we determined during the earlier
+ call to create_addr. Doing this got the translation paths set up
+ as expected when using transcode_via_sln. The changes presented
+ in channel.c fixed the second issue for me. First and foremost,
+ when Asterisk is started, we'll read codecs.conf to see the value
+ of the genericplc option. If this option is set, and ast_write is
+ called for a frame with no data, then we will attempt to fill in
+ the missing samples for the frame. The implementation uses a
+ channel datastore for maintaining the PLC state and for creating
+ a buffer to store PLC samples in. Even when we receive a frame
+ with data, we'll call plc_rx so that the PLC state will have
+ knowledge of the previous voice frame, which it can use as a
+ basis for when it comes time to actually do a PLC fill-in. So,
+ reviewers, now I ask for your help. First off, there's the one
+ line change in chan_sip that I have put in. Is it right? By my
+ logic it seems correct, but I'm sure someone can tell me why it
+ is not going to work. This is probably the change I'm least
+ concerned about, though. What concerns me much more is the set of
+ changes in channel.c. First off, am I even doing it right? When I
+ run tests, I can clearly see that when PLC is activated, I see a
+ significant increase in RTP traffic where I would expect it to
+ be. However, in my humble opinion, the audio sounds kind of
+ crappy whenever the PLC fill-in is done. It sounds worse to me
+ than when no PLC is used at all. I need someone to review the
+ logic I have used to be sure that I'm not misusing anything. As
+ far as I can see my pointer arithmetic is correct, and my use of
+ AST_FRIENDLY_OFFSET should be correct as well, but I'm sure
+ someone can point out somewhere where I've done something
+ incorrectly. As I was writing this review request up, I decided
+ to give the code a test run under valgrind, and I find that for
+ some reason, calls to plc_rx are causing some invalid reads.
+ Apparently I'm reading past the end of a buffer somehow. I'll
+ have to dig around a bit to see why that is the case. If it's
+ obvious to someone reviewing, speak up! Finally, I have one other
+ proposal that is not reflected in my code review. Since without
+ transcode_via_sln set, one cannot predict or control where a
+ translation path will be up, it seems to me that the current
+ practice of using PLC only when transcoding to SLINEAR is not
+ useful. I recommend that once it has been determined that the
+ method used in this code review is correct and works as expected,
+ then the code in translate.c that invokes PLC should be removed.
+ Review: https://reviewboard.asterisk.org/r/622/ ........
+
+2010-05-19 20:31 +0000 [r264405] David Vossel <dvossel@digium.com>
+
+ * main/udptl.c, /: Merged revisions 264400 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264400 |
+ dvossel | 2010-05-19 15:30:33 -0500 (Wed, 19 May 2010) | 11 lines
+ fixes infinite loop during udptl.c's decode_open_type When
+ decode_length returns the length there is a check to see if that
+ length is negative, if so the decode loop breaks as this means
+ the limit has been reached. The problem here is that length is an
+ unsigned int, so length can never be negative. This resulted in
+ an infinite loop. (issue #17352) ........
+
+2010-05-19 20:27 +0000 [r264336-264388] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/udptl.c, /: Merged revisions 264379 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264379 |
+ mnicholson | 2010-05-19 15:26:27 -0500 (Wed, 19 May 2010) | 4
+ lines Cast an unsigned int to a signed int when comparing it with
+ 0. (AST-377) ........
+
+ * apps/app_speech_utils.c, /: Merged revisions 264335 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r264335 | mnicholson | 2010-05-19 15:02:57 -0500
+ (Wed, 19 May 2010) | 12 lines Merged revisions 264334 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May
+ 2010) | 5 lines Set quieted flag when receiving a dtmf tone
+ during playback in speechbackground. (closes issue #16966)
+ Reported by: asackheim ........ ................
+
+2010-05-19 19:25 +0000 [r264332] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 264331 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264331 |
+ dvossel | 2010-05-19 14:21:04 -0500 (Wed, 19 May 2010) | 13 lines
+ fixes crash in check_rtp_timeout During deadlock avoidance the
+ sip dialog pvt is locked and unlocked. When this occurs we have
+ no guarantee the pvt's owner is still valid. We were trying to
+ access the pvt's owner after this without checking to see if it
+ still existed first. (closes issue #17271) Reported by: under
+ Patches: check_rtp_timeout.diff uploaded by under (license 914)
+ Tested by: dvossel ........
+
+2010-05-19 17:49 +0000 [r264205-264250] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/options.h, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 264249 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r264249 | tilghman | 2010-05-19 12:48:31 -0500 (Wed, 19 May 2010)
+ | 24 lines Merged revisions 264248 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010)
+ | 17 lines Internal timing is now on by default, if you're using
+ DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is
+ that this version ensures that a timer is always available,
+ whereas in previous versions, it was possible for DAHDI to be
+ loaded, but have no drivers to actually generate timing. If
+ internal_timing was turned on in this circumstance, a complete
+ lack of audio would result. This is the reason why
+ internal_timing was not on by default. However, now that DAHDI
+ ensures the availability of a timer, there is no reason for this
+ setting to be off (and in fact, it solves a great many initial
+ user problems). (closes issue #15932) Reported by: dimas Patches:
+ 20100519__issue15932.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........ ................
+
+ * main/dsp.c, /: Merged revisions 264204 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264204 |
+ tilghman | 2010-05-19 11:42:20 -0500 (Wed, 19 May 2010) | 9 lines
+ Keep track of digit duration, when we're decoding inband to pass
+ DTMF frames. (closes issue #17235) Reported by: frawd Patches:
+ new_dtmf_dsp_len.patch uploaded by frawd (license 610)
+ 20100518__issue17235.diff.txt uploaded by tilghman (license 14)
+ Tested by: frawd ........
+
+2010-05-19 14:47 +0000 [r264115] David Vossel <dvossel@digium.com>
+
+ * main/rtp.c, /: Merged revisions 264114 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264114 |
+ dvossel | 2010-05-19 09:38:02 -0500 (Wed, 19 May 2010) | 13 lines
+ fixes crash during dtmf During the processing of Cisco dtmf the
+ dtmf samples were not being calculated correctly. In an attempt
+ to determine what sample rate was being used, a NULL frame was
+ processed which caused a crash. This patch resolves this. (closes
+ issue #17248) Reported by: falves11 Patches: issue_17248.diff
+ uploaded by dvossel (license 671) ........
+
+2010-05-19 08:15 +0000 [r264032] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, configs/indications.conf.sample: Merged revisions 264031 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r264031 | alecdavis | 2010-05-19 20:09:14 +1200 (Wed, 19
+ May 2010) | 8 lines fix incorrectly typed indications for [nz]
+ stutter and dialrecall (closes issue #17359) Reported by:
+ alecdavis Patches: bug17359.diff.txt uploaded by alecdavis
+ (license 585) ........
+
+2010-05-19 06:41 +0000 [r263951] Tilghman Lesher <tlesher@digium.com>
+
+ * main/dsp.c, /: Merged revisions 263950 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r263950 | tilghman | 2010-05-19 01:41:04 -0500 (Wed, 19 May 2010)
+ | 15 lines Merged revisions 263949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010)
+ | 8 lines Because progress is called multiple times, across
+ several frames, we must persist states when detecting multitone
+ sequences. (closes issue #16749) Reported by: dant Patches:
+ dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by:
+ dant ........ ................
+
+2010-05-18 22:49 +0000 [r263906] David Vossel <dvossel@digium.com>
+
+ * main/strings.c, /: Merged revisions 263904 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r263904 |
+ dvossel | 2010-05-18 17:48:51 -0500 (Tue, 18 May 2010) | 9 lines
+ fixes segfault on logging (closes issue #17331) Reported by:
+ under Patches: utils.diff uploaded by under (license 914)
+ segfault_on_logging.diff uploaded by dvossel (license 671) Tested
+ by: under, dvossel ........
+
+2010-05-18 19:41 +0000 [r263809] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_directory.c, /: Merged revisions 263807 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r263807 | jpeeler | 2010-05-18 14:27:34 -0500
+ (Tue, 18 May 2010) | 17 lines Merged revisions 263769 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010)
+ | 10 lines Modify directory name reading to be interrupted with
+ operator or pound escape. In the case of accidentally entering
+ the wrong first three letters for the reading, users could be
+ very frustrated if the name listing is very long. This allows
+ interrupting the reading by pressing 0 or #. 0 will attempt to
+ execute a configured operator (o) extension and # will exit and
+ proceed in the dialplan. ABE-2200 ........ ................
+
+2010-05-17 22:10 +0000 [r263642] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/devicestate.c: Merged revisions 263640 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r263640 | mmichelson | 2010-05-17 17:08:01 -0500 (Mon, 17 May
+ 2010) | 16 lines Merged revisions 263639 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May
+ 2010) | 10 lines Fix logic error when checking for a devstate
+ provider. When using strsep, if one of the list of specified
+ separators is not found, it is the first parameter to strsep
+ which is now NULL, not the pointer returned by strsep. This issue
+ isn't especially severe in that the worst it is likely to do is
+ waste some cycles when a device with no '/' and no ':' is passed
+ to ast_device_state. ........ ................
+
+2010-05-17 19:37 +0000 [r263587-263590] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 263589 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r263589 | tilghman | 2010-05-17 14:31:15 -0500 (Mon, 17 May 2010)
+ | 9 lines With IMAP backend, messages in INBOX were counted twice
+ for MWI. (closes issue #17135) Reported by: edhorton Patches:
+ 20100513__issue17135.diff.txt uploaded by tilghman (license 14)
+ 17135_2.diff uploaded by ebroad (license 878) Tested by:
+ edhorton, ebroad ........
+
+ * main/app.c: Don't close 'n', just close 'above_n'. (closes issue
+ #17345) Reported by: wdoekes
+
+2010-05-17 14:41 +0000 [r263376-263458] Leif Madsen <lmadsen@digium.com>
+
+ * main/manager.c, /: Merged revisions 263457 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r263457 | lmadsen | 2010-05-17 09:37:35 -0500 (Mon, 17 May 2010)
+ | 19 lines Recorded merge of revisions 263456 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010)
+ | 11 lines Manager cookies are not compatible with RFC2109. The
+ Version field in the cookies we're setting contain quotes around
+ the version number which is not compatible with RFC2109 and
+ breaks some implementations. (closes issue #17231) Reported by:
+ ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by
+ ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by
+ ecarruda (license 559) Tested by: ecarruda, russell ........
+ ................
+
+ * sounds/Makefile, /: Merged revisions 263375 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r263375 | lmadsen | 2010-05-17 09:05:33 -0500 (Mon, 17 May 2010)
+ | 16 lines Merged revisions 263374 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010)
+ | 8 lines Update link to new version of core sounds. The latest
+ version of the core sounds files 1.4.19 now includes the missing
+ queue-minute sound file which is called by app_queue but which
+ has been missing. (closes issue #17123) Reported by: n8ideas
+ ........ ................
+
+2010-05-17 13:03 +0000 [r263293] David Vossel <dvossel@digium.com>
+
+ * CHANGES, channels/chan_dahdi.c: backport of DAHDI dynamic buffer
+ policy dialstring option
+
+2010-05-15 23:41 +0000 [r263202] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, codecs/gsm/Makefile: Merged revisions 252488 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252488 |
+ tilghman | 2010-03-15 12:27:08 -0400 (Mon, 15 Mar 2010) | 9 lines
+ Make the Makefile logic more explicit and move the Snow Leopard
+ logic down to where it's not executed on non-Darwin systems.
+ (closes issue #17028) Reported by: pabelanger Patches:
+ issue17028_20100315.patch uploaded by seanbright (license 71)
+ 20100315__issue17028.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman, pabelanger ........
+
+2010-05-13 22:13 +0000 [r263070] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 263069 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r263069 | rmudgett | 2010-05-13 17:01:36 -0500 (Thu, 13 May 2010)
+ | 1 line Fix inverted logic in cli command: ss7 set debug on/off
+ ........
+
+2010-05-13 15:36 +0000 [r262898] Russell Bryant <russell@digium.com>
+
+ * channels/chan_console.c, /: Merged revisions 262897 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r262897 | russell | 2010-05-13 10:36:12 -0500 (Thu, 13 May 2010)
+ | 4 lines Fix an off by one error that causes a crash. Thanks to
+ Raymond Burke for pointing it out. ........
+
+2010-05-12 20:01 +0000 [r262801] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/loader.c, main/cli.c, /: Merged revisions 262800 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r262800 | pabelanger | 2010-05-12 15:59:16 -0400 (Wed,
+ 12 May 2010) | 8 lines Notify CLI when modules is loaded /
+ unloaded (closes issue #17308) Reported by: pabelanger Patches:
+ cli.modules.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger, russell ........
+
+2010-05-12 19:53 +0000 [r262797-262799] Leif Madsen <lmadsen@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 262798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r262798 |
+ lmadsen | 2010-05-12 14:53:10 -0500 (Wed, 12 May 2010) | 7 lines
+ Revert previous WARNING message removal. Marquis42 suggested a
+ better method of doing what I wanted because I ended up removing
+ the WARNING message for all instances when really I just wanted
+ to remove it for the 'return' keyword, not everything. (issue
+ #17145) ........
+
+ * res/ael/pval.c, /: Merged revisions 262796 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r262796 |
+ lmadsen | 2010-05-12 14:31:42 -0500 (Wed, 12 May 2010) | 4 lines
+ Remove unnecessary WARNING message in ael/pval.c (closes issue
+ #17145) Reported by: okrief ........
+
+2010-05-12 18:03 +0000 [r262746] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 262744 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r262744 | dvossel | 2010-05-12 13:01:20 -0500 (Wed, 12 May 2010)
+ | 17 lines Merged revisions 262662 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010)
+ | 11 lines fixes app_meetme dsp error We attempted to detect
+ silence after translating a frame from signed linear. This caused
+ a flooding of errors. To resolve this the code to detect silence
+ was moved before the translation. (closes issue #17133) Reported
+ by: jsdyer ........ ................
+
+2010-05-12 16:29 +0000 [r262516-262659] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_privacy.c: Merged revisions 262656 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r262656 |
+ tilghman | 2010-05-12 11:23:26 -0500 (Wed, 12 May 2010) | 8 lines
+ Ensure the arguments are initialized. Also miscellaneous CG
+ cleanup. (closes issue #16576) Reported by: uxbod Patches:
+ 20100505__issue16576.diff.txt uploaded by tilghman (license 14)
+ Tested by: uxbod ........
+
+ * /, include/asterisk/causes.h: Merged revisions 262513 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r262513 | tilghman | 2010-05-11 16:25:05 -0500 (Tue, 11
+ May 2010) | 7 lines Move cause 200 to cause 26, as specified in
+ Q.850. Also cleanup the formatting and add a few more that seem
+ like good candidates. (closes issue #16157) Reported by: wimpy
+ ........
+
+2010-05-11 19:58 +0000 [r262425] Jason Parker <jparker@digium.com>
+
+ * /, res/Makefile: Merged revisions 262422 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r262422 | qwell | 2010-05-11 14:57:24 -0500 (Tue, 11 May 2010) |
+ 18 lines Merged revisions 262421 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) |
+ 11 lines Use a less silly method for modifying a flex-generated
+ file. The sed syntax that was used wasn't actually valid, causing
+ some versions to choke. This is the method that is used in 1.6.x+
+ for similar changes. (closes issue #16696) Reported by: bklang
+ Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested
+ by: qwell ........ ................
+
+2010-05-11 19:41 +0000 [r262415-262420] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 262419 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r262419 |
+ pabelanger | 2010-05-11 15:40:37 -0400 (Tue, 11 May 2010) | 8
+ lines Improve logging by displaying line number (closes issue
+ #16303) Reported by: dant Patches: issue16303.patch.v2 uploaded
+ by pabelanger (license 224) Tested by: dant, lmadsen, pabelanger
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 262414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r262414 |
+ pabelanger | 2010-05-11 15:26:17 -0400 (Tue, 11 May 2010) | 8
+ lines Improve logging information for misconfigured contexts
+ (closes issue #17238) Reported by: pprindeville Patches:
+ chan_sip-bug17238.patch uploaded by pprindeville (license 347)
+ Tested by: pprindeville ........
+
+2010-05-11 17:25 +0000 [r262340] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /, Makefile.rules: Merged revisions 262330
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r262330 | tilghman | 2010-05-11 12:23:51 -0500
+ (Tue, 11 May 2010) | 9 lines Merged revisions 262321 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11
+ May 2010) | 2 lines Fix issue #17302 a slightly different way
+ (mad props to Qwell) ........ ................
+
+2010-05-10 19:06 +0000 [r262237-262241] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 262240 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r262240 | dvossel | 2010-05-10 14:06:08 -0500 (Mon, 10
+ May 2010) | 9 lines fixes PickupChan application (closes issue
+ #16863) Reported by: schern Patches: app_directed_pickup.c.patch
+ uploaded by schern (license 995) for_trunk.diff uploaded by
+ cjacobsen (license 1029) Tested by: Graber, cjacobsen, lathama,
+ rickead2000, dvossel ........
+
+ * channels/chan_console.c, /: Merged revisions 262236 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r262236 | dvossel | 2010-05-10 13:36:10 -0500 (Mon, 10 May 2010)
+ | 11 lines fixes crash in chan_console There is a race condition
+ between console_hangup() and start_stream(). It is possible for
+ console_hangup() to be called and then the stream thread to begin
+ after the hangup. To avoid this a check in start_stream() to make
+ sure the pvt-owner still exists while the pvt lock is held is
+ made. If the owner is gone that means the channel hung up and
+ start_stream should be aborted. ........
+
+2010-05-10 16:39 +0000 [r262155] Tilghman Lesher <tlesher@digium.com>
+
+ * /, Makefile.rules: Merged revisions 262152 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r262152 | tilghman | 2010-05-10 11:36:25 -0500 (Mon, 10 May 2010)
+ | 17 lines Merged revisions 262151 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010)
+ | 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes
+ issue #17297) Reported by: jcovert Patches:
+ 20100506__issue17297.diff.txt uploaded by tilghman (license 14)
+ (closes issue #17302) Reported by: jcovert ........
+ ................
+
+2010-05-09 02:17 +0000 [r261916-262105] Tilghman Lesher <tlesher@digium.com>
+
+ * autoconf/ast_ext_lib.m4, autoconf/ast_c_compile_check.m4,
+ autoconf/ast_c_define_check.m4, /, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 262102 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r262102 | tilghman | 2010-05-08 21:14:04 -0500 (Sat, 08
+ May 2010) | 5 lines Cleanup a bit more by getting rid of useless
+ version defines. Also make library detection use passed CFLAGS.
+ (closes issue #17309) Reported by: stuarth ........
+
+ * /, configure, configure.ac: Merged revisions 262048 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r262048 | tilghman | 2010-05-07 21:40:01 -0500 (Fri, 07 May 2010)
+ | 2 lines Use CPPFLAGS to pass PTHREAD_CFLAGS for vpb only
+ ........
+
+ * /, funcs/func_odbc.c: Merged revisions 261917 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261917 |
+ tilghman | 2010-05-07 15:54:35 -0500 (Fri, 07 May 2010) | 8 lines
+ Double free crash (closes issue #17245) Reported by:
+ thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by
+ tilghman (license 14) Tested by: murraytm ........
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 261913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261913 |
+ tilghman | 2010-05-07 15:35:17 -0500 (Fri, 07 May 2010) | 14
+ lines Use the detected pthread building flags in every place,
+ instead of hardcoding -lpthread. We nicely detect the right flags
+ on each system for building Asterisk with pthreads, then ignore
+ it for every other build option that requires us to build with
+ pthreads. This caused some items to return a false negative. Also
+ cleanup some minor naming issues that caused "library library"
+ redundancy in the output. (closes issue #17303) Reported by:
+ stuarth Patches: 20100507__issue17303.diff.txt uploaded by
+ tilghman (license 14) Tested by: stuarth ........
+
+2010-05-07 16:08 +0000 [r261868] Leif Madsen <lmadsen@digium.com>
+
+ * UPGRADE-1.6.txt, /: Merged revisions 261867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261867 |
+ lmadsen | 2010-05-07 11:05:24 -0500 (Fri, 07 May 2010) | 6 lines
+ Update UPGRADE-1.6.txt stating insecure=very has been removed.
+ (closes issue #17282) Reported by: stuarth Tested by: stuarth
+ ........
+
+2010-05-06 20:13 +0000 [r261739] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 261736 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r261736 | jpeeler | 2010-05-06 15:11:53 -0500
+ (Thu, 06 May 2010) | 15 lines Merged revisions 261735 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010)
+ | 8 lines Only allow the operator key to be accepted after
+ leaving a voicemail. Or rather disallow the operator key from
+ being accepted when not offered, such as after finishing a
+ recording from within the mailbox options menu. ABE-2121 SWP-1267
+ ........ ................
+
+2010-05-06 17:08 +0000 [r261612] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 261609 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r261609 | qwell | 2010-05-06 12:06:40 -0500 (Thu, 06 May 2010) |
+ 11 lines Merged revisions 261608 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) |
+ 4 lines Use the versioned MOH tarballs, now that we have them.
+ This makes for more reproducibility. Prompted by a discussion in
+ #asterisk-dev ........ ................
+
+2010-05-06 15:43 +0000 [r261563] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 261560 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261560 |
+ tilghman | 2010-05-06 10:39:10 -0500 (Thu, 06 May 2010) | 8 lines
+ Permit more lines within a SIP body to be parsed. The example
+ given within the related issue showed 120 lines, which was mostly
+ a result of the body being XML. (closes issue #17179) Reported
+ by: khw ........
+
+2010-06-01 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.8 Released.
+
+2010-05-26 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.8-rc2 Released.
+
+2010-05-26 10:56 -0500 [r265891] Matt Nicholson <mnicholson@digium.com>
+
+ * Merged r265610 from 1.4:
+
+ Don't mark the cdr records of unanswered queue calls with "NOANSWER".
+ This restores the behavior prior to r258670.
+
+ (closes issue #17334)
+ Reported by: jvandal
+ Patches:
+ queue-cdr-fixes1.diff uploaded by mnicholson (license 96)
+ Tested by: aragon, jvandal
+
+2010-05-06 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.8-rc1 Released
+
+2010-05-06 14:07 +0000 [r261498-261499] Russell Bryant <russell@digium.com>
+
+ * tests/test_heap.c: Add test case that ensures the heap handles
+ arbitrary removals properly. (issue #17277) Reported by:
+ cappucinoking Patches: test_heap.diff uploaded by cappucinoking
+ (license 1036) Tested by: cappucinoking, russell
+
+ * /, main/heap.c: Merged revisions 261496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261496 |
+ russell | 2010-05-06 08:58:07 -0500 (Thu, 06 May 2010) | 40 lines
+ Fix handling of removing nodes from the middle of a heap. This
+ bug surfaced in 1.6.2 and does not affect code in any other
+ released version of Asterisk. It manifested itself as SIP qualify
+ not happening when it should, causing peers to go unreachable.
+ This was debugged down to scheduler entries sometimes not getting
+ executed when they were supposed to, which was in turn caused by
+ an error in the heap code. The problem only sometimes occurs, and
+ it is due to the logic for removing an entry in the heap from an
+ arbitrary location (not just popping off the top). The scheduler
+ performs this operation frequently when entries are removed
+ before they run (when ast_sched_del() is used). In a normal pop
+ off of the top of the heap, a node is taken off the bottom,
+ placed at the top, and then bubbled down until the max heap
+ property is restored (see max_heapify()). This same logic was
+ used for removing an arbitrary node from the middle of the heap.
+ Unfortunately, that logic is full of fail. This patch fixes that
+ by fully restoring the max heap property when a node is thrown
+ into the middle of the heap. Instead of just pushing it down as
+ appropriate, it first pushes it up as high as it will go, and
+ _then_ pushes it down. Lastly, fix a minor problem in
+ ast_heap_verify(), which is only used for debugging. If a parent
+ and child node have the same value, that is not an error. The
+ only error is if a parent's value is less than its children. A
+ huge thanks goes out to cappucinoking for debugging this down to
+ the scheduler, and then producing an ast_heap test case that
+ demonstrated the breakage. That made it very easy for me to focus
+ on the heap logic and produce a fix. Open source projects are
+ awesome. (closes issue #16936) Reported by: ib2 Tested by:
+ cappucinoking, crjw (closes issue #17277) Reported by:
+ cappucinoking Patches: heap-fix.rev2.diff uploaded by russell
+ (license 2) Tested by: cappucinoking, russell ........
+
+2010-05-06 07:43 +0000 [r261453] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 261451 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (ה', 06 מאי 2010) |
+ 4 lines When failing to configure, don't destroy 'cfg' twice
+ Fixes a crash when some config section had an incorrect channel
+ config. ........
+
+2010-05-05 19:08 +0000 [r261233-261315] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_sip.c: Merged revisions 261314 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May
+ 2010) | 19 lines Merged revisions 261274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May
+ 2010) | 12 lines Registration fix for SIP realtime. Make sure
+ realtime fields are not empty. (closes issue #17266) Reported by:
+ Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
+ Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
+ https://reviewboard.asterisk.org/r/643/ ........ ................
+
+ * apps/app_queue.c, /: Merged revisions 261232 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261232 |
+ pabelanger | 2010-05-05 11:42:07 -0400 (Wed, 05 May 2010) | 10
+ lines 'queue reset stats' erroneously clears wrapuptime
+ configuration. Resets each member's lastcall to 0 now. (closes
+ issue #17262, #16519) Reported by: rain Patches:
+ wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested
+ by: rain ........
+
+2010-05-04 23:55 +0000 [r261098] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 261095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010)
+ | 18 lines Merged revisions 261093-261094 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010)
+ | 7 lines Protect against overflow, when calculating how long to
+ wait for a frame. (closes issue #17128) Reported by: under
+ Patches: d.diff uploaded by under (license 914) ........ r261094
+ | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2
+ lines Add a tiny corner case to the previous commit ........
+ ................
+
+2010-05-04 19:01 +0000 [r260927] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 260924 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260924 | jpeeler | 2010-05-04 13:51:28 -0500
+ (Tue, 04 May 2010) | 18 lines Merged revisions 260923 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010)
+ | 12 lines Voicemail transfer to operator should occur
+ immediately, not after main menu. There were two scenarios in the
+ advanced options that while using the operator=yes and review=yes
+ options, the transfer occurred only after exiting the main menu
+ (after sending a reply or leaving a message for an extension).
+ Now after the audio is processed for the reply or message the
+ transfer occurs immediately as expected. ABE-2107 ABE-2108
+ ........ ................
+
+2010-05-04 16:58 +0000 [r260884] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/sip.conf.sample, include/asterisk/frame.h,
+ main/channel.c, /, channels/chan_sip.c: Merged revisions 254450
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25
+ Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that
+ arrive before a T.38-capable application is executing on a
+ channel. This patch addresses an issue found during working with
+ end-users using res_fax. If an incoming call is answered in the
+ dialplan, or jumps to the 'fax' extension due to reception of a
+ CNG tone (with faxdetect enabled), and then the remote endpoint
+ sends a T.38 re-INVITE, it is possible for the channel's T.38
+ state to be 'T38_STATE_NEGOTIATING' when the application starts
+ up. Unfortunately, even if the application wants to use T.38, it
+ can't respond to the peer's negotiation request, because the
+ AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent
+ originally has been lost, and the application needs the content
+ of that frame to be able to formulate a reply. This patch adds a
+ new 'request' type to AST_CONTROL_T38_PARAMETERS,
+ AST_T38_REQUEST_PARMS. If the application sends this request,
+ chan_sip will re-send the original control frame (with
+ AST_T38_REQUEST_NEGOTIATE as the request type), and the
+ application can respond as normal. If this occurs within the five
+ second timeout in chan_sip, the automatic cancellation of the
+ peer reinvite will be stopped, and the application will 'own' the
+ negotiation process from that point onwards. This also improves
+ the code path in chan_sip to allow sip_indicate(), when called
+ for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero
+ response, which should have been in place before since the
+ control frame *can* fail to be processed properly. It also
+ modifies ast_indicate() to return whatever result the channel
+ driver returned for this control frame, rather than converting
+ all non-zero results into '-1'. Finally, the new request type
+ intentionally returns a positive value, so that an application
+ that sends AST_T38_REQUEST_PARMS can know for certain whether the
+ channel driver accepted it and will be replying with a control
+ frame of its own, or whether it was ignored (if the
+ sip_indicate()/ast_indicate() path had properly supported failure
+ responses before, this would not be necessary). This patch also
+ modifies res_fax to take advantage of the new request. In
+ addition, this patch makes sip_t38_abort() actually lock the
+ private structure before doing its work... bad programmer, no
+ donut. This patch also enhances chan_sip's 'faxdetect' support to
+ allow triggering on T.38 re-INVITEs received as well as CNG tone
+ detection. Review: https://reviewboard.asterisk.org/r/556/
+ ........
+
+2010-05-04 15:51 +0000 [r260746-260805] Jason Parker <jparker@digium.com>
+
+ * /, build_tools/make_build_h: Merged revisions 260802 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260802 | qwell | 2010-05-04 10:49:57 -0500
+ (Tue, 04 May 2010) | 9 lines Merged revisions 260801 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May
+ 2010) | 1 line Fix fallout from removing from configure script.
+ Pointed out by philipp64 on #asterisk-dev ........
+ ................
+
+ * /: Fix merge props
+
+2010-05-03 17:42 +0000 [r260743] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * Makefile, /: Merged revisions 260661-260662 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May
+ 2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend
+ libdir when executing mkpkgconfig allowing non-root installs to
+ work. (closes issue #17268) Reported by: pabelanger Patches:
+ issue17268.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41
+ -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/
+ part. Thanks Qwell. ........
+
+2010-05-03 14:59 +0000 [r260571] Leif Madsen <lmadsen@digium.com>
+
+ * doc/HOWTO_collect_debug_information.txt: Merged revisions 260570
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260570 | lmadsen | 2010-05-03 09:58:23 -0500
+ (Mon, 03 May 2010) | 9 lines Merged revisions 260569 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03
+ May 2010) | 1 line Minor typo pointed out by pabelanger on IRC.
+ ........ ................
+
+2010-04-30 22:48 +0000 [r260441] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 260437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500
+ (Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010)
+ | 11 lines Ensure channel state is not incorrectly set in the
+ case of a very early answer. The needringing bit was being read
+ in dahdi_read after answering thereby setting the state to
+ ringing from up. This clears needringing upon answering so that
+ is no longer possible. (closes issue #17067) Reported by: tzafrir
+ Patches: needringing.diff uploaded by tzafrir (license 46)
+ ........ ................
+
+2010-04-30 20:22 +0000 [r260373] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 260346 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260346 | mmichelson | 2010-04-30 15:11:02 -0500
+ (Fri, 30 Apr 2010) | 24 lines Merged revisions 260345 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr
+ 2010) | 18 lines Fix potential crash from race condition due to
+ accessing channel data without the channel locked. In
+ res_musiconhold.c, there are several places where a channel's
+ stream's existence is checked prior to calling ast_closestream on
+ it. The issue here is that in several cases, the channel was not
+ locked while checking the stream. The result was that if two
+ threads checked the state of the channel's stream at
+ approximately the same time, then there could be a situation
+ where both threads attempt to call ast_closestream on the
+ channel's stream. The result here is that the refcount for the
+ stream would go below 0, resulting in a crash. I have added
+ proper channel locking to res_musiconhold.c to ensure that we do
+ not try to check chan->stream without the channel locked. A
+ Digium customer has been using this patch for several weeks and
+ has not had any crashes since applying the patch. ABE-2147
+ ........ ................
+
+2010-04-30 06:22 +0000 [r260281-260303] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/app.c: Merged revisions 260292 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r260292 |
+ tilghman | 2010-04-30 01:19:35 -0500 (Fri, 30 Apr 2010) | 13
+ lines Don't allow file descriptors to go above 64k, when we're
+ closing them in a fork(2). This saves time, when, even though the
+ system allows the process limit to be that high, the practical
+ limit is much lower. (closes issue #17223) Reported by:
+ dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by
+ tilghman (license 14) Tested by: dbackeberg ........
+
+ * configs/extensions.conf.sample, /: Merged revisions 260280 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r260280 | tilghman | 2010-04-30 00:23:56 -0500 (Fri, 30
+ Apr 2010) | 7 lines Logic fixups for a sample FREENUM dialplan
+ context. (closes issue #17263) Reported by: pprindeville Patches:
+ freenum-dialplan.patch#3 uploaded by pprindeville (license 347)
+ ........
+
+2010-04-29 23:13 +0000 [r260234] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 260231 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500
+ (Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010)
+ | 26 lines DTMF CallerID detection problems. The code handling
+ DTMF CallerID drops digits on long CallerID numbers and may
+ timeout waiting for the first ring with shorter numbers. The DTMF
+ emulation mode was not turned off when processing DTMF CallerID.
+ When the emulation code gets behind in processing the DTMF digits
+ it can skip a digit. For shorter numbers, the timeout may have
+ been too short. I increased it from 2 seconds to 4 seconds. Four
+ seconds is a typical time between rings for many countries.
+ (closes issue #16460) Reported by: sum Patches: issue16460.patch
+ uploaded by rmudgett (license 664) issue16460_v1.6.2.patch
+ uploaded by rmudgett (license 664) Tested by: sum, rmudgett
+ Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA
+ AST-334 JIRA SWP-901 ........ ................
+
+2010-04-29 18:18 +0000 [r260156] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 260148 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29
+ Apr 2010) | 2 lines Pattern match fail. ........
+
+2010-04-29 15:35 +0000 [r260051] David Vossel <dvossel@digium.com>
+
+ * main/audiohook.c, /, include/asterisk/audiohook.h: Merged
+ revisions 260050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010)
+ | 21 lines Merged revisions 260049 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010)
+ | 14 lines Fixes crash in audiohook_write_list The middle_frame
+ in the audiohook_write_list function was being freed if a
+ audiohook manipulator returned a failure. This is incorrect
+ logic. This patch resolves this and adds detailed descriptions of
+ how this function should work and why manipulator failures must
+ be ignored. (closes issue #17052) Reported by: dvossel Tested by:
+ dvossel (closes issue #16196) Reported by: atis Review:
+ https://reviewboard.asterisk.org/r/623/ ........ ................
+
+2010-04-28 22:36 +0000 [r259959] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 259957 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 |
+ mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11
+ lines Don't override peer context with domain context. (closes
+ issue #17040) Reported by: pprindeville Patches:
+ asterisk-1.6-bugid17040.patch uploaded by pprindeville (license
+ 347) Tested by: pprindeville Review:
+ https://reviewboard.asterisk.org/r/565/ ........
+
+2010-04-28 21:26 +0000 [r259899] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, channels/chan_local.c, /: Merged revisions 259870
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259870 | dvossel | 2010-04-28 16:20:03 -0500
+ (Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010)
+ | 33 lines resolves deadlocks in chan_local Issue_1. In the
+ local_hangup() 3 locks must be held at the same time... pvt,
+ pvt->chan, and pvt->owner. Proper deadlock avoidance is done when
+ the channel to hangup is the outbound chan_local channel, but
+ when it is not the outbound channel we have an issue... We
+ attempt to do deadlock avoidance only on the tech pvt, when both
+ the tech pvt and the pvt->owner are locked coming into that loop.
+ By never giving up the pvt->owner channel deadlock avoidance is
+ not entirely possible. This patch resolves that by doing deadlock
+ avoidance on both the pvt->owner and the pvt when trying to get
+ the pvt->chan lock. Issue_2. ast_prod() is used in
+ ast_activate_generator() to queue a frame on the channel and make
+ the channel's read function get called. This function is used in
+ ast_activate_generator() while the channel is locked, which
+ mean's the channel will have a lock both from the generator code
+ and the frame_queue code by the time it gets to chan_local.c's
+ local_queue_frame code... local_queue_frame contains some of the
+ same crazy deadlock avoidance that local_hangup requires, and
+ this recursive lock prevents that deadlock avoidance from
+ happening correctly. This patch removes ast_prod() from the
+ channel lock so only one lock is held during the
+ local_queue_frame function. (closes issue #17185) Reported by:
+ schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
+ (license 671) issue_17185_v2.diff uploaded by dvossel (license
+ 671) Tested by: schmoozecom, GameGamer43 Review:
+ https://reviewboard.asterisk.org/r/631/ ........ ................
+
+2010-04-28 21:09 +0000 [r259854] Leif Madsen <lmadsen@digium.com>
+
+ * config.guess: Merged revisions 259853 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010)
+ | 14 lines Merged revisions 259852 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010)
+ | 6 lines Update config.guess. Updating config.guess because
+ after installing Ubuntu Server 9.10 and running all the update
+ scripts, running ./configure would not continue because it was
+ unable to determine what kind of system I had. After updating
+ config.guess things started working again. ........
+ ................
+
+2010-04-28 20:34 +0000 [r259781-259851] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 259848 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259848 | qwell | 2010-04-28 15:32:14 -0500
+ (Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr
+ 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so
+ systems without install can use install-sh from our source dir.
+ ........ ................
+
+ * makeopts.in, /: Merged revisions 259837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) |
+ 9 lines Merged revisions 259833 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) |
+ 1 line Missed this when removing $ID ........ ................
+
+ * Makefile, /, configure, configure.ac: Merged revisions 259760 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259760 | qwell | 2010-04-28 14:19:54 -0500
+ (Wed, 28 Apr 2010) | 14 lines Merged revisions 259748 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) |
+ 7 lines Remove usage of `id` since it isn't useful and was
+ causing breakge. Solaris `id` doesn't support the -u argument.
+ Instead of figuring out how to fix this to work on Solaris, I
+ decided to check why it was necessary and where else it was used.
+ It was only used in one place, and it hasn't been needed for a
+ very long time (I question whether it was ever needed). ........
+ ................
+
+2010-04-28 17:19 +0000 [r259681] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 259672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259672 | jpeeler | 2010-04-28 12:18:43 -0500
+ (Wed, 28 Apr 2010) | 11 lines Merged revisions 259664 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010)
+ | 4 lines Do not play goodbye prompt after timeout of message
+ review. ABE-2124 ........ ................
+
+2010-04-27 22:46 +0000 [r259616] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 259538 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500
+ (Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010)
+ | 11 lines DAHDI "WARNING" message is confusing and vague
+ "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
+ failed: Success" Changed the warning to "Failed to decode
+ CallerID on channel 'name'". The message before it is likely more
+ specific about why the CallerID decode failed. SWP-501 AST-283
+ ........ ................
+
+2010-04-27 21:50 +0000 [r259528] Leif Madsen <lmadsen@digium.com>
+
+ * sounds/Makefile: Merged revisions 259527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010)
+ | 23 lines Merged revisions 259526 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010)
+ | 15 lines Update sounds files. * Add additional sounds prompts
+ for say_enumeration * Update the English conference sounds
+ prompts so they are better quality and all sound more consistent
+ * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files
+ to include all present sound files Both core (en, fr, es) and
+ extra (en, fr) sounds files have been updated. (closes issue
+ #16200) Reported by: murf (closes issue #17137) Reported by:
+ lmadsen ........ ................
+
+2010-04-27 21:25 +0000 [r259356-259486] Jason Parker <jparker@digium.com>
+
+ * main/editline/configure.in, /, main/editline/configure,
+ main/editline/Makefile.in: Merged revisions 259439 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) |
+ 5 lines Add gar to the check for AR for those silly OSes
+ (Solaris) that don't have ar. autoconf2.13 couldn't handle
+ AC_PROG_GREP, so I removed it. This is fine, since we don't need
+ to use anything that the configure script doesn't. ........
+
+ * /: Unblock revision 259439.
+
+ * /, configure, configure.ac: Merged revisions 259353 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259353 | qwell | 2010-04-27 14:31:55 -0500
+ (Tue, 27 Apr 2010) | 12 lines Merged revisions 259352 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) |
+ 5 lines Support the silly OSes that don't have ar and strip.
+ Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't
+ specified, and AC_PATH_TOOLS doesn't exist, we'll just switch to
+ AC_CHECK_TOOLS. ........ ................
+
+2010-04-27 19:03 +0000 [r259310] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
+ revisions 259307 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010)
+ | 21 lines Merged revisions 259270 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010)
+ | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue
+ #7321 implements a new chan_dahdi configuration option. However,
+ a change mentioned in the issue was never implemented. This is
+ the change that will allow the feature to work. I added a note to
+ chan_dahdi.conf.sample about the feature. (closes issue #17143)
+ Reported by: djensen99 Patches: diff.txt uploaded by djensen99
+ (license NA) (One line change) Tested by: djensen99 ........
+ ................
+
+2010-04-26 21:48 +0000 [r259103-259109] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 259105 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr
+ 2010) | 9 lines Merged revisions 259104 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr
+ 2010) | 3 lines Let compilation succeed warning-free when
+ DONT_OPTIMIZE is turned off. ........ ................
+
+ * main/channel.c, /: Merged revisions 259023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr
+ 2010) | 19 lines Merged revisions 259018 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr
+ 2010) | 13 lines Prevent Newchannel manager events for dummy
+ channels. No Newchannel manager event will be fired for channels
+ that are allocated to not match a registered technology type.
+ Thus bogus channels allocated solely for variable substitution or
+ CDR operations do not result in a Newchannel event. (closes issue
+ #16957) Reported by: atis Review:
+ https://reviewboard.asterisk.org/r/601 ........ ................
+
+2010-04-26 16:00 +0000 [r258935] Leif Madsen <lmadsen@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 258934 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r258934 |
+ lmadsen | 2010-04-26 10:59:34 -0500 (Mon, 26 Apr 2010) | 7 lines
+ Small error in the T.140 RTP port verbose log. (closes issue
+ #16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff
+ uploaded by frawd (license 610) Tested by: russell ........
+
+2010-04-25 18:14 +0000 [r258779] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 258776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010)
+ | 13 lines Merged revisions 258775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010)
+ | 6 lines When StopMonitor is called, ensure that it will not be
+ restarted by a channel event. (closes issue #16590) Reported by:
+ kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm
+ (license 888) ........ ................
+
+2010-04-22 22:15 +0000 [r258676] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, main/channel.c, /, main/features.c: Merged revisions
+ 258671,258675 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr
+ 2010) | 32 lines Merged revisions 193391,258670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May
+ 2009) | 8 lines Set the proper disposition on originated calls.
+ (closes issue #14167) Reported by: jpt Patches:
+ call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
+ Tested by: dlotina, rmartinez, mnicholson ........ r258670 |
+ mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11
+ lines Fix broken CDR behavior. This change allows a CDR record
+ previously marked with disposition ANSWERED to be set as BUSY or
+ NO ANSWER. Additionally this change partially reverts r235635 and
+ does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated
+ from ast_call(). To preserve proper CDR behavior, the
+ AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in
+ ast_bridge_call(). (closes issue #16797) Reported by:
+ VarnishedOtter Tested by: mnicholson ........ (closes issue
+ #16222) Reported by: telles Tested by: mnicholson
+ ................ r258675 | mnicholson | 2010-04-22 17:11:23 -0500
+ (Thu, 22 Apr 2010) | 2 lines Fix previous commit.
+ ................
+
+2010-04-22 21:58 +0000 [r258516-258672] Russell Bryant <russell@digium.com>
+
+ * /, main/event.c: Merged revisions 258632 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk For 1.6.2, only
+ merge the bug fixes, not the unit test. ........ r258632 |
+ russell | 2010-04-22 16:06:53 -0500 (Thu, 22 Apr 2010) | 22 lines
+ Add ast_event subscription unit test and fix some ast_event API
+ bugs. This patch introduces another test in test_event.c that
+ exercises most of the subscription related ast_event API calls. I
+ made some minor additions to the existing event allocation test
+ to increase API coverage by the test code. Finally, I made a list
+ in a comment of API calls not yet touched by the test module as a
+ to-do list for future test development. During the development of
+ this test code, I discovered a number of bugs in the event API.
+ 1) subscriptions to AST_EVENT_ALL were not handled appropriately
+ in a couple of different places. The API allows a subscription to
+ all event types, but with IE parameters, just as if it was a
+ subscription to a specific event type. However, the parameters
+ were being ignored. This affected ast_event_check_subscriber()
+ and event distribution to subscribers. 2) Some of the logic in
+ ast_event_check_subscriber() for checking subscriptions against
+ query parameters was wrong. Review:
+ https://reviewboard.asterisk.org/r/617/ ........
+
+ * /, doc/tex/channelvariables.tex: Merged revisions 258515 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r258515 | russell | 2010-04-22 12:36:34 -0500 (Thu, 22
+ Apr 2010) | 2 lines Add MEETMEBOOKID from r256019. ........
+
+2010-04-21 22:11 +0000 [r258436] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 258433 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r258433 | jpeeler | 2010-04-21 16:56:09 -0500
+ (Wed, 21 Apr 2010) | 15 lines Merged revisions 258432 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010)
+ | 8 lines Fix looping forever when no input received in certain
+ voicemail menu scenarios. Specifically, prompting for an
+ extension (when leaving or forwarding a message) or when
+ prompting for a digit (when saving a message or changing
+ folders). ABE-2122 SWP-1268 ........ ................
+
+2010-04-21 19:44 +0000 [r258384-258386] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/asterisk.tex: Remove missed line in previous merge.
+ (issue #17220)
+
+ * configure: Forgot to merge the updated configure script. (issue
+ #17220)
+
+ * doc/tex/localchannel.tex, doc/tex/enum.tex, makeopts.in,
+ doc/tex/asterisk.tex, Makefile, /, doc/tex/Makefile,
+ configure.ac, doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex,
+ build_tools/prep_tarball: Merged revisions 258351 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r258351 | lmadsen | 2010-04-21 14:18:35 -0500 (Wed, 21 Apr 2010)
+ | 20 lines Add ability to generate ASCII documentation from the
+ TeX files. These changes add the ability to run 'make
+ asterisk.txt' just like the existing 'make asterisk.pdf' commands
+ to generate a text document from the TeX files we have in the
+ doc/tex/ directory. I've also updated a few of the .tex files
+ because they weren't properly escaping certain characters so they
+ would show up as Unicode characters (like [U+021C]). Made changes
+ to the configure scripts so it would detect the catdvi program
+ which is required to convert the .dvi file generated by latex.
+ I've also added a few lines to the build_tools/prep_tarball
+ script so that the text documentation gets generated and added to
+ future tarballs of Asterisk releases. (closes issue #17220)
+ Reported by: lmadsen Patches: asterisk.txt.patch uploaded by
+ lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger
+ (license 224) Tested by: lmadsen, pabelanger ........
+
+2010-04-21 18:19 +0000 [r258314] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 258305 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 |
+ dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines
+ fixes issue with double "sip:" in header field This is a clear
+ mistake in logic. Future discussions about how to avoid having to
+ handle uri's like this should take place in the future, but this
+ fix needs to go in for now. (closes issue #15847) Reported by:
+ ebroad Patches: doublesip.patch uploaded by ebroad (license 878)
+ ........
+
+2010-04-20 19:03 +0000 [r258148-258150] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/cli_aliases.conf.sample: Merged revisions 258149 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r258149 | lmadsen | 2010-04-20 14:02:49 -0500 (Tue, 20
+ Apr 2010) | 1 line Add 'soft hangup' alias per Steve Johnson on
+ asterisk-users. ........
+
+ * configs/extensions.conf.sample, /: Merged revisions 258147 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r258147 | lmadsen | 2010-04-20 13:38:39 -0500 (Tue, 20
+ Apr 2010) | 8 lines Add example dialplan for dialing ISN numbers
+ (http://www.freenum.org). Minor tweaks and documentation added by
+ me. (closes issue #17058) Reported by: pprindeville Patches:
+ freenum.patch#5 uploaded by pprindeville (license 347) Tested by:
+ lmadsen ........
+
+2010-04-20 18:04 +0000 [r258108] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 258065 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r258065 | jpeeler | 2010-04-20 12:06:19 -0500
+ (Tue, 20 Apr 2010) | 17 lines Merged revisions 258029 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010)
+ | 11 lines Play correct prompt when voicemail store failure
+ occurs after attempted forward. If a user's mailbox was full and
+ a message was attempted to be forwarded to said box, warnings on
+ the console would indicate failure. However, the played prompt
+ was that of success (vm-msgsaved). Now storage failure is taken
+ into account and the correct prompt (vm-mailboxfull) is played
+ when appropriate. ABE-2123 SWP-1262 ........ ................
+
+2010-04-20 18:02 +0000 [r258107] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/sip-friends.sql, /: Merged revisions 258106 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r258106 | lmadsen | 2010-04-20 13:01:28 -0500 (Tue, 20
+ Apr 2010) | 7 lines Add missing 'useragent' field to
+ sip-friends.sql file. (closes issue #17171) Reported by: thehar
+ Patches: sip-friends.patch uploaded by thehar (license 831)
+ Tested by: pabelanger, thehar ........
+
+2010-04-19 21:58 +0000 [r257948-257950] Jason Parker <jparker@digium.com>
+
+ * main/indications.c, /: Merged revisions 257949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r257949 |
+ qwell | 2010-04-19 16:57:56 -0500 (Mon, 19 Apr 2010) | 1 line
+ Change log message to match severity. ........
+
+ * main/indications.c, /: Merged revisions 257947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r257947 |
+ qwell | 2010-04-19 16:49:30 -0500 (Mon, 19 Apr 2010) | 6 lines
+ Don't consider a missing indications.conf to be a critical error.
+ There were many changes in revision 176627 which would avoid the
+ error that a missing config would have caused. Other than this,
+ there are no other config files (including asterisk.conf,
+ surprisingly) that are required. ........
+
+2010-04-19 18:30 +0000 [r257850] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 257810 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r257810 |
+ twilson | 2010-04-19 12:57:41 -0500 (Mon, 19 Apr 2010) | 5 lines
+ Fix incomplete CDR merge from r195881 Because res/res_features.c
+ was removed and main/cdr.c added, these changes didn't make it to
+ trunk and the 1.6.x branches ........
+
+2010-04-18 17:28 +0000 [r257771] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/cdr_odbc.conf.sample, /: Merged revisions 257768 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r257768 | tilghman | 2010-04-18 12:25:53 -0500 (Sun, 18
+ Apr 2010) | 2 lines Removing unused configuration parameters
+ ........
+
+2010-04-16 21:47 +0000 [r257740] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 257713 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r257713 | dhubbard | 2010-04-16 16:22:30 -0500
+ (Fri, 16 Apr 2010) | 28 lines Merged revisions 257686 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010)
+ | 21 lines Make the mixmonitor thread process audio frames faster
+ Mantis issue 17078 reports MixMonitor recordings have shorter
+ durations than the call duration. This was because the mixmonitor
+ thread was not processing frames from the audiohook fast enough.
+ The mixmonitor thread would slowly fall behind the most recent
+ audio frame and when the channel hangs up, the mixmonitor thread
+ would exit without processing the same number of frames as the
+ channel; leaving the mixmonitor recording shorter than actual
+ call duration. This revision fixes this issue by moving the
+ ast_audiohook_trigger_wait() and the subsequent audiohook.status
+ check into the block where the ast_audiohook_read_frame()
+ function returns NULL. (closes issue #17078) Reported by:
+ geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license
+ 733) Tested by: dhubbard, geoff2010 Review:
+ https://reviewboard.asterisk.org/r/611/ ........ ................
+
+2010-04-15 21:34 +0000 [r257510-257597] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, /, main/app.c: Merged revisions 257560
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r257560 | tilghman | 2010-04-15 16:26:19 -0500
+ (Thu, 15 Apr 2010) | 13 lines Merged revisions 257544 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010)
+ | 6 lines Allow application options with arguments to contain
+ parentheses, through a variety of escaping techniques. Fixes
+ SWP-1194 (ABE-2143). Review:
+ https://reviewboard.asterisk.org/r/604/ ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 257493 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010)
+ | 20 lines Merged revisions 257467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010)
+ | 13 lines Don't recreate peer, when responding to a repeated
+ deregistration attempt. When a reply to a deregistration is lost
+ in transmit, the client retries the deregistration. Previously,
+ this would cause a realtime/autocreate peer to be loaded back
+ into memory, after it had already been correctly purged. Instead,
+ we just want to resend the reply without loading the peer.
+ (closes issue #16908) Reported by: kkm Patches:
+ 20100412__issue16908.diff.txt uploaded by tilghman (license 14)
+ Tested by: kkm ........ ................
+
+2010-04-15 19:42 +0000 [r257344-257428] Leif Madsen <lmadsen@digium.com>
+
+ * doc/backtrace.txt: Merged revisions 257427 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r257427 | lmadsen | 2010-04-15 14:41:05 -0500 (Thu, 15 Apr 2010)
+ | 21 lines Merged revisions 257426 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010)
+ | 13 lines Update backtrace.txt documentation. Update the
+ backtrace.txt documentation so it conforms to the same layout as
+ other documents we've been working on recently. Additionally, add
+ a bunch of new information about gathering backtraces for crashes
+ and deadlocks, along with ways of verifying your file before
+ uploading it. Create a couple of one line commands for people to
+ generate the files we need. (closes issue #17190) Reported by:
+ lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen
+ (license 10) Tested by: lmadsen, pabelanger ........
+ ................
+
+ * doc/backtrace.txt: Merged revisions 257343 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r257343 | lmadsen | 2010-04-15 08:44:38 -0500 (Thu, 15 Apr 2010)
+ | 9 lines Merged revisions 257342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010)
+ | 1 line Update address of the bug tracker. ........
+ ................
+
+2010-04-14 23:00 +0000 [r257265] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/features.conf.sample, /, main/features.c: Merged
+ revisions 257262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r257262 |
+ tilghman | 2010-04-14 17:57:35 -0500 (Wed, 14 Apr 2010) | 15
+ lines Yet another issue where the conversion of the application
+ delimiter to comma caused an issue. Application arguments within
+ the feature map could possibly contain a comma, which conflicts
+ with the syntax of the features.conf configuration file. This
+ patch allows the argument to be wrapped in parentheses or quoted,
+ to allow the application arguments to be interpreted as a single
+ configuration parameter. (closes issue #16646) Reported by:
+ pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman Review:
+ https://reviewboard.asterisk.org/r/547/ ........
+
+2010-04-13 19:20 +0000 [r257210] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 257191 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 |
+ tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10
+ lines Also unref the pvt when we delete the provisional keepalive
+ job. (closes issue #16774) Reported by: kowalma Patches:
+ 20100315__issue16774.diff.txt uploaded by tilghman (license 14)
+ Tested by: falves11, jamicque Review:
+ https://reviewboard.asterisk.org/r/591/ ........
+
+2010-04-13 18:43 +0000 [r257184] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /, configs/manager.conf.sample: Merged revisions
+ 257146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r257146 | mnicholson | 2010-04-13 13:10:30 -0500 (Tue, 13 Apr
+ 2010) | 16 lines Merged revisions 257070 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr
+ 2010) | 9 lines Add an option to restore past broken behavor of
+ the Events manager action Before r238915, certain values for the
+ EventMask parameter of the Events action would result in no
+ response being returned. This patch adds an option to restore
+ that broken behavior. Also while fixing this bug I discovered
+ that passing an empty EventMasks parameter would also result in
+ no response being returned, this has been fixed as well while
+ being preserved when the broken behavior is requested. (closes
+ issue #17023) Reported by: nblasgen Review:
+ https://reviewboard.asterisk.org/r/602/ ........ ................
+
+2010-04-13 16:38 +0000 [r257068] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 257065 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r257065 | tilghman | 2010-04-13 11:33:21 -0500 (Tue, 13 Apr 2010)
+ | 8 lines Ensure that we can have commas within cdr values.
+ (closes issue #17001) Reported by: snuffy Patches:
+ 20100412__issue17001.diff.txt uploaded by tilghman (license 14)
+ Tested by: snuffy ........
+
+2010-04-12 17:30 +0000 [r256822-256902] Leif Madsen <lmadsen@digium.com>
+
+ * doc/HOWTO_collect_debug_information.txt (added): Merged revisions
+ 256901 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r256901 | lmadsen | 2010-04-12 12:29:53 -0500 (Mon, 12 Apr 2010)
+ | 23 lines Merged revisions 256900 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010)
+ | 15 lines Add How-To document on collecting debugging info for
+ issues.asterisk.org Paul Belanger has been helping a lot with bug
+ tracking recently and created this document that we can now point
+ to when additional debugging information is required. This
+ document will help those filing issues to know how to get the
+ information required when filing their issues. This will make
+ things easier on the developers. Initial text and changes by
+ pabelanger. Tweaks and editing by myself. (closes issue #17159)
+ Reported by: pabelanger Patches:
+ HOWTO_collect_debug_information.txt.patch uploaded by lmadsen
+ (license 10) Tested by: tzafrir, pabelanger, lmadsen ........
+ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 256860 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r256860 | lmadsen | 2010-04-12 11:16:43 -0500 (Mon, 12 Apr 2010)
+ | 3 lines Remove silly debug message that is not useful. (issue
+ #17159) ........
+
+ * /, main/logger.c: Merged revisions 256821 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r256821 |
+ lmadsen | 2010-04-12 09:39:37 -0500 (Mon, 12 Apr 2010) | 8 lines
+ CLI command logger set level auto complete. A simple patch to
+ enable auto tab complete. (closes issue #17152) Reported by:
+ pabelanger Patches: 0017152.patch uploaded by pabelanger (license
+ 224) ........
+
+2010-04-08 22:03 +0000 [r256483] Tilghman Lesher <tlesher@digium.com>
+
+ * main/app.c: Backport /proc/%d/fd method of closing file
+ descriptors to 1.6.2.
+
+2010-04-06 19:40 +0000 [r256373] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h: Merged revisions 256370 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010)
+ | 2 lines Mac OS X does not support comparing a mutex to its
+ initializer. Create a test for this. ........
+
+2010-04-06 18:53 +0000 [r256268-256368] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: CallerID channel DAHDI port FXS are empty
+ after the first call. The bug is exposed if MFC/R2 support is
+ built into asterisk (i.e., openr2.h is present in the include
+ path). Code that unconditionally clears the CallerID name and
+ number is included. Also fixed a malformed if test in mkintf()
+ added by issue 15883. Converted the if statement to a switch
+ statement for clarity. Regression of the issue 15883 fix. (closes
+ issue #16968) Reported by: grecco Patches: issue16968.patch
+ uploaded by rmudgett (license 664) (closes issue #16747) Reported
+ by: viniciusfontes
+
+ * channels/chan_dahdi.c, /: Merged revisions 256265 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500
+ (Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010)
+ | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by
+ PRI lock. SWP-1231 ABE-2163 ........ ................
+
+2010-05-03 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.7 Released
+
+2010-04-29 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.7-rc3 Released
+
+2010-04-29 10:31 +0000 [r260053] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in
+ audiohook_write_list. (closes issue 0017052) Reported by: dvossel
+ Tested by: dvossel. (closes issue 0016196) Reported by: atis.
+ Review: https://reviewboard.asterisk.org/r/623/
+
+2010-04-28 10:31 +0000 [r259899] David Vossel <dvossel@digium.com>
+
+ * channels/chan_local.c, main/channel.c: Resolves deadlocks in
+ chan_local. (closes issue 0017185) Reported by: schmoozecom
+ Patches: issue_17185_v1.diff uploaded by dvossel (license 671)
+ issue_17185_v2.diff uploaded by dvossel (license 671) Tested
+ by: schmoozecom, GameGamer43
+ Review: https://reviewboard.asterisk.org/r/631/
+
+2010-04-13 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.7-rc2 Released
+
+2010-04-13 [r257210] Tilghman Lesher <tlesher@digium.com>
+
+ Also unref the pvt when we delete the provisional keepalive job.
+
+ (closes issue #16774)
+ Reported by: kowalma
+ Patches:
+ 20100315__issue16774.diff.txt uploaded by tilghman (license 14)
+ Tested by: falves11, jamicque
+
+ Review: https://reviewboard.asterisk.org/r/591/
+
+2010-04-05 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.7-rc1 Released
+
+2010-04-05 15:15 +0000 [r256162] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/localchannel.tex, /: Merged revisions 256161 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r256161 | lmadsen | 2010-04-05 10:14:53 -0500 (Mon, 05 Apr 2010)
+ | 1 line Fix for localchannel.tex to allow PDFs to be generated
+ again. ........
+
+2010-04-02 23:56 +0000 [r256013-256020] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 256019 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r256019 |
+ russell | 2010-04-02 18:55:57 -0500 (Fri, 02 Apr 2010) | 10 lines
+ Export MEETMEBOOKID and fix pin-less conferences with realtime
+ conferences (closes issue #16866) Reported by: DEA Patches:
+ rt-meetme-options.txt uploaded by DEA (license 3) Tested by: DEA
+ Review: https://reviewboard.asterisk.org/r/582/ ........
+
+ * channels/chan_local.c, /: Merged revisions 256015 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r256015 | russell | 2010-04-02 18:46:45 -0500
+ (Fri, 02 Apr 2010) | 16 lines Merged revisions 256014 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010)
+ | 9 lines Resolve a deadlock that occurs due to a pointless call
+ to ast_bridged_channel() (closes issue #16840) Reported by:
+ bzing2 Patches: patch.txt uploaded by bzing2 (license 902)
+ issue_16840.rev1.diff uploaded by russell (license 2) Tested by:
+ bzing2, russell ........ ................
+
+ * main/channel.c, /: Merged revisions 256010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r256010 | russell | 2010-04-02 18:30:58 -0500 (Fri, 02 Apr 2010)
+ | 9 lines Merged revisions 256009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010)
+ | 2 lines Remove extremely verbose debug message. ........
+ ................
+
+2010-04-02 20:20 +0000 [r255955] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 255952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r255952 |
+ tilghman | 2010-04-02 15:19:01 -0500 (Fri, 02 Apr 2010) | 8 lines
+ Pass the PID of the Asterisk process, not the PID of the canary.
+ (closes issue #17065) Reported by: globalnetinc Patches:
+ astcanary.patch uploaded by makoto (license 38) Tested by: frawd,
+ globalnetinc ........
+
+2010-04-01 18:21 +0000 [r255676-255816] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 255796 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010)
+ | 7 lines Fix DEBUG_THREADS build on Darwin. (closes issue
+ #16828) Reported by: oej Patches: 20100331__issue16828.diff.txt
+ uploaded by tilghman (license 14) ........
+
+ * apps/app_voicemail.c, /: Recorded merge of revisions 255592 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r255592 | tilghman | 2010-03-31 14:13:02 -0500
+ (Wed, 31 Mar 2010) | 22 lines Recorded merge of revisions 255591
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010)
+ | 15 lines Ensure line terminators in email are consistent. Fixes
+ an issue with certain Mail Transport Agents, where attachments
+ are not interpreted correctly. (closes issue #16557) Reported by:
+ jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by
+ tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt
+ uploaded by tilghman (license 14)
+ 20100308__issue16557__trunk.diff.txt uploaded by tilghman
+ (license 14) Tested by: ebroad, zktech Reviewboard:
+ https://reviewboard.asterisk.org/r/544/ ........ ................
+
+2010-03-31 17:49 +0000 [r255505] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample, apps/app_dial.c: Merged revisions 255504
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31
+ Mar 2010) | 5 lines Add documentation clarifying when 't' and 'T'
+ can be used. (closes issue #17021) Reported by: kovzol Tested by:
+ lmadsen, kovzol, davidw, ebroad ........
+
+2010-03-30 20:58 +0000 [r255326-255413] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_h323.c: Merged revisions 255410 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r255410 | russell | 2010-03-30 15:56:26 -0500
+ (Tue, 30 Mar 2010) | 9 lines Merged revisions 255409 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30
+ Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does
+ not start. ........ ................
+
+ * /, pbx/pbx_dundi.c: Merged revisions 255323 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r255323 | russell | 2010-03-30 11:07:49 -0500 (Tue, 30 Mar 2010)
+ | 9 lines Merged revisions 255322 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010)
+ | 2 lines Don't make Asterisk not start if pbx_dundi fails to
+ initialize. ........ ................
+
+2010-03-26 19:28 +0000 [r255023-255067] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample, /: Merged revisions 255066 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r255066 | lmadsen | 2010-03-26 14:27:56 -0500 (Fri, 26 Mar 2010)
+ | 6 lines Replace some documentation from 1.6.x back into trunk.
+ This documentation associated wth tlsbindaddr is still useful so
+ lets synchronize it between trunk and 1.6.x branches. (issue
+ #17054) ........
+
+ * configs/sip.conf.sample, /: Merged revisions 255021 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010)
+ | 8 lines Update confusing documentation for tlsbindaddr. Update
+ some confusing documentation for the tlsbindaddr option in
+ sip.conf.sample. Point at a link instead which has better
+ documentation. (closes issue #17054) Reported by: klaus3000
+ ........
+
+2010-03-25 20:43 +0000 [r254770-254805] Jason Parker <jparker@digium.com>
+
+ * utils/Makefile, /: Merged revisions 254802 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r254802 | qwell | 2010-03-25 15:41:49 -0500 (Thu, 25 Mar 2010) |
+ 9 lines Merged revisions 254800 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) |
+ 1 line Don't remove local copies of utils in uninstall. ........
+ ................
+
+ * main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS
+ issue with out-of-tree modules. Take 2, without ABI breakage this
+ time. Review: https://reviewboard.asterisk.org/r/588/
+
+2010-03-25 20:09 +0000 [r254721] Russell Bryant <russell@digium.com>
+
+ * channels/chan_usbradio.c, /: Merged revisions 254718 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010)
+ | 2 lines chan_usbradio depends on alsa. ........
+
+2010-03-25 17:47 +0000 [r254556] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/acl.h, /: Merged revisions 254553 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r254553 | mmichelson | 2010-03-25 12:42:36 -0500
+ (Thu, 25 Mar 2010) | 11 lines Merged revisions 254552 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar
+ 2010) | 5 lines Add doxygen for acl.h Review:
+ https://reviewboard.asterisk.org/r/528 ........ ................
+
+2010-03-25 17:21 +0000 [r254548] Sean Bright <sean@malleable.com>
+
+ * channels/chan_sip.c: Initialize stream to avoid a compilation
+ error.
+
+2010-03-25 17:12 +0000 [r254542] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix potential crashes from trying to
+ reference nonexistent RTP streams.
+
+2010-03-25 16:26 +0000 [r254499] Terry Wilson <twilson@digium.com>
+
+ * /, main/file.c: Merged revisions 254453 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r254453 | twilson | 2010-03-25 11:03:51 -0500 (Thu, 25 Mar 2010)
+ | 9 lines Merged revisions 254451 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010)
+ | 2 lines Handle new SRCCHANGE control message here too ........
+ ................
+
+2010-03-25 16:22 +0000 [r254482] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, /: Recorded merge of revisions 254454 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r254454 | mmichelson | 2010-03-25 11:04:48 -0500
+ (Thu, 25 Mar 2010) | 50 lines Recorded merge of revisions 254452
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar
+ 2010) | 44 lines Several fixes regarding RFC2833 DTMF detection.
+ Here is a copy and paste of the details from my request on
+ reviewboard that dealt with these changes: Fix 1. The first
+ change in place is to fix Mantis issue 15811, which deals with a
+ situation where Asterisk will incorrectly interpret out of order
+ RFC2833 frames as duplicate DTMF digits. For instance, we would
+ receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3:
+ DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1
+ seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch
+ when we received the frame with seqno 5, we would interpret this
+ as a new DTMF 1. With this patch, we will check the seqno of the
+ incoming digit and not process the frame if the seqno is lower
+ than the last recorded seqno. Note that we do not record the
+ seqno of the dropped DTMF frame for future processing. While the
+ above situation is what was designed to be fixed, the patch is
+ written in such a way that the following would also be fixed too:
+ seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end)
+ seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno
+ 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In
+ this second situation, the beginning of the DTMF 2 arrives before
+ the final end frame of the DTMF 1. With the patch, seqno 12 is no
+ processed and thus we properly interpret the DTMF. Fix 2. The
+ second change in place is to fix an issue like the following:
+ seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet
+ lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end)
+ *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had
+ code in place that was supposed to properly end the previously
+ unended DTMF 1. The problem was that the code was essentially a
+ no-op. The code would set up an end frame for the DTMF 1 but
+ would immediately overwrite the frame with the begin for DTMF 2.
+ I changed process_dtmf_rfc2833() so that instead of returning a
+ single frame, it is given as an output parameter a list of
+ frames. Each frame that needs to be returned is appended to this
+ list. Fix 3. The final change is a minor one where an
+ AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco
+ DTMF or an RFC 3389 frame and no frame was returned, then we
+ would return &ast_null_frame. The problem is that earlier in the
+ function, we may have generated an AST_CONTROL_SRCCHANGE frame
+ and put it in the list of frames we wish to return. This frame
+ would be lost in such a case. The patch fixes this problem
+ ........ ................
+
+2010-03-25 15:21 +0000 [r254447] Leif Madsen <lmadsen@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 254446 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r254446 |
+ lmadsen | 2010-03-25 10:21:26 -0500 (Thu, 25 Mar 2010) | 9 lines
+ handle_speechset has 4 arguments. Update code to reflect that
+ handle_speechset has 4 arguments. (closes issue #17093) Reported
+ by: gpatri Patches: res_agi.patch uploaded by gpatri (license
+ 1014) Tested by: pabelanger, mmichelson ........
+
+2010-03-24 17:19 +0000 [r254288] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 254277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r254277 | jpeeler | 2010-03-24 12:15:05 -0500 (Wed, 24 Mar 2010)
+ | 78 lines Merged revisions 254235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010)
+ | 72 lines Ensure that monitor recordings are written to the
+ correct location (again) This is an extension to 248860. As such
+ the dialplan test has been extended: ; non absolute path, not
+ combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
+ exten => 5040, n, dial(sip/5001) ; absolute path, not combined
+ exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten =>
+ 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1,
+ monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
+ exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n,
+ dial(sip/5001) ; combined: changemonitor from no path to non
+ absolute path exten => 5044, 1, monitor(wav,monitor_test6,m)
+ exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this
+ wasn't possible before exten => 5044, n, dial(sip/5001) ; non
+ absolute path, combined exten => 5045, 1,
+ monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n,
+ dial(sip/5001) ; absolute path, combined exten => 5046, 1,
+ monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n,
+ dial(sip/5001) ; no path, combined exten => 5047, 1,
+ monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to absolute (leaves
+ tmp/jeff) exten => 5048, 1,
+ monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n,
+ changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5049, 1,
+ monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n,
+ changemonitor(tmp/jeff/monitor_test14) exten => 5049, n,
+ dial(sip/5001) ; combined: changemonitor from no path to absolute
+ exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n,
+ changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to no path
+ (leaves /tmp/jeff) exten => 5051, 1,
+ monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n,
+ changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ;
+ not combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
+ exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to non
+ absolute exten => 5053, 1, monitor(wav,monitor_test21) exten =>
+ 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n,
+ dial(sip/5001) ; not combined: changemonitor from non absolute to
+ absolute (leaves tmp/jeff) exten => 5054, 1,
+ monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n,
+ changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n,
+ dial(sip/5001) ; not combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5055, 1,
+ monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n,
+ changemonitor(tmp/jeff/monitor_test25) exten => 5055, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to
+ absolute exten => 5056, 1, monitor(wav,monitor_test26) exten =>
+ 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056,
+ n, dial(sip/5001) ; not combined: changemonitor from absolute to
+ no path (leaves /tmp/jeff) exten => 5057, 1,
+ monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n,
+ changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001)
+ ........ ................
+
+2010-03-23 22:05 +0000 [r254131] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * tests/Makefile, /: Merged revisions 254001 via svnmerge from
+ http://svn.digium.com/svn/asterisk/trunk ........ r254001 |
+ tzafrir | 2010-03-23 21:19:52 +0200 (Tue, 23 Mar 2010) | 2 lines
+ Change the name of the category 'TEST' to match the name of the
+ subdir ........
+
+2010-03-23 21:20 +0000 [r254068] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 254050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r254050 |
+ jpeeler | 2010-03-23 16:17:23 -0500 (Tue, 23 Mar 2010) | 14 lines
+ Exit native bridging early for greater timing accuracy with
+ warnings This changes native bridging to break one millisecond
+ early so that the more accurate timeval calculations done in the
+ generic bridge can be performed using the bridge config.
+ Currently the time between exiting native bridging slightly late
+ can sometimes cause a large enough discrepancy for warnings to be
+ missed. For the record, 1.4 does not attempt to native bridge at
+ all when warnings are enabled. (closes issue #15815) Reported by:
+ adomjan Review: https://reviewboard.asterisk.org/r/577/ ........
+
+2010-03-22 19:55 +0000 [r253801] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 253800 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r253800 | mnicholson | 2010-03-22 14:52:52 -0500 (Mon, 22 Mar
+ 2010) | 11 lines Merged revisions 253799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar
+ 2010) | 4 lines Unconditionally copy the caller's account code to
+ the called party. (related to issue #16331) ........
+ ................
+
+2010-03-22 19:06 +0000 [r253714-253760] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/scripts/dbsep.cgi: Merged revisions 253758 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r253758 | tilghman | 2010-03-22 14:05:27 -0500 (Mon, 22
+ Mar 2010) | 2 lines Update query should be an UPDATE, not a
+ SELECT. ........
+
+ * /, contrib/scripts/dbsep.cgi: Merged revisions 253755 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r253755 | tilghman | 2010-03-22 13:58:48 -0500 (Mon, 22
+ Mar 2010) | 4 lines Return the list for later manipulation. This
+ fixes an issue with the update procedure. Debugging with
+ mmichelson. ........
+
+ * configs/dbsep.conf.sample, /, contrib/scripts/dbsep.cgi: Merged
+ revisions 253712 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253712 |
+ tilghman | 2010-03-22 11:59:35 -0500 (Mon, 22 Mar 2010) | 2 lines
+ Accomodate equal signs in DSNs and add documentation, based upon
+ mmichelson's feedback. ........
+
+2010-03-20 17:33 +0000 [r253595-253620] Russell Bryant <russell@digium.com>
+
+ * cdr/cdr_pgsql.c, main/stdtime/localtime.c, main/tcptls.c, /,
+ main/features.c: Merged revisions 253540 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253540 |
+ russell | 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines
+ Resolve more compiler warnings on FreeBSD. ........
+
+ * apps/app_followme.c, apps/app_dial.c, /: Merged revisions 253538
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r253538 | russell | 2010-03-20 06:43:08 -0500 (Sat, 20
+ Mar 2010) | 2 lines Resolve compiler warnings on FreeBSD.
+ ........
+
+ * /, pbx/pbx_dundi.c: Merged revisions 253537 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253537 |
+ russell | 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines
+ Resolve a compiler warning on FreeBSD. ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 253536 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r253536 | russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010)
+ | 4 lines Use SHRT_MAX instead of MAXSHORT. These changes fix
+ build issues I had with this module on FreeBSD. ........
+
+2010-03-19 08:05 +0000 [r253492] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/astobj2.c, /: Merged revisions 253490 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253490 |
+ alecdavis | 2010-03-19 20:37:00 +1300 (Fri, 19 Mar 2010) | 19
+ lines prevent segfault if bad magic number is encountered.
+ internal_ao2_ref uses INTERNAL_OBJ which mzy report 'bad magic
+ number', but internal_ao2_ref continues on, causing segfault.
+ Although AO2_MAGIC number is checked by INTERNAL_OBJ before
+ internal_ao2_ref is called, A02_MAGIC is being destroyed (or a
+ wrong pointer) by the time internal_ao2_ref uses INTERNAL_OBJ.
+ internal_ao2_ref now returns -1 if INTERNAL_OBJ encouters a bad
+ magic number. (issue #17037) Reported by: alecdavis Patches:
+ bug17037.diff.txt uploaded by alecdavis (license 585) Tested by:
+ alecdavis ........
+
+2010-03-18 17:54 +0000 [r253257-253346] Leif Madsen <lmadsen@digium.com>
+
+ * /, apps/app_userevent.c: Merged revisions 253345 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r253345 | lmadsen | 2010-03-18 12:52:35 -0500 (Thu, 18 Mar 2010)
+ | 7 lines Change usage of pipe to comma in UserEvent docs. Change
+ the example usage of pipe as a separator to comma in the
+ UserEvent documentation. (closes issue #16961) Reported by:
+ jlpedrosa ........
+
+ * doc/tex/localchannel.tex: Merged revisions 253256 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r253256 | lmadsen | 2010-03-18 10:46:52 -0500 (Thu, 18 Mar 2010)
+ | 9 lines Update to new Local channel documentation. Add same
+ changes as commit to 1.4, but convert to TeX. (issue #16963)
+ Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz
+ (license 834) ........
+
+2010-03-17 16:25 +0000 [r253158] Terry Wilson <twilson@digium.com>
+
+ * main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c,
+ channels/chan_mgcp.c, channels/chan_sip.c,
+ include/asterisk/rtp.h: Revert API change in release branches
+ This re-renames ast_rtp_update_source to ast_rtp_new_source
+
+2010-03-17 00:41 +0000 [r253029-253033] Leif Madsen <lmadsen@digium.com>
+
+ * main/xmldoc.c, /: Merged revisions 253032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253032 |
+ lmadsen | 2010-03-16 19:40:51 -0500 (Tue, 16 Mar 2010) | 1 line
+ Fix a typo. ........
+
+ * configs/say.conf.sample, /: Merged revisions 253028 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r253028 | lmadsen | 2010-03-16 19:29:06 -0500
+ (Tue, 16 Mar 2010) | 13 lines Merged revisions 253018 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010)
+ | 6 lines Add french snipset to say.conf. Add the french snipset
+ to say.conf. (Closes issue #15799) ........ ................
+
+2010-03-16 23:54 +0000 [r252978] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, /: Merged revisions 252976 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252976 |
+ tilghman | 2010-03-16 18:49:35 -0500 (Tue, 16 Mar 2010) | 8 lines
+ Mask out previous arguments on each nested invocation of Gosub.
+ (closes issue #16758) Reported by: wdoekes Patches:
+ 20100316__issue16758.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/561/ ........
+
+2010-03-16 19:38 +0000 [r252850] Sean Bright <sean@malleable.com>
+
+ * res/res_clialiases.c, /: Merged revisions 252848 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r252848 | seanbright | 2010-03-16 15:36:24 -0400 (Tue, 16 Mar
+ 2010) | 10 lines Include an extra newline after "Aliased CLI
+ command" to get back the prompt. The other issue mentioned in
+ this bug will be more difficult to resolve since we have no idea
+ (right now) of knowing if the command that is aliased has been
+ installed yet. (issue #16978) Reported by: jw-asterisk Tested by:
+ seanbright ........
+
+2010-03-16 19:02 +0000 [r252770] Russell Bryant <russell@digium.com>
+
+ * utils/Makefile, /: Merged revisions 252767 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r252767 | russell | 2010-03-16 14:01:04 -0500 (Tue, 16 Mar 2010)
+ | 13 lines Merged revisions 252766 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010)
+ | 6 lines Don't treat warnings as errors for muted. muted
+ supports OS X, but uses functions marked as deprecated in 10.6.
+ However, the functions are still supported, so just ignore the
+ warnings for now and allow the build to proceed. ........
+ ................
+
+2010-03-16 18:49 +0000 [r252763] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.ael.sample, /: Merged revisions 252762 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r252762 | lmadsen | 2010-03-16 13:48:22 -0500
+ (Tue, 16 Mar 2010) | 15 lines Merged revisions 252761 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010)
+ | 7 lines Additional extensions.ael global variable fixes. Fixing
+ up a couple more overlapping global variable namespaces shared
+ with extensions.conf.sample. Also noticed a few of the lines that
+ were commented out didn't have the closing semi-colon so I added
+ that as well. (issue #17035) ........ ................
+
+2010-03-15 21:59 +0000 [r252626] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_meetme.c: Merged revisions 252623 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252623 |
+ seanbright | 2010-03-15 17:55:44 -0400 (Mon, 15 Mar 2010) | 4
+ lines Resolve a crash in SLATrunk when the specified trunk
+ doesn't exist. Reported by philipp64 in #asterisk-dev. ........
+
+2010-03-15 21:54 +0000 [r252622] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions
+ 252619 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r252619 | tilghman | 2010-03-15 16:51:55 -0500 (Mon, 15 Mar 2010)
+ | 9 lines Merged revisions 252617 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010)
+ | 2 lines Uh, yeah. Umask. I'm stupid. ........ ................
+
+2010-03-15 20:53 +0000 [r252535] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.ael.sample: Merged revisions 252534 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r252534 | lmadsen | 2010-03-15 15:52:32 -0500
+ (Mon, 15 Mar 2010) | 15 lines Merged revisions 252533 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010)
+ | 7 lines Update extensions.ael file to not overlap
+ extensions.conf. Updated the extensions.ael file so the global
+ variables don't overlap those that we have in extensions.conf
+ (sample files). This way unexpected things won't happed hopefully
+ if both pbx_ael and res_config are loaded. (closes issue #17035)
+ Reported by: pprindeville ........ ................
+
+2010-03-15 05:04 +0000 [r252365-252444] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 252442 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252442 |
+ tilghman | 2010-03-14 23:25:35 -0500 (Sun, 14 Mar 2010) | 7 lines
+ THIS IS NOT PYTHON. Indentation doesn't matter, only braces do.
+ (closes issue #17025) Reported by: smurfix Patches: sip.patch
+ uploaded by smurfix (license 547) ........
+
+ * main/asterisk.c, Makefile,
+ contrib/init.d/org.asterisk.asterisk.plist (added), /: Merged
+ revisions 252362 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r252362 | tilghman | 2010-03-14 20:37:04 -0500 (Sun, 14 Mar 2010)
+ | 11 lines Merged revisions 252361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010)
+ | 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard:
+ https://reviewboard.asterisk.org/r/551/ ........ ................
+
+2010-03-14 17:48 +0000 [r252317] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 252314 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r252314 | seanbright | 2010-03-14 13:43:46 -0400 (Sun, 14 Mar
+ 2010) | 8 lines Fix building CDR and CEL SQLite3 modules. They
+ added a sqlite3_log() function which was conflicting with our
+ function names. (closes issue #17017) Reported by: alephlg
+ ........
+
+2010-03-13 00:32 +0000 [r252137-252178] Terry Wilson <twilson@digium.com>
+
+ * main/rtp.c: Remove unusued field
+
+ * configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c,
+ channels/chan_mgcp.c, main/channel.c, /, channels/chan_sip.c,
+ channels/chan_skinny.c, include/asterisk/rtp.h,
+ channels/chan_h323.c: Merged revisions 252089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 |
+ twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
+ Only change the RTP ssrc when we see that it has changed This
+ change basically reverts the change reviewed in
+ https://reviewboard.asterisk.org/r/374/ and instead limits the
+ updating of the RTP synchronization source to only those times
+ when we detect that the other side of the conversation has
+ changed the ssrc. The problem is that SRCUPDATE control frames
+ are sent many times where we don't want a new ssrc, including
+ whenever Asterisk has to send DTMF in a normal bridge. This is
+ also not the first time that this mistake has been made. The
+ initial implementation of the ast_rtp_new_source function also
+ changed the ssrc--and then it was removed because of this same
+ issue. Then, we put it back in again to fix a different issue.
+ This patch attempts to only change the ssrc when we see that the
+ other side of the conversation has changed the ssrc. It also
+ renames some functions to make their purpose more clear. Review:
+ https://reviewboard.asterisk.org/r/540/ ........
+
+2010-03-12 22:05 +0000 [r252090] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 252088 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r252088 | moy | 2010-03-12 16:57:40 -0500 (Fri, 12 Mar 2010) | 1
+ line add missing mfcr2_skip_category setting ........
+
+2010-03-12 19:50 +0000 [r251994] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 251989 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r251989 | tilghman | 2010-03-12 13:43:23 -0600 (Fri, 12 Mar 2010)
+ | 8 lines Don't override a user option with the global option.
+ (closes issue #16849) Reported by: ip-rob Patches:
+ 20100311__issue16849.diff.txt uploaded by tilghman (license 14)
+ Tested by: ip-rob ........
+
+2010-03-12 19:49 +0000 [r251991] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 251946 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r251946 | rmudgett | 2010-03-12 13:05:40 -0600 (Fri, 12 Mar 2010)
+ | 1 line Doxegen this chan_dahdi lock. ........
+
+2010-03-11 21:08 +0000 [r251879-251887] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_exec.c, /: Merged revisions 251884 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r251884 |
+ tilghman | 2010-03-11 15:07:07 -0600 (Thu, 11 Mar 2010) | 8 lines
+ Because ExecIf needs to reprocess arguments, it's best if we
+ don't remove quotes during parsing. (closes issue #16905)
+ Reported by: ip-rob Patches: 20100303__issue16905.diff.txt
+ uploaded by tilghman (license 14) Tested by: ip-rob ........
+
+ * apps/app_system.c, /: Merged revisions 251877 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r251877 |
+ tilghman | 2010-03-11 14:25:02 -0600 (Thu, 11 Mar 2010) | 8 lines
+ If the argument to the system application is quoted, ensure we
+ remove the quotes before trying to execute. (closes issue #16842)
+ Reported by: ip-rob Patches: 20100310__issue16842.diff.txt
+ uploaded by tilghman (license 14) Tested by: ip-rob ........
+
+2010-03-11 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.6 released
+
+2010-03-05 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.6-rc2 released
+
+2010-03-05 Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 250913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r250913 | tilghman
+ | 2010-03-04 22:37:36 -0600 (Thu, 04 Mar 2010) | 7 lines Missing quote in
+ ODBC query. (closes issue #16953) Reported by: elguero Patches:
+ app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37)
+ ........
+
+2010-03-04 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.6-rc1 released
+
+2010-03-03 21:24 +0000 [r250610] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/localchannel.tex, /: Merged revisions 250609 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r250609 | lmadsen | 2010-03-03 16:22:55 -0500 (Wed, 03 Mar 2010)
+ | 11 lines Update existing Local channel documentation. A
+ complete re-write of the Local channel documentation has been
+ performed, with the existing information from localchannel.txt
+ and localchannel.tex merged in. (closes issue #16637) Reported
+ by: kobaz Patches: localchannel.tex uploaded by lmadsen (license
+ 10) localchannel.txt uploaded by lmadsen (license 10) Tested by:
+ lmadsen, jsmith, mmichelson ........
+
+2010-03-03 19:13 +0000 [r250484] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 250481 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600
+ (Wed, 03 Mar 2010) | 22 lines Merged revisions 250480 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010)
+ | 15 lines Make sure to clear red alarm after polarity reversal.
+ From the issue: The automatic overnight line tests (or manual
+ ones) used on UK (BT) lines causes a red alarm on a dahdi /
+ TDM400P connected channel. This is because the line uses voltage
+ tests (battery loss) and polarity reversal. The polarity reversal
+ causes chan_dahdi to initiate v23 CallerID processing but during
+ this the event DAHDI_EVENT_NOALARM is ignored so that the alarm
+ is never cleared. (closes issue #14163) Reported by: jedi98
+ Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license
+ 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........
+ ................
+
+2010-03-03 18:05 +0000 [r250253-250396] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 250395 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r250395 | dvossel | 2010-03-03 12:03:19 -0600
+ (Wed, 03 Mar 2010) | 22 lines Merged revisions 250394 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010)
+ | 16 lines fixes problem with duplicate TXREQ packets When
+ Asterisk receives an IAX2 TXREQ packet, try_transfer() will call
+ store_by_transfercallno() to link the chan_iax2_pvt struct into
+ iax_transfercallno_pvts. If a duplicate TXREQ packet is received
+ for the same call, the pvt struct will be linked into
+ iax_transfercallno_pvts multiple times. This patch fixes this.
+ Thanks rain for debugging this and providing a patch! (closes
+ issue #16904) Reported by: rain Patches:
+ iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
+ by: rain, dvossel ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 250246 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 |
+ dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines
+ fixes signed to unsigned int comparision issue for FaxMaxDatagram
+ value. ........
+
+2010-03-02 21:10 +0000 [r249953-250052] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/imapstorage.tex, /: Merged revisions 250051 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010)
+ | 8 lines Update IMAP documentation. Update the IMAP
+ documentation to make it clear that storing voicemails in the
+ same folder as a large number of emails could potentially cause
+ significant slow downs when writing or retrieving voicemails.
+ (issue #16704) Reported by: TimeHider Tested by: lmadsen,
+ TimeHider ........
+
+ * configs/cdr.conf.sample: Merged revisions 250045 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500
+ (Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010)
+ | 7 lines Update documentation to clarify purpose of unanswered
+ option. (closes issue #16267) Reported by: elsto Patches:
+ cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested
+ by: davidw, elsto ........ ................
+
+ * doc/tex/configuration.tex, /: Merged revisions 250037 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02
+ Mar 2010) | 4 lines Update documentation to not imply we support
+ overriding options. (closes issue #16855) Reported by: davidw
+ ........
+
+ * apps/app_directory.c, /: Merged revisions 249950 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r249950 | lmadsen | 2010-03-02 14:49:48 -0500 (Tue, 02 Mar 2010)
+ | 4 lines Fix literal values wrapped in documentation. (closes
+ issue #16145) Reported by: tilghman ........
+
+2010-03-02 19:50 +0000 [r249952] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * UPGRADE-1.6.txt, main/editline/makelist.in, apps/app_echo.c,
+ UPGRADE.txt: revert ability to exit echo app caused a regression,
+ as only supported VOICE, not VIDEO etc. (issue #16880)
+
+2010-03-02 19:26 +0000 [r249916-249933] Leif Madsen <lmadsen@digium.com>
+
+ * /, main/features.c: Merged revisions 249925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r249925 |
+ lmadsen | 2010-03-02 14:24:43 -0500 (Tue, 02 Mar 2010) | 6 lines
+ Add missing description of the PARKINGLOT variable in XML
+ documentation. (closes issue #16743) Reported by: snuffy Patches:
+ parkingdoc.diff uploaded by snuffy (license 35) ........
+
+ * /, pbx/pbx_dundi.c: Merged revisions 249912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r249912 |
+ lmadsen | 2010-03-02 14:21:19 -0500 (Tue, 02 Mar 2010) | 6 lines
+ Convert some DUNDI functions to XML documentation. (closes issue
+ #16798) Reported by: snuffy Patches: xml_dundi.diff uploaded by
+ snuffy (license 35) ........
+
+2010-03-02 19:12 +0000 [r249895] David Vossel <dvossel@digium.com>
+
+ * channels/chan_console.c, channels/chan_gtalk.c,
+ channels/chan_oss.c, channels/misdn_config.c,
+ include/asterisk/abstract_jb.h, configs/alsa.conf.sample,
+ channels/chan_jingle.c, channels/chan_usbradio.c,
+ channels/chan_dahdi.c, channels/chan_skinny.c,
+ configs/mgcp.conf.sample, main/abstract_jb.c,
+ channels/chan_h323.c, channels/chan_alsa.c,
+ configs/sip.conf.sample, channels/chan_mgcp.c,
+ channels/chan_unistim.c, configs/console.conf.sample,
+ configs/chan_dahdi.conf.sample, channels/chan_local.c,
+ configs/oss.conf.sample, channels/chan_sip.c, /,
+ configs/usbradio.conf.sample, configs/misdn.conf.sample: Merged
+ revisions 249893 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 |
+ dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines
+ fixes adaptive jitterbuffer configuration When configuring the
+ adaptive jitterbuffer, the target_extra value not only could not
+ be set from the configuration, but was not even being set to its
+ proper default. This value is required in order for the adaptive
+ jitterbuffer to work correctly. To resolve this a config option
+ has been added to expose this value to the conf files, and a
+ default value is provided when no config specific value is
+ present. ........
+
+2010-03-02 19:09 +0000 [r249894] Leif Madsen <lmadsen@digium.com>
+
+ * /, apps/app_confbridge.c: Merged revisions 249892 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r249892 | lmadsen | 2010-03-02 14:02:56 -0500 (Tue, 02 Mar 2010)
+ | 1 line Fix several XML documentation validate errors. ........
+
+2010-03-02 09:05 +0000 [r249844] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_echo.c: fixes ability to exit echo app when called from
+ a ISDN channel, null frames prevent '#' exit. Now only echo back
+ VOICE and DTMF frames (issue #16880) Reported by: alecdavis
+ Patches: echo_exit_1-6-1.diff.txt uploaded by alecdavis (license
+ 585) Tested by: alecdavis
+
+2010-03-01 19:40 +0000 [r249675] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 249672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r249672 | seanbright | 2010-03-01 14:36:30 -0500
+ (Mon, 01 Mar 2010) | 18 lines Merged revisions 249671 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar
+ 2010) | 11 lines Fix crash in app_voicemail related to message
+ counting. We were passing a 'struct inprocess **' and treating it
+ like a 'struct inprocess *' causing a segfault. (closes issue
+ #16921) Reported by: whardier Patches: 20100301_issue16921.patch
+ uploaded by seanbright (license 71) Tested by: whardier ........
+ ................
+
+2010-03-01 18:47 +0000 [r249625] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 249623 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r249623 | tilghman | 2010-03-01 12:36:06 -0600 (Mon, 01 Mar 2010)
+ | 2 lines Constify a bit of app_voicemail, to make ODBC and IMAP
+ compile once again. ........
+
+2010-03-01 17:25 +0000 [r249580] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 249538 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600
+ (Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010)
+ | 11 lines Modify queued frames from local channels to not set
+ the other side to up In this case, attended transfers were broken
+ due to ast_feature_request_and_dial detecting the channel being
+ set to up before the answer frame could be read and therefore
+ failing to mark the channel as ready. This fix is a regression
+ fix for 244785, which should continue to work properly as well.
+ (closes issue #16816) Reported by: jamhed Tested by: jamhed,
+ corruptor ........ ................
+
+2010-02-28 20:51 +0000 [r249407-249493] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 249491 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r249491 | tilghman | 2010-02-28 14:50:01 -0600 (Sun, 28 Feb 2010)
+ | 5 lines Fix unit test that Alec Davis broke. (closes issue
+ #16927) Reported by: alecdavis ........
+
+ * apps/app_voicemail.c, include/asterisk/app.h, /: Merged revisions
+ 249405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r249405 |
+ tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines
+ Properly document voicemail API documents. Also fix a crash
+ reported via the -dev list. ........
+
+2010-02-27 23:04 +0000 [r249321] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_dahdi.c: overlap receiving: automatically send CALL
+ PROCEEDING when dialplan starts Following Q.931 5.2.4 When the
+ user has determined that sufficient call information has been
+ received the user shall stop T302 and send CALL PROCEEDING to the
+ network. Previously timeouts were possible if the dialplan took a
+ long time to issue any response back to the network. Verified
+ that our local TELCO also does the same. (issue #16789) Reported
+ by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded
+ by alecdavis (license 585) Tested by: alecdavis
+
+2010-02-27 14:10 +0000 [r249238] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 249235 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500
+ (Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27
+ Feb 2010) | 1 line add a reference to the now-published IAX2 RFC
+ ........ ................
+
+2010-02-26 18:49 +0000 [r249190] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 249187 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r249187 | tilghman | 2010-02-26 12:41:57 -0600 (Fri, 26 Feb 2010)
+ | 18 lines Cleanups to fix bugs in the VM count API functions. -
+ Urgent voicemails were not attached, because the attachment code
+ looked in the wrong folder. - Urgent voicemails were sometimes
+ counted twice when displaying the count of new messages. -
+ Backends were inconsistent as to which voicemails each API
+ counted. (closes issue #15654) Reported by: tomo1657 Patches:
+ 20100225__issue15654.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman (closes issue #16448) Reported by: hevad
+ Review: https://reviewboard.asterisk.org/r/525/ ........
+
+2010-02-26 17:06 +0000 [r249104] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 249101 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb
+ 2010) | 14 lines Merged revisions 249100 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb
+ 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488.
+ (closes issue #16792) Reported by: vrban Patches: t38_606.patch
+ uploaded by vrban (license 756) ........ ................
+
+2010-02-25 23:12 +0000 [r248955] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 248952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010)
+ | 24 lines Merged revisions 248860 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010)
+ | 18 lines Ensure that monitor recordings are written to the
+ correct location (again) This is an extension to 248757. As such
+ the dialplan test has been extended: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) exten => 5043, 1,
+ monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n,
+ changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001)
+ exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n,
+ changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by
+ design and emits a warning exten => 5044, n, dial(sip/5001)
+ ........ ................
+
+2010-02-25 22:42 +0000 [r248949] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/acl.c: Merged revisions 248946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r248946 |
+ mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5
+ lines Fix incorrect ACL behavior when CIDR notation of "/0" is
+ used. AST-2010-003 ........
+
+2010-02-25 21:25 +0000 [r248864] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 248861 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010)
+ | 22 lines Merged revisions 248859 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010)
+ | 15 lines Some platforms clear /var/run at boot, which makes
+ connecting a remote console... difficult. Previously, we only
+ created the default /var/run/asterisk directory at install time.
+ While we could create it in the init script, that would not work
+ for those who start asterisk manually from the command line. So
+ the safest thing to do is to create it as part of the Asterisk
+ boot process. This also changes the ownership of the directory,
+ because the pid and ctl files are created after we setuid/setgid.
+ (closes issue #16802) Reported by: Brian Patches:
+ 20100224__issue16802.diff.txt uploaded by tilghman (license 14)
+ Tested by: tzafrir ........ ................
+
+2010-02-25 18:52 +0000 [r248797] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 248793 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010)
+ | 22 lines Merged revisions 248757 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010)
+ | 15 lines Ensure that monitor recordings are written to the
+ correct location. Recordings should be placed in the monitor
+ directory when a non-absolute path is used. Exact dialplan used
+ for testing: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) ABE-2101 ........
+ ................
+
+2010-02-24 21:29 +0000 [r248643] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/logger.c: Merged revisions 248584 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010)
+ | 14 lines Merged revisions 248582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010)
+ | 7 lines Remove color code sequences from verbose messages that
+ go to logfiles. (closes issue #16786) Reported by: dodo Patches:
+ logger2.patch uploaded by dodo (license 989) Tested by: tilghman
+ ........ ................
+
+2010-02-23 16:37 +0000 [r248398] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 248397 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010)
+ | 15 lines Merged revisions 248396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010)
+ | 9 lines fixes invite with replaces deadlock (closes issue
+ #16862) Reported by: pwalker Patches: replaces_deadlock_1.4
+ uploaded by dvossel (license 671) Tested by: pwalker, dvossel
+ ........ ................
+
+2010-02-19 19:07 +0000 [r248011] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_console.c, main/loader.c, /: Merged revisions
+ 228798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228798 |
+ tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14
+ lines Fix various problems detected with Valgrind. * chan_console
+ accessed pvts after deallocation. * The module loader did not
+ check usecount on shutdown, which led to chan_iax2 reading a
+ timer that was already unloaded. (closes issue #16062) Reported
+ by: alexanderheinz Patches: 20091109__issue16062.diff.txt
+ uploaded by tilghman (license 14) Tested by: tilghman ........
+
+2010-02-19 19:00 +0000 [r248005] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 248003 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r248003 | moy | 2010-02-19 13:38:34 -0500 (Fri, 19 Feb 2010) | 1
+ line mfcr2 issue 0016844 - Fix portability bit fields and make
+ mfcr2_immediate_accept work again, reported and patched by
+ korihor ........
+
+2010-02-19 18:45 +0000 [r248004] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 247914 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600
+ (Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600
+ (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
+ .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri,
+ 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing
+ consistent with other channel technologies. The processing of
+ DTMF tones on the receiving side of an ISDN channel is
+ inconsistent with the way it is handled in other channels,
+ especially DAHDI analog. This causes DTMF tones sent from an ISDN
+ phone to be doubled at the connected party. We are using the
+ following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes
+ Option one is necessary because the asterisk DSP DTMF detection
+ is better than mISDN's internal DSP. Not as many false positives.
+ Option two is necessary to transmit DTMF tones end to end when
+ mISDN channels are connected to SIP channels with out of band
+ DTMF for example. The symptom is that DTMF tones sent by an ISDN
+ phone are doubled on the way through asterisk when two mISDN
+ channels are connected with a Local channel in between or if it
+ is bridged to an analog channel. The doubling of DTMF tones is
+ because DTMF is passed inband to asterisk by the mISDN channel
+ and passed out of band once again after the release of the DTMF
+ tone. Passing it inband is wrong. Neither an analog channel nor
+ SIP channel passes DTMF inband if configured to inband DTMF.
+ Analog and SIP channels filter out the DTMF tones because they
+ use the voice frames returned by ast_dsp_process. But chan_misdn
+ passes the unfiltered input voice frames instead. To overcome one
+ aspect of the problem, the doubling of DTMF tones when two mISDN
+ channels are directly bridged, someone made an 'optimization',
+ where in that case the DTMF tone passed out-of-band to the peer
+ channel is not translated to an inband tone at the transmit side.
+ This optimization is bad because it does not work in general. For
+ example, analog channels or mISDN channels when bridged through
+ an intermediary local channel will generate DTMF tones from
+ out-of-band information. Also, of course, it must not be done
+ when there is no inband DTMF available. This patch fixes the
+ issue. Now chan_misdn will filter the received inband DTMF signal
+ the same as other channel types. Another change included: No need
+ to build an extra translation path because ast_process_dsp does
+ it if required. Patches: misdn-dtmf.patch JIRA ABE-2080
+ ................ ................
+
+2010-02-19 17:41 +0000 [r247916] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 247915 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247915 |
+ dvossel | 2010-02-19 11:40:26 -0600 (Fri, 19 Feb 2010) | 7 lines
+ handle_request_invite revise comment, fix coding guideline issues
+ I'm working with this code right now trying to analyze a
+ deadlock. This change is just to clean up a few things before I
+ make a more complex patch. ........
+
+2010-02-18 23:15 +0000 [r247792-247845] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_speech.c, /: Merged revisions 247841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247841 |
+ tilghman | 2010-02-18 17:13:46 -0600 (Thu, 18 Feb 2010) | 7 lines
+ Revert an errant part of a previous cleanup, to fix a memory
+ corruption issue. (closes issue #16368) Reported by: thirionjwf
+ Patches: res_speech.c.patch uploaded by thirionjwf (license 955)
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 247787 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247787 |
+ tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17
+ lines If the peer record is from realtime, it could be set to 0,
+ due to MySQL not representing NULL well in integer columns. NULL
+ means the value is not specified for the column, which normally
+ means the driver uses whatever is the default value. However, on
+ MySQL, placing a NULL in either a float or integer column results
+ in a retrieval of the 0 value. Hence, users get an errant error
+ on load. This patch suppresses that error and makes the value as
+ if it was not there. Note that this cannot be done in the
+ realtime driver, because the lack of difference between NULL and
+ 0 can only be intepreted correctly by the driver itself. If we
+ did it in the realtime driver, then it would be effectively
+ impossible to set any realtime field to 0, because it would act
+ as if the field were unspecified and possibly take on a different
+ value. (closes issue #16683) Reported by: wdoekes ........
+
+2010-02-18 21:25 +0000 [r247737-247776] David Vossel <dvossel@digium.com>
+
+ * /, bridges/bridge_softmix.c: Merged revisions 247770 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r247770 | dvossel | 2010-02-18 15:23:48 -0600 (Thu, 18 Feb 2010)
+ | 9 lines fixes confbridge crash when no timing module is loaded.
+ (closes issue #16471) Reported by: kjotte Patches: M16471.diff
+ uploaded by junky (license 177) Tested by: kjotte, junky ........
+
+ * apps/app_queue.c, /: Merged revisions 247736 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247736 |
+ dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines
+ fixes Queue with C option crash (closes issue #16475) Reported
+ by: okrief Patches: queue_crash.diff uploaded by dvossel (license
+ 671) ........
+
+2010-02-18 19:41 +0000 [r247653] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 247652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r247652 | mnicholson | 2010-02-18 13:39:37 -0600 (Thu, 18 Feb
+ 2010) | 13 lines Merged revisions 247651 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb
+ 2010) | 6 lines Copy the calling party's account code to the
+ called party if they don't already have one. (closes issue
+ #16331) Reported by: bluefox Tested by: mnicholson ........
+ ................
+
+2010-02-18 16:58 +0000 [r247506-247512] Leif Madsen <lmadsen@digium.com>
+
+ * README-SERIOUSLY.bestpractices.txt: Merged revisions 247509 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r247509 | lmadsen | 2010-02-18 11:54:43 -0500
+ (Thu, 18 Feb 2010) | 9 lines Merged revisions 247508 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18
+ Feb 2010) | 1 line Add additional link to best practices document
+ per jsmith. ........ ................
+
+ * README-SERIOUSLY.bestpractices.txt (added): Merged revisions
+ 247503 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r247503 | lmadsen | 2010-02-18 11:41:04 -0500 (Thu, 18 Feb 2010)
+ | 18 lines Merged revisions 247502 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010)
+ | 10 lines Add best practices documentation. (issue #16808)
+ Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis
+ Tested by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/507/ ........ ................
+
+2010-02-18 04:21 +0000 [r247426] Russell Bryant <russell@digium.com>
+
+ * sounds/Makefile, Makefile, /: Merged revisions 247423 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r247423 | russell | 2010-02-17 22:20:11 -0600
+ (Wed, 17 Feb 2010) | 17 lines Merged revisions 247422 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010)
+ | 10 lines Tweak argument handling for wget in the sounds
+ Makefile. 1) Fix the check to see if we are using wget to not be
+ full of fail. The configure script populates this variable with
+ the absolute path to wget if it is found, so it didn't work. 2)
+ Allow some extra arguments to be passed in for wget. This is just
+ a simple change to allow our Bamboo build script to tell wget to
+ be quiet and not fill up our logs with download status output.
+ ........ ................
+
+2010-02-17 21:32 +0000 [r246989-247337] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/strings.h, main/strings.c, /: Merged revisions
+ 247335 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247335 |
+ mmichelson | 2010-02-17 15:22:40 -0600 (Wed, 17 Feb 2010) | 20
+ lines Fix two problems in ast_str functions found while writing a
+ unit test. 1. The documentation for ast_str_set and
+ ast_str_append state that the max_len parameter may be -1 in
+ order to limit the size of the ast_str to its current allocated
+ size. The problem was that the max_len parameter in all cases was
+ a size_t, which is unsigned. Thus a -1 was interpreted as
+ UINT_MAX instead of -1. Changing the max_len parameter to be
+ ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an
+ off-by-one error in the case where we attempted to write a string
+ larger than the current allotted size to a string when -1 was
+ passed as the max_len parameter. When trying to write more than
+ the allotted size, the ast_str's __AST_STR_USED was set to 1
+ higher than it should have been. Thanks to Tilghman for quickly
+ spotting the offending line of code. Oh, and the unit test that I
+ referenced in the top line of this commit will be added to
+ reviewboard shortly. Sit tight... ........
+
+ * apps/app_queue.c, /: Merged revisions 247169 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r247169 | mmichelson | 2010-02-17 10:24:54 -0600 (Wed, 17 Feb
+ 2010) | 9 lines Merged revisions 247168 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb
+ 2010) | 3 lines Make sure that when autofill is disabled that
+ callers not in the front of the queue cannot place calls.
+ ........ ................
+
+ * main/strings.c, /: Merged revisions 247076 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247076 |
+ mmichelson | 2010-02-16 17:44:33 -0600 (Tue, 16 Feb 2010) | 12
+ lines Add va_end calls to __ast_str_helper. According to the man
+ page for stdarg(3), "Each invocation of va_copy() must be matched
+ by a corresponding invocation of va_end() in the same function."
+ There were several cases in __ast_str_helper where va_copy was
+ not matched with a corresponding call to va_end. ........
+
+ * include/asterisk/strings.h, /: Merged revisions 246985 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r246985 | mmichelson | 2010-02-16 15:15:38 -0600 (Tue,
+ 16 Feb 2010) | 3 lines Add some clarifying documentation to the
+ ast_str_set and ast_str_append functions. ........
+
+2010-02-16 21:03 +0000 [r246900-246982] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 246980 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r246980 |
+ dvossel | 2010-02-16 14:54:48 -0600 (Tue, 16 Feb 2010) | 8 lines
+ warning message if openssl support is missing while attempting
+ tls connection (closes issue #16673) Reported by: michaesc
+ Patches: tls_error_msg.diff uploaded by dvossel (license 671)
+ ........
+
+ * main/channel.c: fixes merge error with Monitor calculation fix
+
+ * main/channel.c, /: Merged revisions 246899 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r246899 |
+ dvossel | 2010-02-16 11:07:41 -0600 (Tue, 16 Feb 2010) | 16 lines
+ fixes sample rate conversion issue with Monitor application When
+ using ast_seekstream with the read/write streams of a monitor,
+ the number of samples we are seeking must be of the same rate as
+ the stream or the jump calculation will be incorrect. This patch
+ adds logic to correctly convert the number of samples to jump to
+ the sample rate the read/write stream is using. For example, if
+ the call is G722 (16khz) and the read/write stream is recording a
+ 8khz wav, seeking 320 samples of 16khz audio is not the same as
+ seeking 320 samples of 8khz audio when performing the
+ ast_seekstream on the stream. ABE-2044 ........
+
+2010-02-15 23:45 +0000 [r246713] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, /: Merged revisions 246710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r246710 | tilghman | 2010-02-15 17:43:28 -0600 (Mon, 15 Feb 2010)
+ | 12 lines Merged revisions 246709 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010)
+ | 5 lines Make the menuselect instructions correct by allowing
+ 'make menuselect' to actually solve dependency problems.
+ (Previously, it would fail out again with the same message about
+ running 'make menuselect', which was NOT at all helpful.)
+ ........ ................
+
+2010-02-12 23:33 +0000 [r246547] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 246546 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r246546 | dvossel | 2010-02-12 17:32:33 -0600 (Fri, 12 Feb 2010)
+ | 21 lines Merged revisions 246545 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010)
+ | 16 lines lock channel during datastore removal On channel
+ destruction the channel's datastores are removed and destroyed.
+ Since there are public API calls to find and remove datastores on
+ a channel, a lock should be held whenever datastores are removed
+ and destroyed. This resolves a crash caused by a race condition
+ in app_chanspy.c. (closes issue #16678) Reported by:
+ tim_ringenbach Patches: datastore_destroy_race.diff uploaded by
+ tim ringenbach (license 540) Tested by: dvossel ........
+ ................
+
+2010-02-12 19:08 +0000 [r246464] Jason Parker <jparker@digium.com>
+
+ * main/channel.c: Fix some silly formatting that made my head hurt.
+
+2010-02-10 21:28 +0000 [r246199-246207] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 246204 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r246204 | tilghman | 2010-02-10 15:24:10 -0600 (Wed, 10 Feb 2010)
+ | 2 lines Fussy compiler on another machine... ........
+
+ * /, funcs/func_strings.c: Merged revisions 246200 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r246200 | tilghman | 2010-02-10 15:19:35 -0600 (Wed, 10 Feb 2010)
+ | 2 lines Fix weird issue with unit tests on optimized build -
+ turned out to be a signing issue. ........
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ res/res_agi.c: Merged revisions 246030 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r246030 |
+ tilghman | 2010-02-10 10:01:28 -0600 (Wed, 10 Feb 2010) | 12
+ lines Solaris doesn't like outputting a NULL to a %s in format
+ strings. Detect all platforms that don't like that, either, and
+ ensure that when documentation is missing, we pass a non-NULL
+ pointer when outputting the corresponding documentation. (closes
+ issue #16689) Reported by: bklang Patches:
+ 20100209__issue16689__with_tests.diff.txt uploaded by tilghman
+ (license 14) Review: https://reviewboard.asterisk.org/r/497/
+ ........
+
+2010-02-10 17:51 +0000 [r246117] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 246116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r246116 | dvossel | 2010-02-10 11:49:34 -0600 (Wed, 10 Feb 2010)
+ | 14 lines Merged revisions 246115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010)
+ | 8 lines fixes random deadlock in app_queue with use_weight
+ during reload (closes issue #16677) Reported by: tim_ringenbach
+ Patches: app_queue_use_weight_deadlock.diff uploaded by tim
+ ringenbach (license 540) ........ ................
+
+2010-02-10 16:58 +0000 [r246073] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 246070 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r246070 | jpeeler | 2010-02-10 10:47:37 -0600 (Wed, 10 Feb 2010)
+ | 22 lines Change channel state on local channels for
+ busy,answer,ring. Previously local channels channel state never
+ changed. This became problematic when the state of the other side
+ of the local channel was lost, for example during a masquerade.
+ Changing the state of the local channel allows for the scenario
+ to be detected when the channel state is set to ringing, but the
+ peer isn't ringing. The specific problem scenario is described in
+ 164201. Although this was noted on one of the issues, here is the
+ tested dialplan verified to work: exten =>
+ 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten =>
+ *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
+ exten => *9700,n,wait(3) ;3 works, 1 did not exten =>
+ *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did
+ not exten =>
+ 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes
+ issue #14992) Reported by: davidw ........
+
+2010-02-10 15:38 +0000 [r245948-246025] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 246022 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r246022 | tilghman | 2010-02-10 09:36:57 -0600 (Wed, 10 Feb 2010)
+ | 2 lines Enable warnings on atypical conditions for the FILTER
+ function (suggested by mmichelson on the -dev list). ........
+
+ * configs/extensions.conf.sample, /, funcs/func_strings.c: Merged
+ revisions 245945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r245945 | tilghman | 2010-02-10 08:06:12 -0600 (Wed, 10 Feb 2010)
+ | 9 lines Merged revisions 245944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010)
+ | 2 lines Include examples of FILTER usage in extension patterns
+ where a "." may be a risk. ........ ................
+
+2010-02-09 23:11 +0000 [r245794] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 245793 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r245793 | dvossel | 2010-02-09 17:07:17 -0600
+ (Tue, 09 Feb 2010) | 18 lines Merged revisions 245792 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010)
+ | 12 lines Fixes iaxs and iaxsl size off by one issue. 2^15 =
+ 32768 which is the maximum allowed iax2 callnumber. Creating the
+ iaxs and iaxsl array of size 32768 means the maximum callnumber
+ is actually out of bounds. This causes a nasty crash. (closes
+ issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded
+ by dvossel (license 671) ........ ................
+
+2010-02-09 18:09 +0000 [r245732] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 245729 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r245729 |
+ tilghman | 2010-02-09 12:06:30 -0600 (Tue, 09 Feb 2010) | 8 lines
+ Ensure frames are only freed once. (closes issue #16361) Reported
+ by: vlad Patches: 20100208__issue16361.diff.txt uploaded by
+ tilghman (license 14) Tested by: kenny, bloodoff, misaksen
+ ........
+
+2010-02-09 17:43 +0000 [r245728] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 245727 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r245727 |
+ mnicholson | 2010-02-09 11:40:04 -0600 (Tue, 09 Feb 2010) | 2
+ lines This commit removes an extra newline in T.38 generated SDP
+ packets. This bug was caused by the fix introduced in r243860.
+ (closes issue #16766) Reported by: raivisr Patches:
+ t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: raivisr ........
+
+2010-02-09 16:26 +0000 [r245683] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 245680 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r245680 |
+ kpfleming | 2010-02-09 10:24:52 -0600 (Tue, 09 Feb 2010) | 8
+ lines Don't offer MMR or JBIG transcoding during T.38
+ negotiation. After further discussion with Steve Underwood, we
+ should not (yet) be offering to receive MMR or JBIG transcoded
+ streams from T.38 endpoints. A future spandsp release will
+ support those features, and then they can be enabled during
+ negotiation ........
+
+2010-02-08 23:47 +0000 [r245626] Russell Bryant <russell@digium.com>
+
+ * /, main/event.c: Merged revisions 245624 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r245624 |
+ russell | 2010-02-08 17:43:00 -0600 (Mon, 08 Feb 2010) | 5 lines
+ Fix return value of get_ie_str() and get_ie_str_hash() for
+ non-existent IE. I found this bug while developing a unit test
+ for event allocation. Testing is awesome. ........
+
+2010-02-08 22:46 +0000 [r245581] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/Makefile, /, main/Makefile: Merged revisions 245578 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r245578 | tilghman | 2010-02-08 16:31:40 -0600 (Mon, 08
+ Feb 2010) | 12 lines Actually use _ASTLDFLAGS in the main/ and
+ channels/ Makefiles. They were previously passed correctly, but
+ they simply weren't used. This caused issues with various
+ platforms whose builds needed to pass special linker flags via
+ the configure script. (closes issue #16596) Reported by:
+ pprindeville Patches: asterisk-1.6-astldflags.patch uploaded by
+ pprindeville (license 347) Tested by: tilghman ........
+
+2010-02-08 20:43 +0000 [r245500] Jason Parker <jparker@digium.com>
+
+ * main/ast_expr2.fl, /, main/ast_expr2f.c: Merged revisions 245497
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r245497 | qwell | 2010-02-08 14:41:05 -0600
+ (Mon, 08 Feb 2010) | 11 lines Merged revisions 245496 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) |
+ 4 lines Remove reference of documentation in source directory.
+ People don't always build Asterisk from source (distro packages,
+ anybody?). ........ ................
+
+2010-02-05 19:27 +0000 [r245097] Jeff Peeler <jpeeler@digium.com>
+
+ * contrib/firmware (removed), /, LICENSE: Merged revisions 245090
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r245090 | jpeeler | 2010-02-05 13:26:22 -0600
+ (Fri, 05 Feb 2010) | 11 lines Merged revisions 245044 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb
+ 2010) | 5 lines Remove contrib/firmware directory as it is empty
+ Remove explicit license for IAXy firmware as it is no longer
+ included in the tree ........ ................
+
+2010-02-05 17:10 +0000 [r244930] Sean Bright <sean@malleable.com>
+
+ * main/asterisk.c, /: Merged revisions 244927 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r244927 | seanbright | 2010-02-05 12:05:32 -0500 (Fri, 05 Feb
+ 2010) | 9 lines Merged revisions 244926 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb
+ 2010) | 1 line Update main copyright date. ........
+ ................
+
+2010-02-03 19:28 +0000 [r244555] Mark Michelson <mmichelson@digium.com>
+
+ * main/sched.c, /: Merged revisions 244547 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r244547 |
+ mmichelson | 2010-02-03 13:26:53 -0600 (Wed, 03 Feb 2010) | 3
+ lines Initialize counters in ast_sched_report so that resulting
+ data is not bogus. ........
+
+2010-02-03 18:47 +0000 [r244508] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /, main/ast_expr2f.c: Merged revisions
+ 244505 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r244505 |
+ tilghman | 2010-02-03 12:34:29 -0600 (Wed, 03 Feb 2010) | 8 lines
+ The chanvar= setting should inherit the entire list of variables,
+ not just the first one. (closes issue #16359) Reported by: raarts
+ Patches: dahdi-setvars.diff uploaded by raarts (license 937)
+ Tested by: raarts ........
+
+2010-02-02 22:29 +0000 [r244445] David Vossel <dvossel@digium.com>
+
+ * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
+ Merged revisions 244443 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r244443 |
+ dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines
+ fixes crash during T.38 negotiation caused by invalid or missing
+ FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported
+ by: krn (closes issue #16724) Reported by: barthpbx (closes issue
+ #16517) Reported by: bklang (closes issue #16485) Reported by:
+ elsto ........
+
+2010-02-02 20:35 +0000 [r244395] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 244393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r244393 |
+ tilghman | 2010-02-02 14:32:29 -0600 (Tue, 02 Feb 2010) | 18
+ lines Properly respect GOSUB_RESULT as to what to do with the
+ master channel. Previously, we would parse GOSUB_RESULT, but not
+ actually do anything with it. (closes issue #16686) Reported by:
+ bklang Patches: app_dial-respect-gosub_result.patch uploaded by
+ bklang (license 919) (with modifications) ........
+
+2010-02-02 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.2
+
+ * AST-2010-001: An attacker attempting to negotiate T.38 over SIP can
+ remotely crash Asterisk by modifying the FaxMaxDatagram field of
+ the SDP to contain either a negative or exceptionally large value.
+ The same crash occurs when the FaxMaxDatagram field is omitted from
+ the SDP as well.
+
+2010-01-14 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.1
+
+2010-01-08 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.1-rc1
+
+2010-01-07 21:17 +0000 [r238499] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_console.c, channels/chan_oss.c, main/poll.c,
+ channels/chan_usbradio.c, include/asterisk/utils.h, /,
+ channels/chan_sip.c, channels/chan_alsa.c: Merged revisions
+ 209400 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 |
+ kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3
+ lines Define side-effect-safe MIN and MAX macros and remove
+ duplicate definitions from various files. (closes issue #16251)
+ Reported by: asgaroth ........
+
+2010-01-07 20:17 +0000 [r238362-238416] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 238412 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r238412 | dvossel | 2010-01-07 14:15:27 -0600
+ (Thu, 07 Jan 2010) | 16 lines Merged revisions 238411 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010)
+ | 10 lines fixes crash in "scheduled_destroy" in chan_iax A
+ signed short was used to represent a callnumber. This is makes it
+ possible to attempt to access the iaxs array with a negative
+ index. (closes issue #16565) Reported by: jensvb ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 238405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r238405 |
+ dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines
+ Change in sip show channels display format allowing more digits
+ for CID (closes issue #16459) Reported by: Rzadzins Patches:
+ chan_sip_longer_cid.patch uploaded by Rzadzins (license 953)
+ ........
+
+ * apps/app_queue.c, /: Merged revisions 238361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r238361 |
+ dvossel | 2010-01-07 12:58:23 -0600 (Thu, 07 Jan 2010) | 8 lines
+ cli 'queue show' formatting fix. queue name was truncated over 12
+ characters (closes issue #16078) Reported by: RoadKill Patches:
+ quequename_limit.patch uploaded by ppyy (license 906) Tested by:
+ dvossel ........
+
+2010-01-07 09:49 +0000 [r238349] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * configs/sip.conf.sample, /: Merged revisions 238313 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r238313 | tzafrir | 2010-01-07 11:14:57 +0200 (ה', 07 ינו 2010) |
+ 2 lines Document the usefulness of explicit udp:// in the
+ register string ........
+
+2010-01-06 21:48 +0000 [r238234] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_cdr.c: Merged revisions 238231 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r238231 | tilghman | 2010-01-06 15:45:17 -0600 (Wed, 06 Jan 2010)
+ | 11 lines Merged revisions 238230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010)
+ | 4 lines Revise documentation on disposition values to the
+ actual values used. (closes issue #16289) Reported by: wdoekes
+ ........ ................
+
+2010-01-06 20:40 +0000 [r238137-238185] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 238181 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r238181 |
+ jpeeler | 2010-01-06 14:37:18 -0600 (Wed, 06 Jan 2010) | 8 lines
+ Fix misreverting from 177158. (closes issue #15725) Reported by:
+ shanermn Patches: v1-15725.patch uploaded by dimas (license 88)
+ Tested by: shanermn ........
+
+ * /, main/features.c: Merged revisions 238134 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r238134 |
+ jpeeler | 2010-01-06 13:05:06 -0600 (Wed, 06 Jan 2010) | 10 lines
+ Fix channel name comparison for bridge application. The channel
+ name comparison was not comparing the whole string and therefore
+ if one channel name was a substring of the other, the bridge
+ would fail. (closes issue #16528) Reported by: telecos82 Patches:
+ res_features_r236843.diff uploaded by telecos82 (license 687)
+ ........
+
+2010-01-06 15:22 +0000 [r238013] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_mp3.c: Merged revisions 238010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r238010 | russell | 2010-01-06 09:19:10 -0600 (Wed, 06 Jan 2010)
+ | 14 lines Merged revisions 238009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010)
+ | 7 lines Resolve a crash due to an ast_frame not being fully
+ initialized. (closes issue #16531) Reported by: john8675309
+ (closes SWP-615) ........ ................
+
+2010-01-06 06:54 +0000 [r237969] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 237968 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237968 |
+ tilghman | 2010-01-06 00:53:23 -0600 (Wed, 06 Jan 2010) | 2 lines
+ Whoa, duplicate setting (dead code). ........
+
+2010-01-05 23:10 +0000 [r237924] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_test.c: Add a wait to ensure TestServer thinks it has
+ finished sending the final digit. This was previously committed
+ to 1.4, 1.6.0, 1.6.1, and trunk just after 1.6.2 was created (and
+ missed). 1.6.2 also needs this patch to resolve the bug. (closes
+ issue #16550) Reported by: opticron Patches: apptest.diff
+ uploaded by opticron (license 267)
+
+2010-01-05 23:09 +0000 [r237840-237921] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 237920 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237920 |
+ dvossel | 2010-01-05 17:08:50 -0600 (Tue, 05 Jan 2010) | 16 lines
+ fixes holdtime playback issue in app_queue When reporting hold
+ time, the number of seconds should be mod 60. Otherwise audio
+ playback could be something like "2 minutes 123 seconds" rather
+ than "2 minutes 3 seconds". Also, the "minute" sound file is
+ missing, so for the moment until that file can be created the
+ "minutes" file is used instead. (closes issue #16168) Reported
+ by: nickilo Patches: patch-unified-trunk-rev-222176 uploaded by
+ nickilo (license ) Tested by: nickilo, wonderg ........
+
+ * main/pbx.c, /: Merged revisions 237839 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237839 |
+ dvossel | 2010-01-05 13:29:47 -0600 (Tue, 05 Jan 2010) | 19 lines
+ fixes subscriptions being lost after 'module reload' During a
+ module reload if multiple extension configs are present, such as
+ both extensions.conf and extensions.ael, watchers for one
+ config's hints will be lost during the merging of the other
+ config. This happens because hint watchers are only preserved for
+ the current config being merged. The old context list is
+ destroyed after the merging takes place, meaning any watchers
+ that were not perserved will be removed. Now all hints are
+ preserved during merging regardless of what config file is being
+ merged. These hints are only restored if they are present within
+ the new context list. (closes issue #16093) Reported by: jlaroff
+ ........
+
+2010-01-05 17:25 +0000 [r237743] Russell Bryant <russell@digium.com>
+
+ * /, main/utils.c: Merged revisions 237699 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237699 | russell | 2010-01-05 11:16:01 -0600 (Tue, 05 Jan 2010)
+ | 14 lines Merged revisions 237697 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010)
+ | 7 lines Change a NOTICE log message to DEBUG where it belongs.
+ (closes issue #16479) Reported by: alexrecarey (closes SWP-577)
+ ........ ................
+
+2010-01-05 16:09 +0000 [r237657] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 237656 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r237656 | mvanbaak | 2010-01-05 17:08:12 +0100 (Tue, 05 Jan 2010)
+ | 6 lines Make CLI command 'mixmonitor start|stop <channel> work
+ again. (closes issue #16534) Reported by: jlaguilar Fix as
+ suggested by jlaguilar in the bugreport ........
+
+2010-01-04 21:52 +0000 [r237409-237577] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c: Merged revisions 237574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237574 | tilghman | 2010-01-04 15:48:20 -0600 (Mon, 04 Jan 2010)
+ | 13 lines Merged revisions 237573 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010)
+ | 6 lines Bounds checking for input string (closes issue #16407)
+ Reported by: qwell Patches: 20100104__issue16407.diff.txt
+ uploaded by tilghman (license 14) ........ ................
+
+ * main/pbx.c, /: Merged revisions 237494 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237494 | tilghman | 2010-01-04 14:59:01 -0600 (Mon, 04 Jan 2010)
+ | 15 lines Merged revisions 237493 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010)
+ | 8 lines Regression in issue #15421 - Pattern matching (closes
+ issue #16482) Reported by: wdoekes Patches:
+ astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
+ 20091223__issue16482.diff.txt uploaded by tilghman (license 14)
+ Tested by: wdoekes, tilghman ........ ................
+
+ * main/config.c, /: Merged revisions 237414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237414 |
+ tilghman | 2010-01-04 13:03:20 -0600 (Mon, 04 Jan 2010) | 2 lines
+ Oops, didn't compile (thanks, kpfleming) ........
+
+ * main/config.c, /: Merged revisions 237410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237410 |
+ tilghman | 2010-01-04 12:42:10 -0600 (Mon, 04 Jan 2010) | 7 lines
+ Further reduce the encoded blank values back to blank in the
+ realtime API. (closes issue #16533) Reported by: sergee Patches:
+ 200100104__issue16533.diff.txt uploaded by tilghman (license 14)
+ Tested by: sergee ........
+
+ * main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged
+ revisions 237406 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010)
+ | 23 lines Merged revisions 237405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010)
+ | 16 lines Add a flag to disable the Background behavior, for AGI
+ users. This is in a section of code that relates to two other
+ issues, namely issue #14011 and issue #14940), one of which was
+ the behavior of Background when called with a context argument
+ that matched the current context. This fix broke FreePBX,
+ however, in a post-Dial situation. Needless to say, this is an
+ extremely difficult collision of several different issues. While
+ the use of an exception flag is ugly, fixing all of the issues
+ linked is rather difficult (although if someone would like to
+ propose a better solution, we're happy to entertain that
+ suggestion). (closes issue #16434) Reported by: rickead2000
+ Patches: 20091217__issue16434.diff.txt uploaded by tilghman
+ (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by
+ tilghman (license 14) Tested by: rickead2000 ........
+ ................
+
+2010-01-04 16:50 +0000 [r237328] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 237327 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237327 |
+ dvossel | 2010-01-04 10:39:11 -0600 (Mon, 04 Jan 2010) | 10 lines
+ app_queue segfaults if realtime field uniqueid is NULL (closes
+ issue #16385) Reported by: haakon Patches: app_queue.c.patch
+ uploaded by haakon (license 880) app_queue.c.patch_v2 uploaded by
+ dvossel (license 671) Tested by: haakon ........
+
+2010-01-04 16:27 +0000 [r237326] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 237323 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237323 |
+ jpeeler | 2010-01-04 10:24:51 -0600 (Mon, 04 Jan 2010) | 5 lines
+ Fix timeout for AGI command speech recognize. (closes issue
+ #16297) Reported by: semond ........
+
+2010-01-04 16:21 +0000 [r237322] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 237319 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r237319 | tilghman | 2010-01-04 10:20:03 -0600
+ (Mon, 04 Jan 2010) | 10 lines Merged revisions 237318 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010)
+ | 3 lines It's also possible for the Local channel to directly
+ execute an Application. Reviewboard:
+ https://reviewboard.asterisk.org/r/452/ ........ ................
+
+2010-01-02 10:03 +0000 [r237139] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 237136 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10
+ lines Merged revisions 237135 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2
+ lines Release memory of the contact acl before unloading module
+ ........ ................
+
+2009-12-30 22:00 +0000 [r236985] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 236982 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r236982 | tilghman | 2009-12-30 15:59:18 -0600
+ (Wed, 30 Dec 2009) | 16 lines Merged revisions 236981 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009)
+ | 9 lines Don't queue frames to channels that have no means to
+ process them. (closes issue #15609) Reported by: aragon Patches:
+ 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by
+ tilghman (license 14) Tested by: aragon Review:
+ https://reviewboard.asterisk.org/r/452/ ........ ................
+
+2009-12-30 21:13 +0000 [r236899-236905] Jeff Peeler <jpeeler@digium.com>
+
+ * /, utils/ael_main.c: Merged revisions 236902 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r236902 |
+ jpeeler | 2009-12-30 15:09:28 -0600 (Wed, 30 Dec 2009) | 2 lines
+ One more LOW_MEMORY compile fix. ........
+
+ * main/cli.c, /: Merged revisions 236893 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r236893 |
+ jpeeler | 2009-12-30 14:34:41 -0600 (Wed, 30 Dec 2009) | 11 lines
+ Fix compiling with LOW_MEMORY. Modified handle_verbose to be
+ LOW_MEMORY aware. (closes issue #16381) Reported by:
+ michael_iedema Patches: ast_complete_source_filename.patch
+ uploaded by michael iedema (license 942) modified by me ........
+
+2009-12-30 17:56 +0000 [r236804-236850] Tilghman Lesher <tlesher@digium.com>
+
+ * /, cdr/cdr_adaptive_odbc.c: Merged revisions 236847 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r236847 | tilghman | 2009-12-30 11:53:29 -0600 (Wed, 30 Dec 2009)
+ | 4 lines When the field is blank, don't warn about the field
+ being unable to be coerced, just skip the column. (closes
+ http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html)
+ Reported by Nic Colledge on the -dev list, fixed by me. ........
+
+ * /, channels/chan_sip.c: Merged revisions 236802 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r236802 |
+ tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines
+ Shut down the SIP session timers more gracefully, in order to
+ prevent a possible crash. (closes issue #16452) Reported by:
+ corruptor Patches: 20091221__issue16452.diff.txt uploaded by
+ tilghman (license 14) Tested by: corruptor ........
+
+2009-12-28 22:13 +0000 [r236716] Jason Parker <jparker@digium.com>
+
+ * main/ast_expr2.c, /, main/ast_expr2.y: Merged revisions 236713
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r236713 | qwell | 2009-12-28 16:09:40 -0600 (Mon, 28 Dec
+ 2009) | 8 lines Allow "REMAINDER" to function properly in
+ expressions. (closes issue #16427) Reported by: wdoekes Patches:
+ ast16-reminder-remainder.patch uploaded by wdoekes (license 717)
+ Tested by: wdoekes ........
+
+2009-12-28 17:40 +0000 [r236670] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 236667 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r236667 | tilghman | 2009-12-28 11:37:46 -0600 (Mon, 28 Dec 2009)
+ | 4 lines Use recommended option, not deprecated option. (closes
+ issue #16515) Reported by: ManChicken ........
+
+2009-12-28 15:31 +0000 [r236513-236635] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/threadstorage.h, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 236613 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec
+ 2009) | 14 lines Merged revisions 236585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec
+ 2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT
+ requires extra braces. There was conditional code (based on build
+ platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that
+ was removed since it is fixed in newer versions of
+ Solaris/OpenSolaris, but I am still running into it on Solaris 10
+ x86 so add a configure-time check for it. ........
+ ................
+
+ * /, apps/app_meetme.c: Merged revisions 236510 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec
+ 2009) | 19 lines Merged revisions 236509 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec
+ 2009) | 12 lines Avoid a crash with large numbers of MeetMe
+ conferences. Similar to changes made to Queue(), when we have
+ large numbers of conferences in meetme.conf (1000s) and we use
+ alloca()/strdupa(), we can blow out the stack and crash, so
+ instead just use a single fixed buffer. (closes issue #16509)
+ Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded
+ by seanbright (license 71) Tested by: seanbright ........
+ ................
+
+2009-12-27 18:22 +0000 [r236437] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 236434 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r236434 | tilghman | 2009-12-27 12:20:53 -0600
+ (Sun, 27 Dec 2009) | 9 lines Merged revisions 236433 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27
+ Dec 2009) | 2 lines Turn on colors in the daemon, since there's
+ many requests for it on Ubuntu. ........ ................
+
+2009-12-26 15:32 +0000 [r236361] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 236358 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236358 | kpfleming | 2009-12-26 09:27:44 -0600 (Sat, 26 Dec
+ 2009) | 9 lines Merged revisions 236357 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec
+ 2009) | 1 line update to latest releases with zero uid/gid
+ ........ ................
+
+2009-12-23 18:27 +0000 [r236189-236303] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, /: Merged revisions 236300 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r236300 |
+ tilghman | 2009-12-23 12:25:27 -0600 (Wed, 23 Dec 2009) | 7 lines
+ AGI may be invoked from outside the dialplan (closes issue
+ #16510) Reported by: atis Patches: 20091223__issue16510.diff.txt
+ uploaded by tilghman (license 14) Tested by: atis ........
+
+ * /, res/res_agi.c: Merged revisions 236186 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236186 | tilghman | 2009-12-22 21:07:48 -0600 (Tue, 22 Dec 2009)
+ | 11 lines Merged revisions 236184 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009)
+ | 4 lines If EXEC only gets a single argument, don't crash when
+ the second is used. (closes issue #16504) Reported by: bklang
+ ........ ................
+
+2009-12-22 17:04 +0000 [r236064] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 236063 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009)
+ | 18 lines Merged revisions 236062 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009)
+ | 11 lines fixes issue with p->method incorrectly set to ACK It
+ is possible for a second ACK to come in for a retransmitted
+ message. If an ack does not match an unacked message in our
+ queue, restore the previous p->method as this ACK is completely
+ ignored. (closes issue #16295) Reported by: omolenkamp Patches:
+ issue16295_v2.diff uploaded by dvossel (license 671) ........
+ ................
+
+2009-12-21 19:58 +0000 [r235944] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 235941 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r235941 | jpeeler | 2009-12-21 13:54:20 -0600 (Mon, 21 Dec 2009)
+ | 20 lines Merged revisions 235940 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009)
+ | 13 lines Change Monitor to not assume file to write to does not
+ contain pathing. 227944 changed the fname_base argument to always
+ append the configured monitor path. This change was necessary to
+ properly compare files for uniqueness. If a full path is given
+ though, nothing needs to be appended and that is handled
+ correctly now. (closes issue #16377) (closes issue #16376)
+ Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch
+ uploaded by dant (license 670) ........ ................
+
+2009-12-21 17:11 +0000 [r235826] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/features.c: Merged revisions 235822 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r235822 | tilghman | 2009-12-21 11:00:46 -0600 (Mon, 21 Dec 2009)
+ | 15 lines Merged revisions 235821 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009)
+ | 8 lines Send parking lot announcement to the channel which
+ parked the call, not the park-ee. (closes issue #16234) Reported
+ by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded
+ by tilghman (license 14) 20091221__issue16234__1.4.diff.txt
+ uploaded by tilghman (license 14) Tested by: yeshuawatso ........
+ ................
+
+2009-12-20 08:58 +0000 [r235775] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c: restarts busydetector (if enabled) when DTMF is
+ received after call is bridged. (closes issue #16389) Reported
+ by: alecdavis Tested by: alecdavis Patch
+ dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585)
+
+2009-12-18 23:04 +0000 [r235665] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /, include/asterisk/cdr.h: Merged revisions
+ 235660 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009)
+ | 55 lines Merged revisions 235635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009)
+ | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is
+ simple in that it reorders the disposition defines so that the
+ fix for issue 12946 works properly (the default CDR disposition
+ was changed to AST_CDR_NOANSWER). Also, the
+ AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all
+ CDR records are written. The side effects of CDR changes are
+ scary, so I'm documenting the test cases performed to attempt to
+ catch any regressions. The following tests were all performed
+ using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls
+ B (busy) Hangup C Hangup A (Both SIP and features) A calls B A
+ blind transfers to C Hangup C (Both SIP and features) A calls B A
+ attended transfers to C Hangup C A calls B A attended transfers
+ to C (SIP) C blind transfers to A (features) Hangup A All of the
+ test scenario CDRs matched. The following tests were performed
+ just with the patch to ensure proper operation (with
+ unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten
+ =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w)
+ (closes issue #16180) Reported by: aatef Patches: bug16180.patch
+ uploaded by jpeeler (license 325) ........ ................
+
+2009-12-18 22:42 +0000 [r235576-235659] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 235656 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r235656 | tilghman | 2009-12-18 16:40:46 -0600
+ (Fri, 18 Dec 2009) | 9 lines Merged revisions 235652 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18
+ Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion
+ ........ ................
+
+ * /, configure, configure.ac: Merged revisions 235573 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r235573 | tilghman | 2009-12-18 15:19:43 -0600
+ (Fri, 18 Dec 2009) | 9 lines Merged revisions 235572 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18
+ Dec 2009) | 2 lines Point to the typical missing package, not the
+ cryptic "termcap support". ........ ................
+
+2009-12-17 23:22 +0000 [r235522] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 235521 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r235521 |
+ file | 2009-12-17 19:21:07 -0400 (Thu, 17 Dec 2009) | 3 lines
+ Remove some old code for going to the 'fax' extension when a T.38
+ switchover occurs. This would have already happened when we
+ detected the CNG tone so this was basically a noop. ........
+
+2009-12-17 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0
+
+2009-12-09 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-rc8
+
+2009-12-08 18:33 +0000 [r233731] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 233718 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r233718 | tilghman | 2009-12-08 12:22:44 -0600 (Tue, 08 Dec 2009)
+ | 8 lines Find another ref leak and change how we manage module
+ references. (closes issue #16388) Reported by: parisioa Patches:
+ 20091208__issue16388.diff.txt uploaded by tilghman (license 14)
+ Tested by: parisioa, tilghman Review:
+ https://reviewboard.asterisk.org/r/442/ ........
+
+2009-12-08 18:04 +0000 [r233694] Russell Bryant <russell@digium.com>
+
+ * formats/format_sln16.c, formats/format_wav_gsm.c,
+ formats/format_siren7.c, formats/format_ilbc.c,
+ formats/format_vox.c, formats/format_pcm.c,
+ formats/format_h263.c, formats/format_g723.c,
+ formats/format_h264.c, formats/format_siren14.c,
+ formats/format_jpeg.c, formats/format_g726.c,
+ formats/format_gsm.c, formats/format_g729.c, /,
+ formats/format_sln.c, formats/format_wav.c,
+ formats/format_ogg_vorbis.c: Merged revisions 233692 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r233692 | russell | 2009-12-08 12:00:16 -0600 (Tue, 08 Dec 2009)
+ | 16 lines Set a module load priority for format modules. A
+ recent change to app_voicemail made it such that the module now
+ assumes that all format modules are available while processing
+ voicemail configuration. However, when autoloading modules, it
+ was possible that app_voicemail was loaded before the format
+ modules. Since format modules don't depend on anything, set a
+ module load priority on them to ensure that they get loaded first
+ when autoloading. This fix applies to trunk, 1.6.1, and 1.6.2.
+ The fix for 1.4 and 1.6.0 will require a different approach since
+ the module load priority functionality is not present in the
+ module API. (issue #16412) Reported by: jiddings ........
+
+2009-12-08 07:41 +0000 [r233689] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Fixed compile error with OSP Toolkit 3.6.
+
+2009-12-07 23:54 +0000 [r233615] Atis Lezdins <atis@iq-labs.net>
+
+ * contrib/valgrind.supp, /: Merged revisions 233577 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r233577 | atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8
+ lines Fix compatibility with valgrind 3.3 and older. (noticed in
+ issue #16388) Reported by: parisioa Patches: valgrind.supp
+ uloaded by atis (license 242) Tested by: atis, parisioa ........
+
+2009-12-07 23:29 +0000 [r233473-233612] David Vossel <dvossel@digium.com>
+
+ * /, main/utils.c: Merged revisions 233611 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r233611 |
+ dvossel | 2009-12-07 17:28:51 -0600 (Mon, 07 Dec 2009) | 4 lines
+ fixes incorrect logic in ast_uri_encode issue #16299 ........
+
+ * /, channels/chan_sip.c: Merged revisions 233472 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009)
+ | 15 lines Merged revisions 233471 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009)
+ | 9 lines fixes missing Contact header angle brackets (closes
+ issue #16298) Reported by: mgernoth Patches:
+ reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested
+ by: dvossel ........ ................
+
+2009-12-07 16:16 +0000 [r233396] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 233394 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r233394 |
+ mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8
+ lines Do not reject SDP packets describing only non audio
+ streams. (closes issue #16387) Reported by: zalex1953 Patches:
+ media-level-c-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: mnicholson, zalex1953 ........
+
+2009-12-04 21:55 +0000 [r233281] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, /: Merged revisions 233280 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r233280 | dvossel | 2009-12-04 15:54:44 -0600
+ (Fri, 04 Dec 2009) | 14 lines Merged revisions 233279 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009)
+ | 7 lines clarify requirecalltoken option in iax.sample.conf
+ (closes issue #16223) Reported by: bklang Patches:
+ clarify-iax-requirecalltoken.patch uploaded by bklang (license
+ 919) ........ ................
+
+2009-12-04 21:07 +0000 [r233240] Matthias Nick <mnick@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 233093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r233093 |
+ mnick | 2009-12-04 11:15:47 -0600 (Fri, 04 Dec 2009) | 8 lines
+ Parse global variables or expressions in hint extensions Parse
+ global variables or expressions in hint extensions. Like: exten
+ => 400,hint,DAHDI/i2/${GLOBAL(var)} (closes issue #16166)
+ Reported by: rmudgett Tested by: mnick, rmudgett ........
+
+2009-12-04 17:36 +0000 [r233165] David Vossel <dvossel@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 233121 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r233121 | dvossel | 2009-12-04 11:22:31 -0600
+ (Fri, 04 Dec 2009) | 12 lines Merged revisions 233116 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009)
+ | 6 lines document and rename strip_control() in app_voicemail
+ (closes issue #16291) Reported by: wdoekes ........
+ ................
+
+2009-12-04 17:23 +0000 [r233130] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 233100 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r233100 | russell | 2009-12-04 11:18:22 -0600 (Fri, 04 Dec 2009)
+ | 14 lines Merged revisions 233092 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009)
+ | 7 lines Only do frame payload check for HOLD frames. This code
+ was added for helping to debug the source of invalid HOLD frames.
+ However, a side effect of this is that it will incorrectly report
+ errors for frames that have an integer payload. Make the check
+ for this block specific to the HOLD frame case. ........
+ ................
+
+2009-12-04 15:57 +0000 [r233049] Matthias Nick <mnick@digium.com>
+
+ * main/dsp.c, /: Merged revisions 233046 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r233046 | mnick | 2009-12-04 09:38:33 -0600 (Fri, 04 Dec 2009) |
+ 17 lines Merged revisions 233014 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) |
+ 11 lines Warning message gets displayed only once Added
+ additional field 'int display_inband_dtmf_warning', which when
+ set to '1' displays the warning ('Inband DTMF is not supported on
+ codec %s. Use RFC2833'), and when set to '0' doesn't display the
+ warning. Otherwise you would get hundreds of warnings every
+ second. (closes issue #15769) Reported by: falves11 Patches:
+ patch_15769_14.txt uploaded by mnick (license 874) Tested by:
+ mnick, falves11 ........ ................
+
+2009-12-03 21:03 +0000 [r232866] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 232854 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r232854 | tilghman | 2009-12-03 14:47:07 -0600
+ (Thu, 03 Dec 2009) | 15 lines Merged revisions 232820 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009)
+ | 8 lines Deprecate "cz" in favor of "cs". Also, change the use
+ of language codes so that language registers as a prefix, rather
+ than an exact match. (closes issue #16272) Reported by: patrol-cz
+ Patches: 20091203__issue16272.diff.txt uploaded by tilghman
+ (license 14) ........ ................
+
+2009-12-03 15:14 +0000 [r232813] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: Merged revisions 232587 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232587 |
+ diruggles | 2009-12-02 17:17:22 -0500 (Wed, 02 Dec 2009) | 12
+ lines Prevent double closing of FDs by EIVR This caused a problem
+ when asterisk was under heavy load and running both AGI and EIVR
+ applications. EIVR would close an FD at which point it would be
+ considered freed and be used by a new AGI instance the second
+ close would then close the FD now in use by AGI. (closes issue
+ #16305) Reported by: diLLec Tested by: thedavidfactor, diLLec
+ Review: https://reviewboard.asterisk.org/r/436/ ........
+
+2009-12-03 00:20 +0000 [r232675-232678] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: Oops, really remove it this time
+
+ * res/res_musiconhold.c, /: Recorded merge of revisions
+ 232660-232661 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232660 |
+ tilghman | 2009-12-02 18:08:55 -0600 (Wed, 02 Dec 2009) | 19
+ lines Fix multiple issues with musiconhold, which led to classes
+ not getting destroyed properly. * Classes are now tracked past
+ removal from the core container, and module removal is actively
+ prevented until all references are freed. * A hanging reference
+ stored in the channel has been removed. This could have caused a
+ mismatch and the music state not properly cleared, if two or more
+ reloads occurred between MOH being stopped and MOH being
+ restarted. * In certain circumstances, duplicate classes were
+ possible. * A race existed at reload time between a process being
+ killed and the thread responsible for reading from the related
+ pipe respawning that process. * Several reference counts have
+ also been corrected. At least one could have caused deleted
+ classes to stick around forever, consuming resources. This
+ originally manifested as MOH external processes that were not
+ killed at reload time. (closes issue #16279, closes issue #16207)
+ Reported by: parisioa, dcabot Patches:
+ 20091202__issue16279__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: parisioa, tilghman ........ r232661 | tilghman |
+ 2009-12-02 18:09:36 -0600 (Wed, 02 Dec 2009) | 2 lines Remove
+ debugging line ........
+
+2009-12-02 23:28 +0000 [r232658] David Vossel <dvossel@digium.com>
+
+ * CHANGES, /, UPGRADE.txt: Merged revisions 232657 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r232657 | dvossel | 2009-12-02 17:27:45 -0600 (Wed, 02 Dec 2009)
+ | 6 lines update CHANGES and UPGRADE.txt for early media behavior
+ change between 1.6.1 and 1.6.2 (closes issue #16212) Reported by:
+ miki ........
+
+2009-12-02 22:05 +0000 [r232579-232585] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c, /: Merged revisions 232582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r232582 | jpeeler | 2009-12-02 16:02:43 -0600 (Wed, 02 Dec 2009)
+ | 14 lines Merged revisions 232581 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009)
+ | 7 lines Send ack (response/message) after receiving manager
+ action userevent (closes issue #16264) Reported by: dimas
+ Patches: event-ack.patch uploaded by dimas (license 88) ........
+ ................
+
+ * main/manager.c, /: Merged revisions 232576 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232576 |
+ jpeeler | 2009-12-02 15:32:50 -0600 (Wed, 02 Dec 2009) | 8 lines
+ Make manager response to "Action: events" finish with empty line
+ (closes issue #16275) Reported by: vnovy Patches: manager.c.diff
+ uploaded by vnovy (license 922) ........
+
+2009-12-02 17:11 +0000 [r232359] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_amd.c: Merged revisions 232356 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r232356 | file | 2009-12-02 13:06:54 -0400 (Wed, 02 Dec 2009) |
+ 12 lines Merged revisions 232355 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5
+ lines Fix a bug where if you hung up very quickly after calling
+ AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG.
+ (closes issue #16239) Reported by: CGMChris ........
+ ................
+
+2009-12-02 17:01 +0000 [r232352] David Vossel <dvossel@digium.com>
+
+ * /, main/acl.c: Merged revisions 232351 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r232351 | dvossel | 2009-12-02 11:00:15 -0600 (Wed, 02 Dec 2009)
+ | 12 lines Merged revisions 232350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009)
+ | 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in
+ strace. (closes issue #16290) Reported by: wdoekes ........
+ ................
+
+2009-12-02 16:43 +0000 [r232348] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 232345 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232345 |
+ file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines Add
+ support for handling the 415 Unsupported media type response like
+ we do for a 488 Not acceptable here response. (closes issue
+ #16186) Reported by: atis Patches: sip_t38_response_415.patch
+ uploaded by atis (license 242) ........
+
+2009-12-02 15:43 +0000 [r232270] David Vossel <dvossel@digium.com>
+
+ * funcs/func_groupcount.c, /: Merged revisions 232269 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r232269 | dvossel | 2009-12-02 09:42:54 -0600
+ (Wed, 02 Dec 2009) | 15 lines Merged revisions 232268 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009)
+ | 9 lines fixes segfault in func_groupcount closes issue #16337)
+ Reported by: Parantido Patches: issue_16337.diff uploaded by
+ dvossel (license 671) Tested by: Parantido, dvossel ........
+ ................
+
+2009-12-02 14:55 +0000 [r232232] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 232230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232230 |
+ file | 2009-12-02 10:54:28 -0400 (Wed, 02 Dec 2009) | 5 lines Fix
+ a bug where a scheduled item ID would get retained on
+ registrations in a certain scenario causing code to execute
+ during reload that should not. (issue AST-263) ........
+
+2009-12-02 00:52 +0000 [r232094] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 232091 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r232091 | jpeeler | 2009-12-01 18:45:18 -0600
+ (Tue, 01 Dec 2009) | 17 lines Merged revisions 232090 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009)
+ | 10 lines Do not modify the gain settings on data calls. (The
+ digital flag actually represents a data call.) (closes issue
+ #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis ........
+ ................
+
+2009-12-01 23:40 +0000 [r232011-232015] Russell Bryant <russell@digium.com>
+
+ * /, funcs/func_lock.c: Merged revisions 232012 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232012 |
+ russell | 2009-12-01 17:38:34 -0600 (Tue, 01 Dec 2009) | 2 lines
+ Fix a build error on FreeBSD. ........
+
+ * /, main/file.c: Merged revisions 232008 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r232008 | russell | 2009-12-01 17:27:53 -0600 (Tue, 01 Dec 2009)
+ | 9 lines Merged revisions 232007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009)
+ | 2 lines Fix a warning pointed out by buildbot. ........
+ ................
+
+2009-12-01 22:03 +0000 [r231930] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 231927 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231927 | jpeeler | 2009-12-01 15:54:21 -0600 (Tue, 01 Dec 2009)
+ | 19 lines Merged revisions 231911 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009)
+ | 12 lines Fix crash with invalid frame data The crash was
+ happening as a result of a frame containing an invalid data
+ pointer, but was set with data length of zero. The few times the
+ issue was reproduced it _seemed_ that the frame was queued
+ properly, that is the data pointer was set to NULL. I never could
+ reproduce the crash so as a last resort the crash has been fixed,
+ but a check in __ast_read has been added to give as much
+ information about the source of problematic frames in the future.
+ (closes issue #16058) Reported by: atis ........ ................
+
+2009-12-01 21:21 +0000 [r231870] David Vossel <dvossel@digium.com>
+
+ * main/pbx.c, /: Merged revisions 231867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231867 | dvossel | 2009-12-01 15:20:19 -0600 (Tue, 01 Dec 2009)
+ | 9 lines Merged revisions 231853 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009)
+ | 3 lines WaitExten m option with no parameters generates frame
+ with zero datalen but non-null data ptr ........ ................
+
+2009-12-01 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-rc7
+
+2009-12-01 15:48 +0000 [r231743] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/file.c: Merged revisions 231741 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231741 | mnicholson | 2009-12-01 09:47:36 -0600 (Tue, 01 Dec
+ 2009) | 9 lines Merged revisions 231740 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec
+ 2009) | 2 lines Ignore unknown formats in ast_format_str_reduce()
+ and return an error if no know formats are found. ........
+ ................
+
+2009-11-30 21:59 +0000 [r231695-231696] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
+ Merged revisions 231692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 |
+ kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22
+ lines Another round of UDPTL stack fixes/improvements: 1) Allow
+ users of UDPTL stack to associate a character-string tag with a
+ UDPTL session, so that log/error/debug messages generated by the
+ UDPTL stack can be 'connected' to the endpoint that caused them
+ to be generated. 2) Improve comments (and process) of calculating
+ the far end's maximum IFP size when redundancy mode is in use for
+ error correction. 3) When an IFP larger than the calculated 'far
+ max IFP' size is presented for writing, truncate it rather than
+ putting in the buffer and allowing the buffer to overflow; this
+ will cause the ends to retrain to a lower bit rate that produces
+ IFPs of an appropriate size if possible, and if not possible, the
+ FAX transfer will fail completely. In these cases, it is due to
+ the one endpoint supplying a T38FaxMaxDatagram value that is
+ improperly calculated and is too low to be of use; we have
+ configuration options available to override this behavior. 4)
+ Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no
+ longer needed. ........
+
+ * pbx/pbx_config.c: Backport a tiny fix from trunk that makes GCC
+ 4.4.x happier.
+
+2009-11-30 21:36 +0000 [r231689] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_voicemail.c, include/asterisk/file.h, /, main/file.c,
+ main/app.c: Merged revisions 231688 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov
+ 2009) | 15 lines Merged revisions 231614 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov
+ 2009) | 8 lines Remove duplicate entries from voicemail format
+ lists. This prevents app_voicemail from entering an infinite loop
+ when the same format is specified twice in the format list.
+ (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson
+ Review: https://reviewboard.asterisk.org/r/429/ ........
+ ................
+
+2009-11-30 20:47 +0000 [r231605] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 231602 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231602 |
+ file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines
+ When receiving SDP that matches the version of the last one do
+ not treat it as a fatal error. (closes issue #16238) Reported by:
+ seandarcy ........
+
+2009-11-30 18:57 +0000 [r231505-231558] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 231556 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231556 |
+ dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines
+ app_queue crashes randomly, often during call-transfers This
+ patch adds a ref to the queue_ent object's parent call_queue in
+ queue_exec() so the call_queue won't be destroyed while the the
+ queue_ent still holds a pointer to it. (closes issue 0015686)
+ Tested by: dvossel, aragon ........
+
+ * main/rtp.c, /: Merged revisions 231491 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231491 | dvossel | 2009-11-30 11:28:28 -0600 (Mon, 30 Nov 2009)
+ | 17 lines Merged revisions 231441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009)
+ | 11 lines fixes crash caused by RTP comfort noise payload
+ greater than 24 bytes AST-2009-010 (closes issue #16242) Reported
+ by: amorsen Patches: issue16242.diff uploaded by oej (license
+ 306) Tested by: amorsen, oej, dvossel ........ ................
+
+2009-11-25 22:34 +0000 [r231302] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 231299 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231299 | tilghman | 2009-11-25 16:33:02 -0600 (Wed, 25 Nov 2009)
+ | 9 lines Merged revisions 231298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009)
+ | 2 lines After a frame duplication failure, unlock the channel
+ before returning. ........ ................
+
+2009-11-25 15:48 +0000 [r231191] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, pbx/pbx_lua.c: Merged revisions 231189 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231189 |
+ mnicholson | 2009-11-25 09:42:48 -0600 (Wed, 25 Nov 2009) | 4
+ lines Load pbx_lua with global symbols to allow linking with
+ other lua libraries. Found by Maxim Litnitskiy. ........
+
+2009-11-24 20:36 +0000 [r231136] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 231134 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231134 |
+ tilghman | 2009-11-24 14:31:28 -0600 (Tue, 24 Nov 2009) | 7 lines
+ Found a few places where queue refcounts were counted
+ incorrectly. Also add debug statements. (closes issue #15982,
+ closes issue #15984) Reported by: atis Patches:
+ 20091111__issue15982.diff.txt uploaded by tilghman (license 14)
+ Tested by: atis ........
+
+2009-11-24 18:54 +0000 [r231098] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 231095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231095 |
+ jpeeler | 2009-11-24 12:50:36 -0600 (Tue, 24 Nov 2009) | 11 lines
+ Fix erroneous hangup extension execution ast_spawn_extension
+ behaves differently from 1.4 in that hangups and extensions that
+ do not exist do not return an error, whereas in 1.6 it does. This
+ is now taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN
+ flag gets set properly. (closes issue #16106) Reported by:
+ ajohnson Tested by: ajohnson ........
+
+2009-11-23 15:48 +0000 [r230884] Joshua Colp <jcolp@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 230881 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 |
+ file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines
+ Change fax detection in chan_sip so it behaves as one would
+ expect. Internally the way T.38 is negotiated has changed and the
+ option no longer reflects a behavior that is valid. It will now
+ look for a CNG tone on received calls and if present send the
+ call to the 'fax' extension. It is then up to the application or
+ channel to request the switch over to T.38. ........
+
+2009-11-23 15:38 +0000 [r230796-230880] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 230877 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r230877 | kpfleming | 2009-11-23 09:34:16 -0600 (Mon, 23 Nov
+ 2009) | 9 lines Merged revisions 230839 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov
+ 2009) | 1 line Correct fix for issue #16268... the reporter's
+ original patch was very close to correct. ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 230773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r230773 | kpfleming | 2009-11-23 08:15:48 -0600 (Mon, 23 Nov
+ 2009) | 12 lines Merged revisions 230772 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov
+ 2009) | 5 lines Ensure that SDP parsing does not ignore the last
+ line of the SDP. (closes issue #16268) Reported by: sgimeno
+ ........ ................
+
+2009-11-20 22:36 +0000 [r230727] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 230726 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r230726 | dvossel | 2009-11-20 16:35:54 -0600 (Fri, 20 Nov 2009)
+ | 7 lines fixes iax2 show cache locking error, thanks alecdavis!
+ (closes issue #16094) Reported by: alecdavis Patches:
+ bug16094.diff.txt uploaded by alecdavis (license 585) Tested by:
+ alecdavis, dvossel ........
+
+2009-11-20 21:07 +0000 [r230629] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 230628 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r230628 | mnicholson | 2009-11-20 15:01:10 -0600 (Fri, 20 Nov
+ 2009) | 15 lines Merged revisions 230627 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov
+ 2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR
+ if it exists. This is necessary for the recordagentcalls option
+ in chan_agent to store the recorded file name in the bridge CDR.
+ (closes issue #14590) Reported by: msetim Patches:
+ queue_agent_userfield.patch uploaded by Laureano (license 265)
+ Tested by: Laureano, mnicholson ........ ................
+
+2009-11-20 17:31 +0000 [r230510-230585] David Vossel <dvossel@digium.com>
+
+ * main/audiohook.c, /, include/asterisk/audiohook.h: Merged
+ revisions 230583 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r230583 |
+ dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines
+ audiohook signal trigger on every status change (issue #14618)
+ Review: https://reviewboard.asterisk.org/r/434/ ........
+
+ * apps/app_mixmonitor.c, /: Merged revisions 230509 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r230509 | dvossel | 2009-11-19 15:26:21 -0600
+ (Thu, 19 Nov 2009) | 17 lines Merged revisions 230508 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009)
+ | 10 lines fixes MixMonitor thread not exiting when
+ StopMixMonitor is used (closes issue #16152) Reported by: AlexMS
+ Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license
+ 671) Tested by: dvossel, AlexMS Review:
+ https://reviewboard.asterisk.org/r/424/ ........ ................
+
+2009-11-16 16:41 +0000 [r230250-230384] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 230381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r230381 |
+ kpfleming | 2009-11-16 10:40:25 -0600 (Mon, 16 Nov 2009) | 1 line
+ Fix another buglet in T.38 session teardown at the end of FAX
+ sessions. ........
+
+ * /, apps/app_fax.c: Merged revisions 230343 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r230343 |
+ kpfleming | 2009-11-16 06:51:59 -0600 (Mon, 16 Nov 2009) | 2
+ lines Ensure that only one end of a T.38 session initiates
+ teardown at completion. ........
+
+ * channels/chan_iax2.c, /: Merged revisions 230247 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r230247 | kpfleming | 2009-11-15 11:23:02 -0600
+ (Sun, 15 Nov 2009) | 12 lines Merged revisions 230246 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15 Nov
+ 2009) | 6 lines Correct mistaken option name in error message.
+ The configuration option for allowing hosts to make
+ non-token-based calls is 'calltokenoptional', not
+ 'calltokenignore'. (reported on asterisk-users) ........
+ ................
+
+2009-11-13 22:01 +0000 [r229969-230148] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 230145 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r230145 | file | 2009-11-13 16:00:44 -0600 (Fri, 13 Nov 2009) |
+ 15 lines Merged revisions 230144 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8
+ lines Respect the maddr parameter in the Via header. (closes
+ issue #14446) Reported by: frawd Patches: via_maddr.patch
+ uploaded by frawd (license 610) Tested by: frawd ........
+ ................
+
+ * channels/chan_local.c, /: Merged revisions 230039 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r230039 | file | 2009-11-13 13:44:53 -0600 (Fri,
+ 13 Nov 2009) | 16 lines Merged revisions 230038 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov 2009) | 9
+ lines Fix a crash caused by two threads thinking they should both
+ free the chan_local private structure when only one should.
+ (closes issue #15314) Reported by: sroberts Patches:
+ Issue15314_Move_Nulling_owner.patch uploaded by davidw (license
+ 780) Tested by: davidw, lottc ........ ................
+
+ * configs/extensions.conf.sample, /, apps/app_chanisavail.c: Merged
+ revisions 229966 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229966 | file | 2009-11-13 11:20:26 -0600 (Fri, 13 Nov 2009) |
+ 13 lines Merged revisions 229965 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6
+ lines Document a limitation in the AVAILSTATUS variable from
+ ChanIsAvail and provide a workaround for it that does not change
+ existing behavior. (closes issue #14426) Reported by: macli
+ ........ ................
+
+2009-11-13 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-rc6
+
+2009-11-13 15:57 +0000 [r229915] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 229912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r229912 |
+ file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines Fix
+ T.38 negotiation regression introduced with the SDP parser
+ changes. ........
+
+2009-11-12 23:31 +0000 [r229752] Jason Parker <jparker@digium.com>
+
+ * channels/chan_oss.c, /: Merged revisions 229750 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r229750 |
+ qwell | 2009-11-12 17:30:10 -0600 (Thu, 12 Nov 2009) | 1 line Fix
+ mute toggling on OSS channels. ........
+
+2009-11-12 16:47 +0000 [r229671] David Vossel <dvossel@digium.com>
+
+ * funcs/func_audiohookinherit.c, /: Merged revisions 229670 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r229670 | dvossel | 2009-11-12 10:44:39 -0600
+ (Thu, 12 Nov 2009) | 12 lines Merged revisions 229669 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009)
+ | 6 lines fixes merging error, datastore was being freed in the
+ wrong function. (closes issue #16219) Reported by: aragon
+ ........ ................
+
+2009-11-11 20:49 +0000 [r229570] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: Merged revisions 229568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r229568 |
+ diruggles | 2009-11-11 15:47:06 -0500 (Wed, 11 Nov 2009) | 9
+ lines Remove non-functional feature from ExternalIVR
+ documentation Remove non-functional socket implementation of
+ ExternalIVR from documentation (closes issue #16225) Reported by:
+ thedavidfactor Patches: externalivr.txt.20091111.1542.patch
+ uploaded by thedavidfactor (license 903) ........
+
+2009-11-11 19:56 +0000 [r229492-229502] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c, /: Merged revisions 229499 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229499 | dbrooks | 2009-11-11 13:48:18 -0600 (Wed, 11 Nov 2009)
+ | 15 lines Merged revisions 229498 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009)
+ | 8 lines Solaris doesn't like NULL going to ast_log Solaris will
+ crash if NULL is passed to ast_log. This simple patch simply uses
+ S_OR to get around this. (closes issue #15392) Reported by:
+ yrashk ........ ................
+
+ * /, apps/app_softhangup.c: Merged revisions 229460 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r229460 | dbrooks | 2009-11-11 12:13:56 -0600 (Wed, 11 Nov 2009)
+ | 7 lines Flags not initialized in app_softhangup.c, causing
+ undefined behavior Trivial patch [kobaz] to initialize an
+ ast_flags = {0} (closes issue #16129) Reported by: kobaz ........
+
+2009-11-10 22:17 +0000 [r229366] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 229361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229361 | tilghman | 2009-11-10 16:14:22 -0600 (Tue, 10 Nov 2009)
+ | 19 lines Merged revisions 229360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009)
+ | 12 lines If two pattern classes start with the same digit and
+ have the same number of characters, they will compare equal. The
+ example given in the issue report is that of [234] and [246],
+ which have these characteristics, yet they are clearly not
+ equivalent. The code still uses these two characteristics, yet
+ when the two scores compare equal, an additional check will be
+ done to compare all characters within the class to verify
+ equality. (closes issue #15421) Reported by: jsmith Patches:
+ 20091109__issue15421__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: jsmith, thedavidfactor ........ ................
+
+2009-11-10 22:04 +0000 [r229359] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: Merged revisions 229356 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229356 | diruggles | 2009-11-10 17:01:50 -0500 (Tue, 10 Nov
+ 2009) | 16 lines Merged revisions 229355 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov
+ 2009) | 9 lines Fix ExternalIVR Documentation Remove
+ documentation for event that doesn't function (closes issue
+ #16220) Reported by: thedavidfactor Patches:
+ externalivr.txt.20091110.1622.patch uploaded by thedavidfactor
+ (license 903) ........ ................
+
+2009-11-10 21:33 +0000 [r229354] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, /: Merged revisions 229351 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r229351 |
+ tilghman | 2009-11-10 15:22:50 -0600 (Tue, 10 Nov 2009) | 7 lines
+ When GOSUB is invoked within an AGI, it may not exit correctly.
+ (closes issue #16216) Reported by: atis Patches:
+ 20091110__atis_work.diff.txt uploaded by tilghman (license 14)
+ Tested by: atis ........
+
+2009-11-10 20:09 +0000 [r229285] Joshua Colp <jcolp@digium.com>
+
+ * /, codecs/codec_g726.c: Merged revisions 229282 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229282 | file | 2009-11-10 16:06:13 -0400 (Tue, 10 Nov 2009) |
+ 15 lines Merged revisions 229281 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8
+ lines Remove broken support for direct transcoding between G.726
+ RFC3551 and G.726 AAL2. On some systems the translation core
+ would actually consider g726aal2 -> g726 -> signed linear to be a
+ quicker path then g726aal2 -> signed linear which exposed this
+ problem. (closes issue #15504) Reported by: globalnetinc ........
+ ................
+
+2009-11-10 17:52 +0000 [r229232] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 229168 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r229168 | dvossel | 2009-11-10 11:16:49 -0600
+ (Tue, 10 Nov 2009) | 15 lines Merged revisions 229167 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009)
+ | 9 lines don't crash on log message in solaris AST-2009-006
+ (closes issue #16206) Reported by: bklang Tested by: bklang
+ ........ ................
+
+2009-11-10 17:39 +0000 [r229231] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: Merged revisions 229228 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229228 | diruggles | 2009-11-10 12:33:47 -0500 (Tue, 10 Nov
+ 2009) | 18 lines Merged revisions 229191 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov
+ 2009) | 11 lines Document ExternalIVR event tag collision
+ ExternalIVR uses the D tag for two different event types. This
+ documents that behavior and how to differentiate between the two
+ cases. Also includes a minor spelling fix and clarification
+ (closes issue #16211) Reported by: thedavidfactor Patches:
+ externalivr.txt.20091109.1507.patch uploaded by thedavidfactor
+ (license 903) ........ ................
+
+2009-11-10 15:47 +0000 [r229101] Matthew Nicholson <mnicholson@digium.com>
+
+ * UPGRADE-1.6.txt, main/editline/makelist.in, UPGRADE.txt: Reset
+ props that were accidently deleted in 229088.
+
+2009-11-10 15:28 +0000 [r229094] David Vossel <dvossel@digium.com>
+
+ * res/res_config_pgsql.c, /: Merged revisions 229093 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r229093 | dvossel | 2009-11-10 09:27:45 -0600 (Tue, 10 Nov 2009)
+ | 11 lines fixes pgsql double free of threadstorage A thread
+ storage variable was being freed incorrectly, which resulted in a
+ double free if two queries were made in the same thread. (closes
+ issue #16011) Reported by: cristiandimache Patches:
+ issue16011.diff uploaded by dvossel (license 671) ........
+
+2009-11-10 15:16 +0000 [r229088] Matthew Nicholson <mnicholson@digium.com>
+
+ * UPGRADE-1.6.txt, main/editline/makelist.in, channels/chan_sip.c,
+ UPGRADE.txt: Reverted revision 202007. (closes issue #16175)
+ Reported by: paul-tg Tested by: paul-tg
+
+2009-11-10 11:25 +0000 [r229078] Gavin Henry <ghenry@suretecsystems.com>
+
+ * contrib/scripts/asterisk.ldap-schema, /: Merged revisions 229050
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r229050 | ghenry | 2009-11-10 11:16:10 +0000 (Tue, 10
+ Nov 2009) | 20 lines Schema file additions * Added
+ AsteriskDialplan, AsteriskAccount and AsteriskMailbox
+ objectClasses to allow standalone dialplan, account and mailbox
+ entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage,
+ AstAccountTransport, AstAccountPromiscRedir, -
+ AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
+ - AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed
+ redundant IPaddr (there's already IPAddress) - Gives more
+ configuration Flags for SIP-Users available (tested) - Allows to
+ create Asterisk Attributes in defined Asterisk ObjectClasses
+ without extensibleObject (which really should be the last
+ resort); gives also additional possibilities for LDAP-filter
+ (closes issue #15874) Reported by: Medozas Patches:
+ asterisk.ldap-schema.patch uploaded by Medozas (license 41)
+ Tested by: Medozas, suretec ........
+
+2009-11-09 22:59 +0000 [r229017] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 229015 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r229015 | twilson | 2009-11-09 16:50:22 -0600 (Mon, 09 Nov 2009)
+ | 8 lines Don't crash when bridge->tech_pvt == NULL This is a
+ similar solution to what is in place for chan_agent (closes issue
+ #16003) Reported by: atis Tested by: twilson ........
+
+2009-11-09 22:17 +0000 [r229012] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes segfault when transferring a queue
+ caller In sip_hangup we attempted to lock p->owner after we set
+ it to NULL. Thanks to fhackenberger for reporting the issue and
+ submitting a patch. (closes issue #15848) Reported by:
+ fhackenberger Patches: digium_bug_0015848 uploaded by
+ fhackenberger (license 592) Tested by: fhackenberger, lmadsen,
+ TomS, shin-shoryuken, dvossel
+
+2009-11-09 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-rc5
+
+2009-11-09 15:40 +0000 [r228900] Leif Madsen <lmadsen@digium.com>
+
+ * main/channel.c: Merged revisions 228897 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228897 | lmadsen | 2009-11-09 09:38:38 -0600 (Mon, 09 Nov 2009)
+ | 14 lines Merged revisions 228896 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009)
+ | 6 lines Update WARNING message. Update a WARNING message to
+ give a suggested fix when encountered. (closes issue #16198)
+ Reported by: atis Tested by: atis ........ ................
+
+2009-11-09 14:48 +0000 [r228859] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 228858 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600
+ (Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov
+ 2009) | 8 lines Perform limited bounds checking when destroying
+ ast_mutex_t structures to make sure we don't try to use negative
+ indices. (closes issue #15588) Reported by: zerohalo Patches:
+ 20090820__issue15588.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo ........ ................
+
+2009-11-06 22:37 +0000 [r228694] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 228693 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228693 | dvossel | 2009-11-06 16:35:44 -0600 (Fri, 06 Nov 2009)
+ | 16 lines Merged revisions 228692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009)
+ | 9 lines fixes audiohook write crash occuring in chan_spy
+ whisper mode. After writing to the audiohook list in ast_write(),
+ frames were being freed incorrectly. Under certain conditions
+ this resulted in a double free crash. (closes issue #16133)
+ Reported by: wetwired ........ ................
+
+2009-11-06 20:26 +0000 [r228649] Matthew Nicholson <mnicholson@digium.com>
+
+ * funcs/func_base64.c, /, main/utils.c: Merged revisions 228620 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228620 | mnicholson | 2009-11-06 13:47:11 -0600
+ (Fri, 06 Nov 2009) | 15 lines Merged revisions 228378 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov
+ 2009) | 8 lines Properly handle '=' while decoding base64
+ messages and null terminate strings returned from BASE64_DECODE.
+ (closes issue #15271) Reported by: chappell Patches:
+ base64_fix.patch uploaded by chappell (license 8) Tested by:
+ kobaz ........ ................
+
+2009-11-06 18:43 +0000 [r228551] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 228548 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228548 | file | 2009-11-06 14:37:59 -0400 (Fri, 06 Nov 2009) |
+ 11 lines Merged revisions 228547 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4
+ lines Don't overwrite caller ID name on a trunk with the
+ configured fullname when using users.conf (issue ABE-1989)
+ ........ ................
+
+2009-11-06 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-rc4
+
+2009-11-06 17:54 +0000 [r228504] Joshua Colp <jcolp@digium.com>
+
+ * doc/tex/localchannel.tex, /: Merged revisions 228499 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2
+ lines Fix the localchannel.tex file. ........
+
+2009-11-06 17:24 +0000 [r228421-228447] David Vossel <dvossel@digium.com>
+
+ * codecs/codec_ilbc.c, /: Merged revisions 228441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228441 |
+ dvossel | 2009-11-06 11:22:31 -0600 (Fri, 06 Nov 2009) | 3 lines
+ Fixes merging issue from 1.4, frame data is held in data.ptr in
+ trunk ........
+
+ * codecs/codec_ilbc.c, /: Merged revisions 228420 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228420 | dvossel | 2009-11-06 11:09:01 -0600 (Fri, 06 Nov 2009)
+ | 19 lines Merged revisions 228418 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009)
+ | 13 lines fixes segfault in iLBC For reasons not yet known, it
+ appears possible for an ast_frame to have a datalen greater than
+ zero while the actual data is NULL during Packet Loss
+ Concealment. Most codecs don't support PLC so this doesn't affect
+ them. This patch catches the malformed frame and prevents the
+ crash from occuring. Additional efforts to determine why it is
+ possible for a frame to look like this are still being
+ investigated. (issue #16979) ........ ................
+
+2009-11-06 16:44 +0000 [r228413] Joshua Colp <jcolp@digium.com>
+
+ * /, main/abstract_jb.c: Merged revisions 228410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) |
+ 14 lines Merged revisions 228409 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7
+ lines Fix a bug caused by a partially invalid frame (from the
+ jitterbuffer) passing through the Asterisk core. (closes issue
+ #15560) Reported by: jvandal (closes issue #15709) Reported by:
+ covici ........ ................
+
+2009-11-06 15:43 +0000 [r228269-228340] David Vossel <dvossel@digium.com>
+
+ * /, main/astfd.c: Merged revisions 228339 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009)
+ | 12 lines Merged revisions 228338 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009)
+ | 5 lines fixes crash in astfd.c (closes issue #15981) Reported
+ by: slavon ........ ................
+
+ * funcs/func_audiohookinherit.c, /: Merged revisions 228268 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r228268 | dvossel | 2009-11-06 09:04:24 -0600 (Fri, 06
+ Nov 2009) | 9 lines fixes memory leak in func_audiohookinherit.c
+ (closes issue #15394) Reported by: boroda Patches:
+ bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790)
+ Tested by: dbrooks, boroda ........
+
+2009-11-05 22:13 +0000 [r228198] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 228196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228196 |
+ tilghman | 2009-11-05 16:12:45 -0600 (Thu, 05 Nov 2009) | 2 lines
+ Yet another error message in the dialplan (thanks,
+ rmudgett/russellb) ........
+
+2009-11-05 21:27 +0000 [r228195] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 228189 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228189 |
+ jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines
+ Fix the fix for chanspy option o In 224178, I assumed the
+ uploaded patch was correct as it had received positive feedback.
+ The flags were being checked in the incorrect location. Upon
+ testing the fix this time it was also found that the flags from
+ the dialplan weren't being copied to the
+ chanspy_translation_helper. (closes issue #16167) Reported by:
+ marhbere ........
+
+2009-11-05 21:27 +0000 [r228194] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 228191 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228191 |
+ tilghman | 2009-11-05 15:24:21 -0600 (Thu, 05 Nov 2009) | 7 lines
+ MEETME_INFO should not return a literal error message to the
+ dialplan. (closes issue #15450) Reported by: JimVanM Patches:
+ meetmeinfopatch.diff.txt uploaded by dbrooks (license 790) Tested
+ by: JimVanM ........
+
+2009-11-05 19:42 +0000 [r228148] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 228145 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600
+ (Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009)
+ | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to
+ chan_misdn connection. Patch submitted by gknispel_proformatique,
+ tested by francesco_r. "I have many crash since i have upgraded
+ to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out
+ an ast_frame. (closes issue #16041) Reported by: francesco_r
+ ........ ................
+
+2009-11-05 19:20 +0000 [r228093] Jason Parker <jparker@digium.com>
+
+ * channels/chan_vpb.cc, /: Merged revisions 228080 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228080 | qwell | 2009-11-05 13:16:29 -0600
+ (Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) |
+ 8 lines Fix crash on VPB exception when no hardware is present.
+ (closes issue #14970) Reported by: tzafrir Patches:
+ vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
+ markwaters ........ ................
+
+2009-11-05 17:14 +0000 [r228017] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_externalivr.c, /: Merged revisions 228015 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r228015 | tilghman | 2009-11-05 11:08:02 -0600 (Thu, 05 Nov 2009)
+ | 4 lines Don't crash if no arguments are passed. (closes issue
+ #16119) Reported by: thedavidfactor ........
+
+2009-11-04 23:53 +0000 [r227947] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 227945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009)
+ | 21 lines Merged revisions 227944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009)
+ | 14 lines Fix incorrect filename comparsion after monitor file
+ change The logic to detect if a requested file is indeed a
+ different file from the current file was incorrect. The main
+ issue being confusion of the use of filename_base which was
+ previously set without pathing information and then compared to
+ another full path. Robust file comparison logic has been added to
+ properly check if two files are the same even if symlinks are
+ used. (closes issue #15313) Reported by: caspy Patches:
+ 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
+ 325) but mostly tilghman's work ........ ................
+
+2009-11-04 21:09 +0000 [r227760-227831] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 227829 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov
+ 2009) | 17 lines Merged revisions 227827 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov
+ 2009) | 10 lines This patch modifies the Dial application to
+ monitor the calling channel for hangups while playing back
+ announcements. (closes issue #16005) Reported by: falves11
+ Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: mnicholson, falves11 Review:
+ https://reviewboard.asterisk.org/r/407/ ........ ................
+
+ * channels/chan_sip.c: Modify the SDP parsing code to parse session
+ and media level items separately. With the new code, media level
+ proprieties should no longer be confused with session level
+ proprieties. This change also reorganizes some of the SDP parsing
+ code which should make it easier to manage in the future. (closes
+ issue #14994) Reported by: frawd
+
+2009-11-04 19:28 +0000 [r227733-227748] Joshua Colp <jcolp@digium.com>
+
+ * /, static-http/prototype.js: Merged revisions 227739 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed,
+ 04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5
+ lines Fix a security issue where it may be possible for someone
+ to execute a cross-site AJAX request exploit. (AST-2009-009)
+ ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 227712 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) |
+ 12 lines Merged revisions 227700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5
+ lines Fix a security issue where sending a REGISTER with a
+ differing username in the From URI and Authorization header would
+ reveal whether it was valid or not. (AST-2009-008) ........
+ ................
+
+2009-11-03 20:01 +0000 [r227375] Jason Parker <jparker@digium.com>
+
+ * Makefile, /, main/Makefile: Merged revisions 227372 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) |
+ 9 lines Fix some build issues on Solaris. (closes issue #14517)
+ (SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded
+ by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell
+ ........
+
+2009-11-03 19:49 +0000 [r227364-227371] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_controlplayback.c, /: Merged revisions 227368 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03
+ Nov 2009) | 8 lines Change warning message to debug message.
+ app_controlplayback outputs a warning, when in fact it is normal.
+ (closes issue #16071) Reported by: atis Patches:
+ controlplayback_warning.patch uploaded by atis (license 242)
+ ........
+
+ * configs/extensions.conf.sample, /: Merged revisions 227361 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03
+ Nov 2009) | 11 lines Additional fixes to the
+ extensions.conf.sample file. Update the extensions.conf.sample
+ [stdexten] context so that we use the variable instead of
+ requiring it to be passed explicitly. Also updated uses of the
+ [stdexten] context throughout. (closes issue #15858) Reported by:
+ pprindeville Patches: stdexten-context-update.txt uploaded by
+ lmadsen (license 10) Tested by: pprindeville ........
+
+2009-11-03 18:15 +0000 [r227280] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Merged revisions 227275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009)
+ | 4 lines Make sure the outgoing flag is cleared if a new channel
+ fails to get created for outgoing calls. This is the relevant
+ portion of asterisk/trunk -r226648 ........
+
+2009-11-03 17:14 +0000 [r227239] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 227238 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r227238 |
+ dvossel | 2009-11-03 11:12:52 -0600 (Tue, 03 Nov 2009) | 5 lines
+ user.conf entries in SIP were not having their peer type set.
+ (closes issue #16120) Reported by: jsmith ........
+
+2009-11-03 15:40 +0000 [r227170] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 227167 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) |
+ 12 lines Merged revisions 227166 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5
+ lines Fix a bug where an RPID header could be generated with a
+ blank username in the URI. (closes issue #15909) Reported by:
+ kobaz ........ ................
+
+2009-11-03 15:25 +0000 [r227165] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 227162 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03
+ Nov 2009) | 7 lines Update extensions.conf.sample file to fix
+ incorrect extensions. (closes issue #15857) Reported by:
+ pprindeville Patches: stdexten.patch#2 uploaded by pprindeville
+ (license 347) Tested by: pprindeville ........
+
+2009-11-03 13:51 +0000 [r227156] Olle Johansson <oej@edvina.net>
+
+ * Makefile, /, channels/chan_sip.c: Merged revisions 227091 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis,
+ 03 Nov 2009) | 15 lines Merged revisions 227088 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7
+ lines Use proper response code when violating Contact ACL's.
+ https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a
+ quick review. (EDVX-003) ........ ................
+
+2009-11-02 21:06 +0000 [r226978] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_sip.c: SIP channel name uniqueness SIP channel
+ names were supposed to be unique by way of a name suffix derived
+ from the pointer to the channel's private data. Uniqueness was
+ preserved on 32-bit systems, but not on 64-bit systems. This
+ patch, as suggested by kpfleming, replaces this suffix with a
+ simple incremented unsigned int. (closes issue #15152) Reported
+ by: palbrecht Review: https://reviewboard.asterisk.org/r/420/
+
+2009-11-02 18:12 +0000 [r226893] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 226890 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) |
+ 18 lines Merged revisions 226889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) |
+ 11 lines Fix a bug where the recorded privacy introduction file
+ would not get removed if the caller hung up while the called
+ party had not yet answered. This was fixed by introducing an
+ argument to the 'n' option which, when enabled, removes the
+ introduction file under all scenarios. This was done to preserve
+ the behavior that has existed for quite some time. (closes issue
+ #14674) Reported by: ulogic Patches: bug14674.patch uploaded by
+ jpeeler (license 325) ........ ................
+
+2009-11-02 17:17 +0000 [r226815] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226812 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226812 | tilghman | 2009-11-02 11:15:31 -0600
+ (Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009)
+ | 8 lines Don't allow two separate instances of safe_asterisk
+ when restarting from the init script. (closes issue #14562)
+ Reported by: davidw Patches: Initially
+ 20091022__issue14562.diff.txt uploaded by tilghman (license 14)
+ Modified to 20091030__Issue14562_diff.txt uploaded by davidw
+ (license 780) Tested by: davidw ........ ................
+
+2009-10-29 18:18 +0000 [r226540] Joshua Colp <jcolp@digium.com>
+
+ * doc/tex/localchannel.tex, channels/chan_local.c, /: Merged
+ revisions 226532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) |
+ 13 lines Merged revisions 226531 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6
+ lines Add an option to enabling passing music on hold start and
+ stop requests through instead of acting on them in chan_local.
+ (closes issue #14709) Reported by: dimas ........
+ ................
+
+2009-10-28 21:32 +0000 [r226486] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * build_tools/get_documentation, /: remove empty awk pattern (//)
+ Solaris 10 nawk doesn't like the empty pattern such as '//' for
+ 'always'. Just remove that. No pattern at all always matches.
+ Merged revisions 226453 via svnmerge from
+ http://svn.digium.com/svn/asterisk/trunk
+
+2009-10-28 20:13 +0000 [r226379-226385] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Merged revisions 226384 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500
+ (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009)
+ | 9 lines Update documentation in sip.conf.sample. Update the
+ documentation in sip.conf.sample in order to make it more clear
+ that directmedia/canreinvite do not cause Asterisk to ignore
+ reINVITEs. It is only used to stop Asterisk from generating a
+ reINVITE, but does not stop it from accepting them if necessary.
+ (closes issue #15644) Reported by: lmadsen ........
+ ................
+
+ * doc/tex/channelvariables.tex: Merged revisions 226378 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500
+ (Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009)
+ | 7 lines Update CALLINGSUBADDR channel variable documentation.
+ (closes issue #15734) Reported by: alecdavis Patches:
+ channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
+ Tested by: alecdavis ........ ................
+
+2009-10-28 18:06 +0000 [r226170-226308] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 226305 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500
+ (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28
+ Oct 2009) | 2 lines Fix documentation (pointed out by
+ TheDavidFactor on #-dev) ........ ................
+
+ * main/manager.c, /: Merged revisions 226159 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009)
+ | 14 lines Merged revisions 226138 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009)
+ | 7 lines Manager output is not always NULL-terminated, so force
+ a NULL at the end of the filestream. (closes issue #15495)
+ Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
+ by tilghman (license 14) Tested by: pdf ........ ................
+
+2009-10-27 17:12 +0000 [r226101] Terry Wilson <twilson@digium.com>
+
+ * res/res_http_post.c, /: Merged revisions 226099 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r226099 |
+ twilson | 2009-10-27 11:48:54 -0500 (Tue, 27 Oct 2009) | 2 lines
+ Don't prepend the URI prefix to the post directory ........
+
+2009-10-27 00:16 +0000 [r226055] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, configure, configure.ac: detect ARM Linux EABI OSARCH as
+ linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even
+ if host_os is linux-gnueabi * When checking if we are Linux,
+ check OSARCH rather than host_os The newer ARM ABI ("EABI") shows
+ the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch
+ sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is
+ tested for the value of 'linux-gnu' in one or two places in the
+ tree. This patch also fixes the check libcap to check for $OSARCH
+ rather than $host_os . See also:
+ http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via
+ svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4
+ Merged revisions 226018 via svnmerge from
+ http://svn.digium.com/svn/asterisk/trunk
+
+2009-10-26 19:42 +0000 [r225914] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 225912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225912 |
+ jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines
+ ACL check not present for verifying SIP INVITEs The ACL check in
+ check_peer_ok was missing and has now been restored. The missing
+ check allowed for calls to be made on prohibited networks where
+ an ACL was defined in sip.conf and the allowguest option was set
+ to off. See the AST security advisory below for more information.
+ Merge code associated with AST-2009-007. (closes issue #16091)
+ Reported by: thom4fun ........
+
+2009-10-26 15:56 +0000 [r225871] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: Backport audio handling loop fixes from trunk
+ version of app_fax. This backport resolves some issues handling
+ audio frames during FAX processing, and ensures that the FAX
+ application doesn't accidentally get notified of a T.38
+ switchover at the end of a successful FAX. (closes issue #16127)
+
+2009-10-23 14:46 +0000 [r225651] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 225650 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225650 |
+ dvossel | 2009-10-23 09:41:50 -0500 (Fri, 23 Oct 2009) | 3 lines
+ Fixes an iterator memory leak and uninitialized memory ........
+
+2009-10-23 14:08 +0000 [r225585] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 225582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct
+ 2009) | 17 lines Merged revisions 225581 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct
+ 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on
+ every build. For some reason the menuselect.makeopts file was
+ listed as PHONY in the Makefile, resulting in 'make' needing to
+ rebuild it for every build. This then resulted in the embedded
+ module rules being rebuilt on every build, which can be slow and
+ is unnecessary. This patch fixes the problem by properly allowing
+ 'make' to know when the menuselect.makeopts file needs to be
+ rebuilt (defining the proper dependencies). ........
+ ................
+
+2009-10-22 22:24 +0000 [r225516] Leif Madsen <lmadsen@digium.com>
+
+ * README, /: Merged revisions 225515 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225515 |
+ lmadsen | 2009-10-22 17:24:03 -0500 (Thu, 22 Oct 2009) | 8 lines
+ Update README documentation. Update the README documentation to
+ correctly describe which CLI command you should use when
+ attempting to get help from the CLI. (closes issue #16064)
+ Reported by: thedavidfactor Patches: readme.patch uploaded by
+ thedavidfactor (license 903) ........
+
+2009-10-22 21:55 +0000 [r225489] David Vossel <dvossel@digium.com>
+
+ * apps/app_externalivr.c, include/asterisk/tcptls.h, main/tcptls.c,
+ /, channels/chan_sip.c: Merged revisions 225445 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225445 |
+ dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines
+ SIP TCP/TLS: move client connection setup/write into tcp helper
+ thread, various related locking/memory fixes. What this patch
+ fixes 1.Moves sip TCP/TLS connection setup into the TCP helper
+ thread: Connection setup takes awhile and before this it was
+ being done while holding the monitor lock. 2.Moves TCP/TLS
+ writing to the TCP helper thread: Through the use of a packet
+ queue and an alert pipe, the TCP helper thread can now be woken
+ up to write data as well as read data. 3.Locking error: sip_xmit
+ returned an XMIT_ERROR without giving up the tcptls_session lock.
+ This lock has been completely removed from sip_xmit and placed in
+ the new sip_tcptls_write() function. 4.Memory leak: When creating
+ a tcptls_client the tls_cfg was alloced but never freed unless
+ the tcptls_session failed to start. Now the session_args for a
+ sip client are an ao2 object which frees the tls_cfg on
+ destruction. 5.Pointer to stack variable: During
+ sip_prepare_socket the creation of a client's
+ ast_tcptls_session_args was done on the stack and stored as a
+ pointer in the newly created tcptls_session. Depending on the
+ events that followed, there was a slight possibility that pointer
+ could have been accessed after the stack returned. Given the new
+ changes, it is always accessed after the stack returns which is
+ why I found it. Notable code changes 1.I broke tcptls.c's
+ ast_tcptls_client_start() function into two functions. One for
+ creating and allocating the new tcptls_session, and a separate
+ one for starting and handling the new connection. This allowed me
+ to create the tcptls_session, launch the helper thread, and then
+ establish the connection within the helper thread. 2.Writes to a
+ tcptls_session are now done within the helper thread. This is
+ done by using an alert pipe to wake up the thread if new data
+ needs to be sent. The thread's sip_threadinfo object contains the
+ alert pipe as well as the packet queue. 3.Since the threadinfo
+ object contains the alert pipe, it must now be accessed outside
+ of the helper thread for every write (queuing of a packet). For
+ easy lookup, I moved the threadinfo objects from a linked list to
+ an ao2_container. (closes issue #13136) Reported by: pabelanger
+ Tested by: dvossel, whys (closes issue #15894) Reported by:
+ dvossel Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/380/ ........
+
+2009-10-22 21:54 +0000 [r225488] Leif Madsen <lmadsen@digium.com>
+
+ * doc/valgrind.txt, contrib/valgrind.supp (added): Merged revisions
+ 225485 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009)
+ | 19 lines Merged revisions 225484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009)
+ | 11 lines Clean valgrind output by suppressing false errors.
+ Update valgrind.txt documentation and add valgrind.supp file in
+ order to allow those who are creating valgrind output to have
+ less false errors in the logfile. (closes issue #16007) Reported
+ by: atis Patches: valgrind.txt.diff uploaded by atis (license
+ 242) asterisk2.supp uploaded by atis (license 242) Tested by:
+ atis, amorsen ........ ................
+
+2009-10-22 17:14 +0000 [r225363] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h:
+ Merged revisions 225360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009)
+ | 11 lines Merged revisions 225105 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009)
+ | 4 lines Fix documentation for ast_softhangup() and correct the
+ misuse thereof. (closes issue #16103) Reported by: majorbloodnok
+ ........ ................
+
+2009-10-21 22:00 +0000 [r225035-225308] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 225307 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500
+ (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009)
+ | 13 lines IAX2: VNAK loop caused by signaling frames with no
+ destination call number It is possible for the PBX thread to
+ queue up signaling frames before a destination call number is
+ received. This can result in signaling frames being sent out with
+ no destination call number. Since recent versions of Asterisk
+ require accurate destination callnumbers for all Full Frames,
+ this can cause a VNAK loop to occur. To resolve this no signaling
+ frames are sent until a destination callnumber is received, and
+ destination call numbers are now only required for iax_pvt
+ matching when the frame is an ACK. Review:
+ https://reviewboard.asterisk.org/r/413/ ........ ................
+
+ * configs/sip.conf.sample, channels/chan_iax2.c,
+ configs/iax.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 225033 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009)
+ | 27 lines Merged revisions 225032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009)
+ | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller
+ id removes '(', ' ', ')', non-trailing '.', and '-' from the
+ string. This means values such as 555.5555 and test-test result
+ in 555555 and testtest. There are instances, such as Skype
+ integration, where a specific value is passed via caller id that
+ must be preserved unmodified. This patch makes the shrinking of
+ caller id optional in chan_sip and chan_iax in order to support
+ such cases. By default this option is on to preserve previous
+ expected behavior. (closes issue #15940) Reported by: dimas
+ Patches: v2-15940.patch uploaded by dimas (license 88)
+ 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
+ Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/408/ ........ ................
+
+2009-10-20 22:11 +0000 [r224859] Tilghman Lesher <tlesher@digium.com>
+
+ * main/audiohook.c, funcs/func_speex.c, /: Merged revisions 224856
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r224856 | tilghman | 2009-10-20 17:09:07 -0500
+ (Tue, 20 Oct 2009) | 12 lines Merged revisions 224855 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009)
+ | 5 lines Pay attention to the return value of the manipulate
+ function. While this looks like an optimization, it prevents a
+ crash from occurring when used with certain audiohook callbacks
+ (diagnosed with SVN trunk, backported to 1.4 to keep the source
+ consistent across versions). ........ ................
+
+2009-10-20 17:50 +0000 [r224777] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 224774 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) |
+ 12 lines Merged revisions 224773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5
+ lines Add support for relaying early media in the features
+ attended transfer option. (closes issue #14828) Reported by:
+ licedey ........ ................
+
+2009-10-20 00:00 +0000 [r224674] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/rtp.c, /: Merged revisions 224671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct
+ 2009) | 14 lines Merged revisions 224670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct
+ 2009) | 7 lines Correct timestamp calculations when RTP sample
+ rates over 8kHz are used. While testing some endpoints that
+ support 16kHz and 32kHz sample rates, some log messages were
+ generated due to calc_rxstamp() computing timestamps in a way
+ that produced odd results, so this patch sanitizes the result of
+ the computations. ........ ................
+
+2009-10-19 19:54 +0000 [r224571] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 224567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) |
+ 12 lines Merged revisions 224565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5
+ lines Do not attempt early media bridging (ie: direct RTP setup)
+ if options are enabled that should prevent it. (closes issue
+ #14763) Reported by: cupotka ........ ................
+
+2009-10-19 19:41 +0000 [r224563] Kevin P. Fleming <kpfleming@digium.com>
+
+ * formats/format_siren14.c, /: Merged revisions 224562 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r224562 | kpfleming | 2009-10-19 14:40:26 -0500 (Mon, 19 Oct
+ 2009) | 1 line Remove useless debugging message. ........
+
+2009-10-19 00:13 +0000 [r224447-224451] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 224448 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009)
+ | 3 lines Allow ODBC storage to be queried with multiple
+ mailboxes, and remove multiple goto's. This corrects an issue
+ reported on the -users list. ........
+
+ * configs/res_odbc.conf.sample, /: Merged revisions 224446 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r224446 | tilghman | 2009-10-18 18:41:30 -0500 (Sun, 18
+ Oct 2009) | 2 lines Clarify that "forcecommit" is NOT an alias
+ for "autocommit", but instead controls the default disposition of
+ uncommitted transactions. ........
+
+2009-10-17 01:58 +0000 [r224334] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 224331 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500
+ (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009)
+ | 13 lines Fix stale caller id data from being reported in AMI
+ NewChannel event The problem here is that chan_dahdi is designed
+ in such a way to set certain values in the dahdi_pvt only once.
+ One of those such values is the configured caller id data in
+ chan_dahdi.conf. For PRI, the configured caller id data could be
+ overwritten during a call. Instead of saving the data and
+ restoring, it was decided that for all non-analog channels it was
+ simply best to not set the configured caller id in the first
+ place and also clear it at the end of the call. (closes issue
+ #15883) Reported by: jsmith ........ ................
+
+2009-10-16 20:58 +0000 [r224264] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 224261 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500
+ (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009)
+ | 18 lines Never released PRI channels when using Busy() or
+ Congestion() dialplan apps. When the Busy() or Congestion()
+ application is used towards ISDN (an ISDN progress is sent), the
+ responding ISDN Disconnect or Release may contain the ISDN cause
+ user busy or one of the congestion causes. In chan_dahdi.c these
+ causes will only set the needbusy or needcongestion flags and not
+ activate the softhangup procedure. Unfortunately only the latter
+ can interrupt the endless wait loop of Busy()/Congestion().
+ Result: PRI channels staying in state busy for the rest of
+ asterisk life or until the other end times out and forces the
+ call to clear. (in issue 0014292) Reported by: tomaso Patches:
+ disc_rel_userbusy.patch uploaded by tomaso (license 564) (This
+ patch is unrelated to the issue.) ........ ................
+
+2009-10-15 15:58 +0000 [r224181] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 224178 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 |
+ jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines
+ Readd removed ability to allow listening to one side of the call
+ in app_chanspy (Option o) (closes issue #15675) Reported by:
+ john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks
+ (license 790) Tested by: jgutierrez on users list:
+ http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html
+ ........
+
+2009-10-12 23:55 +0000 [r223835] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 223832 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009)
+ | 15 lines Merged revisions 223804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009)
+ | 8 lines Ensure ringing continues for branched calls after
+ progress is received While waiting for an answer, don't send
+ progress for branched calls for which ringing was sent. (closes
+ issue #15028) Reported by: fnordian ........ ................
+
+2009-10-12 21:01 +0000 [r223757] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, /: Merged revisions 223756 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009)
+ | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2
+ options SWP-151 ........
+
+2009-10-12 14:37 +0000 [r223655] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 223652
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12
+ Oct 2009) | 13 lines Remove automatic switching from T.38 to
+ voice mode in chan_sip. chan_sip has some code to automatically
+ switch from T.38 mode to voice mode when a voice frame is written
+ to the channel while it is in T.38 mode; this was intended to
+ handle the situation when a FAX transmission has ended and the
+ channel is not yet hung up, but is causing problems at the
+ beginning of FAX sessions as well when there are still voice
+ frames 'in flight' at the time the T.38 negotiation completes.
+ This patch removes the automatic switchover, and changes app_fax
+ to explicitly switch off T.38 mode when the FAX transmission
+ process ends. (closes issue #16025) Reported by: jamicque
+ ........
+
+2009-10-11 17:32 +0000 [r223490] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c, /: Merged revisions 223487 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009)
+ | 17 lines Merged revisions 223485-223486 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009)
+ | 6 lines Don't use data outside of its scope. The purpose of
+ this code was to have a hangup frame put on the list of deferred
+ frames. However, the code that read the hangup frame was outside
+ of the scope of where the hangup frame was declared. ........
+ r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009)
+ | 2 lines Remove some unnecessary code. ........ ................
+
+2009-10-09 23:12 +0000 [r223406] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_h323.c: Fix interpretation
+ of PRIREDIRECTIONREASON set by chan_sip. This commit is the
+ simplest way to solve a problem that has already been solved in
+ trunk with the "COLP/CONP and Redirecting party information into
+ Asterisk" commit. In trunk the redirection reason is translated
+ into a generic redirect reason. I would have had to do the same
+ fix except chan_sip never reads PRIREDIRECTREASON. So both
+ chan_dahdi and chan_h323 have been modified to interpret the one
+ different redirect reason of "no-answer" properly and set the
+ ISDN reason code 2 of "no reply". (closes issue #15033) Reported
+ by: steinwej
+
+2009-10-09 21:01 +0000 [r223333] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 223330 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 |
+ kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10
+ lines Initiate T.38 switchover when acting as called party,
+ regardless of FAX direction. SendFAX() and ReceiveFAX() can be
+ given options to indicate whether they should act as the calling
+ or called party; this mode should be used to decide whether to
+ initiate a switchover to T.38, not the direction that the FAX
+ transfer will take place. (closes issue #16039) Reported by:
+ jamicque ........
+
+2009-10-09 18:53 +0000 [r223286] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 223273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct
+ 2009) | 14 lines Merged revisions 223225 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct
+ 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING
+ when originating calls. (closes issue #15104) Reported by:
+ nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
+ (license 96) Tested by: nblasgen, mnicholson ........
+ ................
+
+2009-10-09 18:29 +0000 [r223257] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 223215 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct
+ 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri,
+ 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c
+ ........ ................
+
+2009-10-09 17:55 +0000 [r223208] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 223206 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009)
+ | 16 lines Merged revisions 223205 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009)
+ | 10 lines fixes sip registration using authuser in user.conf
+ (closes issue #14954) Reported by: tornblad Tested by:
+ mmichelson, tornblad, dvossel ........ ................
+
+2009-10-09 17:27 +0000 [r223173] Matthew Nicholson <mnicholson@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 223136 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct
+ 2009) | 8 lines Don't close the sqlite database when reloading.
+ Only close the database when unloading. (closes issue #15953)
+ Reported by: frawd Patches: sqlite3_rev220097.diff uploaded by
+ frawd (license 610) Tested by: frawd ........
+
+2009-10-09 17:09 +0000 [r223089-223133] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 223132 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 |
+ dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines
+ 'auth=' did not parse md5 secret correctly (closes issue #15949)
+ Reported by: ebroad Patches: authparsefix.patch uploaded by
+ ebroad (license 878) 15949_trunk.diff uploaded by dvossel
+ (license 671) Tested by: ebroad ........
+
+ * /, channels/chan_sip.c: Merged revisions 223088 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 |
+ dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines
+ p->peerauth is always empty in transmit_register() When using
+ callbackextension or specifing the peer name in a registration
+ string, the peer's specific auth settings set by the "auth="
+ strings within the peer definition are not used by the
+ registration. Thanks to ebroad for reporting the issue and
+ providing the patch. (closes issue #15955) Reported by: ebroad
+ Patches: regauthfix.patch uploaded by ebroad (license 878)
+ ........
+
+2009-10-08 20:00 +0000 [r222883] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/frame.h, include/asterisk/file.h, main/frame.c,
+ /, main/file.c: Merged revisions 222880 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009)
+ | 51 lines Merged revisions 222878 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009)
+ | 44 lines Make filestream frame handling safer by isolating
+ frames before returning them. This patch is related to a number
+ of issues on the bug tracker that show crashes related to freeing
+ frames that came from a filestream. A number of fixes have been
+ made over time while trying to figure out these problems, but
+ there re still people seeing the crash. (Note that some of these
+ bug reports include information about other problems. I am
+ specifically addressing the filestream frame crash here.) I'm
+ still not clear on what the exact problem is. However, what is
+ _very_ clear is that we have seen quite a few problems over time
+ related to unexpected behavior when we try to use embedded frames
+ as an optimization. In some cases, this optimization doesn't
+ really provide much due to improvements made in other areas. In
+ this case, the patch modifies filestream handling such that the
+ embedded frame will not be returned. ast_frisolate() is used to
+ ensure that we end up with a completely mallocd frame. In
+ reality, though, we will not actually have to malloc every time.
+ For filestreams, the frame will almost always be allocated and
+ freed in the same thread. That means that the thread local frame
+ cache will be used. So, going this route doesn't hurt. With this
+ patch in place, some people have reported success in not seeing
+ the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609)
+ Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt
+ uploaded by russell (license 2) Tested by: aragon, russell
+ (closes issue #15817) Reported by: zerohalo Tested by: zerohalo
+ (closes issue #15845) Reported by: marhbere Review:
+ https://reviewboard.asterisk.org/r/386/ ........ ................
+
+2009-10-08 19:41 +0000 [r222874] David Vossel <dvossel@digium.com>
+
+ * main/netsock.c, /, include/asterisk/netsock.h: Merged revisions
+ 222873 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 |
+ dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines
+ fixes an ast_netsock_list memory leak. ABE-1998 Review:
+ https://reviewboard.asterisk.org/r/395/ ........
+
+2009-10-08 16:51 +0000 [r222695-222802] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn_config.c, /: Merged revisions 222799 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222799 | rmudgett | 2009-10-08 11:44:33 -0500
+ (Thu, 08 Oct 2009) | 19 lines Merged revisions 222797 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009)
+ | 12 lines Fix memory leak if chan_misdn config parameter is
+ repeated. Memory leak when the same config option is set more
+ than once in an misdn.conf section. Why must this be considered?
+ Templates! Defining a template with default port options and
+ later adding to or overriding some of them. Patches:
+ memleak-misdn.patch JIRA ABE-1998 ........ ................
+
+ * channels/chan_misdn.c, /: Merged revisions 222692 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222692 | rmudgett | 2009-10-07 16:56:36 -0500
+ (Wed, 07 Oct 2009) | 21 lines Merged revisions 222691 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009)
+ | 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf:
+ astdtmf must be set to "yes". With "no", buffer loss does not
+ occur. The translated frame "f2" when passing through
+ ast_dsp_process() is not freed whenever it is not used further in
+ process_ast_dsp(). Then in the end it is never ever freed.
+ Patches: translate.patch JIRA ABE-1993 ........ ................
+
+2009-10-07 18:06 +0000 [r222549] Jason Parker <jparker@digium.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 222548 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r222548 | qwell | 2009-10-07 13:04:56 -0500 (Wed, 07 Oct
+ 2009) | 5 lines Remove 'keepstats' queue option from sample
+ config, as it's no longer used.
+ https://reviewboard.asterisk.org/r/115/ (closes issue #15820)
+ Reported by: kshumard ........
+
+2009-10-07 18:00 +0000 [r222547] Sean Bright <sean@malleable.com>
+
+ * funcs/func_strings.c: Fix merge error.
+
+2009-10-07 17:45 +0000 [r222544] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 222543 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r222543 | dvossel | 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009)
+ | 14 lines Merged revisions 222542 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009)
+ | 8 lines crash on transfer handle_invite_replaces() attempts to
+ uplock a pvt's owner channel without first verifing that it
+ exists. (issue #16027) ........ ................
+
+2009-10-06 23:59 +0000 [r222354-222466] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 222463 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222463 | jpeeler | 2009-10-06 18:56:01 -0500
+ (Tue, 06 Oct 2009) | 14 lines Merged revisions 222462 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009)
+ | 8 lines Add missing unlock(s) in dahdi_read (two cases in
+ trunk, and 1.6.2) (closes issue #15683) Reported by: alecdavis
+ ........ ................
+
+ * channels/chan_dahdi.c: Fix potential crash when entire span
+ request is received. The variable index used in this scenario for
+ accessing the dahdi_pvts was wrong and was most likely copied
+ from the several other places it is used correctly. (closes issue
+ #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch
+ uploaded by tsearle (license 373)
+
+ * channels/chan_dahdi.c, /: Merged revisions 222351 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009)
+ | 9 lines Fix 222298 (crash during destruction of second channel
+ when variable set with setvar). I mistakenly reasoned that setvar
+ would be used on all channels. Since it can be set per channel,
+ give each dahdi channel a copy of the variable. (related to
+ #15899) ........
+
+2009-10-06 19:41 +0000 [r222311] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_pgsql.c, res/res_config_pgsql.c, /: Merged revisions
+ 222309 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r222309 |
+ tilghman | 2009-10-06 14:31:39 -0500 (Tue, 06 Oct 2009) | 10
+ lines Change schema query to involve the use of an optional
+ schema parameter. This change is done in such a way as to allow
+ the driver to continue to function with older databases which
+ don't have these features. (closes issue #16000) Reported by:
+ jamicque Patches: 20091002__issue16000.diff.txt uploaded by
+ tilghman (license 14) 20091002__issue16000__1.6.1.diff.txt
+ uploaded by tilghman (license 14) Tested by: jamicque ........
+
+2009-10-06 19:27 +0000 [r222304] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 222298 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009)
+ | 9 lines Fix crash during destruction of second channel when
+ variable set with setvar. The setvar line in chan_dahdi.conf is
+ shared among all the channels, so make sure to only free the
+ resources only when the last channel is destroyed. (closes issue
+ #15899) Reported by: tzafrir ........
+
+2009-10-06 19:22 +0000 [r222289] Tilghman Lesher <tlesher@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 222273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r222273 |
+ tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06 Oct 2009) | 5 lines
+ When we call a gosub routine, the variables should be scoped to
+ avoid contaminating the caller. This affected the ~~EXTEN~~ hack,
+ where a subroutine might have changed the value before it was
+ used in the caller. Patch by myself, tested by ebroad on
+ #asterisk ........
+
+2009-10-06 Leif Madsen <lmadsen@digium.com>
+
+ * Released Asterisk 1.6.2.0-rc3
+
+2009-10-06 01:39 +0000 [r222113-222187] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c,
+ channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c,
+ res/res_clialiases.c, /, channels/chan_sip.c,
+ funcs/func_dialgroup.c, include/asterisk/astobj2.h,
+ res/res_phoneprov.c: Merged revisions 222176 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct
+ 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05
+ Oct 2009) | 20 lines Fix ao2_iterator API to hold references to
+ containers being iterated. See Mantis issue for details of what
+ prompted this change. Additional notes: This patch changes the
+ ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum
+ instead of a macro, with a name that fits our naming policy;
+ also, it is now necessary to call ao2_iterator_destroy() on any
+ iterator that has been created. Currently this only releases the
+ reference to the container being iterated, but in the future this
+ could also release other resources used by the iterator, if the
+ iterator implementation changes to use additional resources.
+ (closes issue #15987) Reported by: kpfleming Review:
+ https://reviewboard.asterisk.org/r/383/ ........ ................
+
+ * configs/sip.conf.sample, main/udptl.c, /, channels/chan_sip.c,
+ configs/udptl.conf.sample, UPGRADE.txt: Merged revisions 222110
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05
+ Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be
+ supportable via configuration option. Many T.38 endpoints
+ incorrectly send the maximum IFP frame size they can accept as
+ the T38FaxMaxDatagram value in their SDP, when in fact this value
+ is supposed to be the maximum UDPTL payload size (datagram size)
+ they can accept. If the value they supply is small enough (a
+ commonly supplied value is '72'), T.38 UDPTL transmissions will
+ likely fail completely because the UDPTL packets will not have
+ enough room for a primary IFP frame and the redundancy used for
+ error correction. If this occurs, the Asterisk UDPTL stack will
+ emit log messages warning that data loss may occur, and that the
+ value may need to be overridden. This patch extends the
+ 't38pt_udptl' configuration option in sip.conf to allow the
+ administrator to override the value supplied by the remote
+ endpoint and supply a value that allows T.38 FAX transmissions to
+ be successful with that endpoint. In addition, in any SIP call
+ where the override takes effect, a debug message will be printed
+ to that effect. This patch also removes the T38FaxMaxDatagram
+ configuration option from udptl.conf.sample, since it has not
+ actually had any effect for a number of releases. In addition,
+ this patch cleans up the T.38 documentation in sip.conf.sample
+ (which incorrectly documented that T.38 support was passthrough
+ only). (issue #15586) Reported by: globalnetinc ........
+
+2009-10-02 17:35 +0000 [r222032] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 222030 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222030 | dvossel | 2009-10-02 12:34:07 -0500
+ (Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02
+ Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a
+ memcpy. ........ ................
+
+2009-10-02 17:01 +0000 [r221923-221974] Tilghman Lesher <tlesher@digium.com>
+
+ * main/astobj2.c, /: Merged revisions 221971 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221971 | tilghman | 2009-10-02 11:59:57 -0500 (Fri, 02 Oct 2009)
+ | 9 lines Merged revisions 221970 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009)
+ | 2 lines Ensure the result of the hash function is positive.
+ Negative array offsets suck. ........ ................
+
+ * /, main/logger.c: Merged revisions 221920 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221920 |
+ tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines
+ Initialize a variable that we check immediately upon startup.
+ (closes issue #15973) Reported by: atis ........
+
+2009-10-02 01:35 +0000 [r221879] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /:
+ Merged revisions 221844 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009)
+ | 33 lines Merged revisions 221769 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009)
+ | 26 lines Occasionally losing use of B channels in chan_misdn. I
+ have not been able to reproduce the problem of losing channels.
+ However, I have seen in the code a reentrancy problem that might
+ give these symptoms. The reentrancy patch does several things: 1)
+ Guards B channel and B channel structure allocation. 2) Makes the
+ B channel structure find routines more precise in locating
+ records. 3) Never leave a B channel allocated if we received
+ cause 44. The last item may cause temporary outgoing call
+ problems, but they should clear when the line becomes idle.
+ (closes issue #15490) Reported by: slutec18 Patches:
+ issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
+ Reported by: FabienToune Patches:
+ issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: FabienToune, rmudgett, slutec18 ........
+ ................
+
+2009-10-02 00:07 +0000 [r221744-221780] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, main/rtp.c, /, main/say.c: Merged revisions
+ 221777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009)
+ | 9 lines Merged revisions 221776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009)
+ | 2 lines Fix a bunch of off-by-one errors ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 221705 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 |
+ tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines
+ Revision 220906 (a merge from 1.4) was not merged correctly,
+ causing a problem with non-dynamic peers. ........
+
+2009-10-01 19:35 +0000 [r221698] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 221697 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 |
+ dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines
+ outbound tls connections were not defaulting to port 5061 (closes
+ issue #15854) Reported by: dvossel Patches:
+ sip_port_config_trunk.diff uploaded by dvossel (license 671)
+ Tested by: dvossel ........
+
+2009-10-01 16:57 +0000 [r221660] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 221554,221589 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu,
+ 01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE
+ constructs when it's just TRUE or FALSE. ................ r221589
+ | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9
+ lines Merged revisions 221588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct
+ 2009) | 2 lines Use unsigned ints for portinuri flags. ........
+ ................
+
+2009-10-01 16:25 +0000 [r221622] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /, configs/udptl.conf.sample, UPGRADE.txt: Merged
+ revisions 221592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 |
+ kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12
+ lines Remove ability to control T.38 FAX error correction from
+ udptl.conf. chan_sip has had the ability to control T.38 FAX
+ error correction mode on a per-peer (or global) basis for a
+ couple of releases now, which is where it should have been all
+ along. This patch removes the ability to configure it in
+ udptl.conf, but issues a warning if the user tries to do, telling
+ them to look at sip.conf.sample for how to configure it now. For
+ any SIP peers that are T.38 enabled in sip.conf, there is already
+ a default for FEC error correction even if the user does not
+ specify any mode, so this change will not turn off error
+ correction by default, it will have the same default value that
+ has been in the udptl.conf sample file. ........
+
+2009-09-30 23:07 +0000 [r221477-221485] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 221484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221484 |
+ mnicholson | 2009-09-30 18:04:03 -0500 (Wed, 30 Sep 2009) | 2
+ lines Cleaned up merge from r221432 ........
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 221432 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep
+ 2009) | 17 lines Merged revisions 221360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep
+ 2009) | 10 lines Fix SRV lookup and Request-URI generation in
+ chan_sip. This patch adds a new field "portinuri" to the sip
+ dialog struct and the sip peer struct. That field is used during
+ RURI generation to determine if the port should be included in
+ the RURI. It is also used in some places to determine if an SRV
+ lookup should occur. (closes issue #14418) Reported by: klaus3000
+ Tested by: klaus3000, mnicholson Review:
+ https://reviewboard.asterisk.org/r/369/ ........ ................
+
+2009-09-30 21:46 +0000 [r221371-221472] Matthias Nick <mnick@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 221436 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221436 |
+ mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines
+ Prevents from division by zero ........
+
+ * configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged
+ revisions 221368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) |
+ 23 lines Merged revisions 221153,221157,221303 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) |
+ 2 lines check bounds - prevents for buffer overflow ........
+ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) |
+ 8 lines added a new dialplan function 'CSV_QUOTE' and changed the
+ cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
+ Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
+ mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed,
+ 30 Sep 2009) | 2 lines changed the prototype definition of
+ csv_quote ........ ................
+
+2009-09-30 19:15 +0000 [r221304] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample, main/rtp.c, /, channels/chan_sip.c,
+ include/asterisk/rtp.h: Merged revisions 221266 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009)
+ | 32 lines Merged revisions 221086 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009)
+ | 25 lines Change the SSRC by default when our media stream
+ changes Be default, change SSRC when doing an audio stream
+ changes Asterisk doesn't honor marker bit when reinvited to
+ already-bridged RTP streams,resulting in far-end stack discarding
+ packets with "old" timestamps that areactually part of a new
+ stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is
+ a reinvite, unless the 'constantssrc' is set to true in sip.conf.
+ The original issue reported to Digium support detailed the
+ following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
+ Application Server Call comes in fromITSP, Asterisk dials the app
+ server which sends a re-invite back toAsterisk--not to negotiate
+ to send media directly to the ITSP, but to indicatethat it's
+ changing the stream it's sending to Asterisk. The app
+ servergenerates a new SSRC, sequence numbers, timestamps, and
+ sets the marker bit on the new stream. Asterisk passes through
+ the teimstamp of the new stream, butdoes not reset the SSRC,
+ sequence numbers, or set the marker bit. When the timestamp on
+ the new stream is older than the timestamp on the originalstream,
+ the ITSP (which doesn't know there has been any change) discards
+ the newframes because it thinks they are too old. This patch
+ addresses this by changing the SSRC on a stream update unless
+ constantssrc=true is set in sip.conf. Review:
+ https://reviewboard.asterisk.org/r/374/ ........ ................
+
+2009-09-30 16:57 +0000 [r221204] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 221201 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221201 | tilghman | 2009-09-30 11:56:42 -0500 (Wed, 30 Sep 2009)
+ | 14 lines Merged revisions 221200 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009)
+ | 7 lines Avoid a potential NULL dereference. (closes issue
+ #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
+ uploaded by tilghman (license 14) Tested by: kobaz ........
+ ................
+
+2009-09-30 14:57 +0000 [r221089] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 221085 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep
+ 2009) | 9 lines Clarify documentation for VoiceMailMain()'s a()
+ option. We require box numbers, not names as the documentation
+ implies. (issue #14740) Reported by: pj Patches:
+ __20090729-app_voicemail-documentation.patch uploaded by lmadsen
+ (license 10) Tested by: seanbright, lmadsen ........
+
+2009-09-30 04:41 +0000 [r221027-221047] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_lock.c: Recorded merge of revisions 221044 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r221044 | tilghman | 2009-09-29 23:32:36 -0500 (Tue, 29
+ Sep 2009) | 8 lines Allow locks to be inherited through a
+ masquerade without causing starvation. (closes issue #14859)
+ Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded
+ by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt
+ uploaded by tilghman (license 14) Tested by: atis, tilghman
+ ........
+
+ * include/asterisk/smdi.h, include/asterisk/optional_api.h
+ (removed), apps/app_voicemail.c, include/asterisk/agi.h,
+ include/asterisk/monitor.h: Remove optional_api from 1.6.2
+ branch, since it is not currently working. This is a blocking
+ issue for the 1.6.2 release. (closes issue #15914) Reported by:
+ mbeckwell Branch:
+ http://svn.digium.com/svn/asterisk/team/tilghman/optional_api_162
+ Tested by: mbeckwell
+
+ * /, channels/chan_sip.c: Merged revisions 220906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009)
+ | 16 lines Merged revisions 220873 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009)
+ | 9 lines Reduce CPU usage related to building a peer merely for
+ devicestates. This fixes a 100% CPU problem in the SIP driver,
+ found by profiling the driver while the problem was occurring.
+ (closes issue #14309) Reported by: pkempgen Patches:
+ 20090924__issue14309.diff.txt uploaded by tilghman (license 14)
+ Tested by: pkempgen, vrban ........ ................
+
+2009-09-29 20:24 +0000 [r220905-220934] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the
+ spyee is masqueraded and chanspy_ds_chan_fixup() is called with
+ the channel locked. (closes issue #15965) Reported by: atis
+ Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: atis
+
+ * /, apps/app_confbridge.c: Merged revisions 220904 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220904 | mnicholson | 2009-09-29 14:49:02 -0500 (Tue, 29 Sep
+ 2009) | 5 lines Fix options 'm' and 's'. They were swapped in the
+ code. Also document the fact that app_confbridge does not
+ automatically answer the channel. (closes issue #15964) Reported
+ by: shrift ........
+
+2009-09-29 17:06 +0000 [r220836] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 220833 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009)
+ | 12 lines Make deletion of temporary greetings work properly
+ with IMAP_STORAGE When imapgreetings was set to yes, the message
+ was being deleted but wasn't actually being expunged. When
+ imapgreetings was set to no, the file based message was not being
+ deleted at all. All good now! (closes issue #14949) Reported by:
+ noahisaac Patches: vm_tempgreeting_removal.patch uploaded by
+ noahisaac (license 748), modified by me ........
+
+2009-09-28 19:13 +0000 [r220725] Sean Bright <sean@malleable.com>
+
+ * /, Makefile.rules: Merged revisions 220721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220721 | seanbright | 2009-09-28 15:11:20 -0400 (Mon, 28 Sep
+ 2009) | 10 lines Merged revisions 220717 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep
+ 2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect,
+ explicitly pass -O0 to the compiler so we override any default
+ optimization levels for a particular install. ........
+ ................
+
+2009-09-28 19:11 +0000 [r220722] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 220718 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r220718 |
+ jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines
+ Fix building of registration entry in build_peer when using
+ callbackextension Check for remotesecret option was
+ unintentionally always true, which therefore caused the secret
+ option to never be used. Thanks to dvossel for pointing out the
+ exact fix. (closes issue #15943) Reported by: tpsast ........
+
+2009-09-27 20:45 +0000 [r220632] Michiel van Baak <michiel@vanbaak.info>
+
+ * funcs/func_callerid.c, /: Merged revisions 220629 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220629 | mvanbaak | 2009-09-27 22:40:16 +0200 (Sun, 27 Sep 2009)
+ | 3 lines add name argument for the CALLERID dialplan function to
+ the xml documentation. Pointed out to me on IRC by snuff-home.
+ Thanks ........
+
+2009-09-26 15:12 +0000 [r220589] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/aes.h: Merged revisions 220586 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009)
+ | 2 lines Allow AES to compile, when OpenSSL is not present.
+ ........
+
+2009-09-24 20:38 +0000 [r220369] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 220365 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r220365 |
+ dvossel | 2009-09-24 15:37:20 -0500 (Thu, 24 Sep 2009) | 8 lines
+ fixes tcptls_session memory leak caused by ref count error
+ (closes issue #15939) Reported by: dvossel Review:
+ https://reviewboard.asterisk.org/r/375/ ........
+
+2009-09-24 19:42 +0000 [r220292] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_playback.c, main/pbx.c, /, apps/app_disa.c: Merged
+ revisions 220289 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009)
+ | 13 lines Merged revisions 220288 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009)
+ | 6 lines Implicitly sending a progress signal breaks some
+ applications. Call Progress() in your dialplan if you explicitly
+ want progress to be sent. (Reverts change 216430, closes issue
+ #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing
+ list
+ http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
+ ........ ................
+
+2009-09-24 18:22 +0000 [r220103-220221] Sean Bright <sean@malleable.com>
+
+ * Makefile, /: Merged revisions 220217 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220217 | seanbright | 2009-09-24 14:19:41 -0400 (Thu, 24 Sep
+ 2009) | 9 lines Merged revisions 220213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep
+ 2009) | 1 line Resolve parallel build warnings. Reported by Klaus
+ Darilion on the asterisk-dev mailing list. ........
+ ................
+
+ * Makefile, build_tools/mkpkgconfig, /: Merged revisions 220100 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r220100 | seanbright | 2009-09-24 10:44:08 -0400
+ (Thu, 24 Sep 2009) | 9 lines Merged revisions 220099 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu,
+ 24 Sep 2009) | 2 lines Remove the remaining bashisms in the
+ Makefile/mkpkgconfig ........ ................
+
+2009-09-24 08:43 +0000 [r220031] Michiel van Baak <michiel@vanbaak.info>
+
+ * build_tools/mkpkgconfig, /: Merged revisions 220028 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r220028 | mvanbaak | 2009-09-24 10:36:18 +0200
+ (Thu, 24 Sep 2009) | 14 lines Merged revisions 220027 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009)
+ | 7 lines mkpkgconfig does not need bash so make it use /bin/sh
+ This fixes building on all systems that don't have bash at
+ /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on
+ #asterisk-dev ........ ................
+
+2009-09-24 07:45 +0000 [r219989] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_directory.c, /: Merged revisions 219987 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r219987 | tilghman | 2009-09-24 02:39:44 -0500 (Thu, 24 Sep 2009)
+ | 8 lines Fix two possible crashes, one only in 1.6.1 and one in
+ 1.6.1 forward. (closes issue #15739) Reported by: DLNoah, jeffg
+ Patches: 20090914__issue15739.diff.txt uploaded by tilghman
+ (license 14) 20090922__issue15739.diff.txt uploaded by tilghman
+ (license 14) Tested by: DLNoah, jeffg ........
+
+2009-09-22 21:48 +0000 [r219821] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 219818 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219818 | tilghman | 2009-09-22 16:43:22 -0500
+ (Tue, 22 Sep 2009) | 17 lines Merged revisions 219816 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009)
+ | 10 lines When IMAP variables were changed during a reload,
+ Voicemail did not use the new values. This change introduces a
+ configuration version variable, which ensures that connections
+ with the old values are not reused but are allowed to expire
+ normally. (closes issue #15934) Reported by: viniciusfontes
+ Patches: 20090922__issue15934.diff.txt uploaded by tilghman
+ (license 14) Tested by: viniciusfontes ........ ................
+
+2009-09-21 17:01 +0000 [r219722] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 219721 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219721 | dvossel | 2009-09-21 11:59:05 -0500
+ (Mon, 21 Sep 2009) | 9 lines Merged revisions 219720 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21
+ Sep 2009) | 3 lines Reverting merge 219520. This change was not
+ necessary. ........ ................
+
+2009-09-20 18:21 +0000 [r219669] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/file.c: Merged revisions 219654 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219654 | tilghman | 2009-09-20 12:55:49 -0500 (Sun, 20 Sep 2009)
+ | 15 lines Merged revisions 219653 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009)
+ | 8 lines Really stop the stream, when ast_closestream() is
+ called. (closes issue #15129) Reported by: bmh Patches:
+ 20090918__issue15129.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/372/ ........
+ ................
+
+2009-09-19 03:14 +0000 [r219590] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 219587 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219587 | russell | 2009-09-18 21:59:52 -0500
+ (Fri, 18 Sep 2009) | 13 lines Merged revisions 219586 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009)
+ | 6 lines Make sure the iax_pvt exists before dereferencing it.
+ This fixes the latest crash posted on issue 15609. (issue #15609)
+ ........ ................
+
+2009-09-18 23:21 +0000 [r219452-219521] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 219520 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219520 | dvossel | 2009-09-18 18:20:58 -0500
+ (Fri, 18 Sep 2009) | 15 lines Merged revisions 219519 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009)
+ | 9 lines iax2 frame double free The iax frame's retrans sched id
+ was written over right before iax2_frame_free was called. In
+ iax2_frame_free that retrans id is used to delete the sched item.
+ By writing over the retrans field before the sched item could be
+ deleted, it was possible for a retransmit to occur on a freed
+ frame. ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 219451 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009)
+ | 20 lines Merged revisions 219450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009)
+ | 14 lines via-header branches not updated correctly on INVITE
+ INVITE requests must always contain a new unique branch id. When
+ a new branch id is created for an INVITE, the dialog's
+ invite_branch variable must be updated so CANCEL requests use the
+ correct branch id. (closes issue #15262) Reported by: maniax
+ Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
+ (license 608) invite_new_branch_trunk.diff uploaded by dvossel
+ (license 671) Tested by: maniax, dvossel ........
+ ................
+
+2009-09-18 13:57 +0000 [r219415] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 219412 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009)
+ | 6 lines Missing value setting line for maxsecs/maxmessage
+ (closes issue #15696) Reported by: fhackenberger Patches:
+ maxsecs.patch uploaded by fhackenberger (license 592) ........
+
+2009-09-17 22:38 +0000 [r219376] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219371 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r219371 |
+ dvossel | 2009-09-17 17:37:28 -0500 (Thu, 17 Sep 2009) | 9 lines
+ fixes deadlock when performing directed pickup w Invite/replaces
+ (closes issue #15340) Reported by: lmsteffan Patches:
+ deadlock.patch uploaded by lmsteffan (license 779) Tested by:
+ lmsteffan ........
+
+2009-09-17 22:37 +0000 [r219370] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219324 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep
+ 2009) | 12 lines Merged revisions 219320 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep
+ 2009) | 6 lines Send a 100 Trying response when we detect a
+ spiral. This was problematic during spiral tests at SIPit...
+ along with some other things as well. ........ ................
+
+2009-09-17 22:06 +0000 [r219307] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219304 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009)
+ | 27 lines Merged revisions 219303 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009)
+ | 21 lines INVITE w/Replaces deadlock fix This patch cleans up
+ the locking logic in chan_sip.c's handle_invite_replaces()
+ function as well as making use of ast_do_masquerade() rather than
+ forcing the masquerade on an ast_read(). The code had several
+ redundant unlocks that would result in 'freed more times than
+ we've locked!' errors. I cleaned these up as well as moving all
+ the unlock logic to the end of the function. This patch should
+ also resolve the issue people were having with the replacecall
+ channel never being unlocked with one legged calls. (closes issue
+ #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff
+ uploaded by dvossel (license 671) Tested by: irroot, dvossel
+ Review: https://reviewboard.asterisk.org/r/371/ ........
+ ................
+
+2009-09-17 19:58 +0000 [r219267] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219264 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r219264 |
+ file | 2009-09-17 14:57:39 -0500 (Thu, 17 Sep 2009) | 2 lines
+ Ensure no spaces exist before "refresher=" when doing the
+ comparison. ........
+
+2009-09-17 Leif Madsen <lmadsen@digium.com>
+
+ * Released Asterisk 1.6.2.0-rc2
+
+2009-09-17 15:38 +0000 [r219194] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /, include/asterisk/cdr.h,
+ include/asterisk/channel.h: Merged revisions 219139 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500
+ (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep
+ 2009) | 10 lines Prevent a potential race condition and crash
+ when hanging up a channel by removing the channel from the
+ channel list before begining channel tear down. This fix may
+ potentially cause problems with CDR backends that access the
+ channel a CDR is associated with via the channel list. This fix
+ makes the channel unavabile at the time when the CDR backend is
+ invoked. This has been documented in include/asterisk/cdr.h.
+ (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
+ Review: https://reviewboard.asterisk.org/r/362/ ........
+ ................
+
+2009-09-16 23:52 +0000 [r219063] Tilghman Lesher <tlesher@digium.com>
+
+ * main/config.c, configs/extensions.conf.sample, /: Merged
+ revisions 219061 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009)
+ | 15 lines Merged revisions 219023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009)
+ | 8 lines Properly deal with quotes in the arguments of '#exec'
+ includes. (closes issue #15583) Reported by: pkempgen Patches:
+ 20090726__issue15583.diff.txt uploaded by tilghman (license 14)
+ 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
+ 169) Tested by: pkempgen ........ ................
+
+2009-09-16 19:40 +0000 [r218938] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c, /: Merged revisions 218868 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009)
+ | 20 lines Merged revisions 218867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009)
+ | 13 lines Fixes CID pattern matching behavior to mirror that of
+ extension pattern matching. Pattern matching for extensions uses
+ a type of scoring system, giving values for specificity to each
+ character in the pattern. Unfortunately, this is done character
+ by character, in order. This does lead to some less specific
+ patterns being first in line for matching, but it will usually
+ get the job done. This patch merely brings CID matching to the
+ same level as extension matching. This patch does not attempt to
+ tackle the problem shared by extension matching. (closes issue
+ #14708) Reported by: klaus3000 ........ ................
+
+2009-09-16 19:29 +0000 [r218937] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 |
+ mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12
+ lines Reverse order of args to fread. This way, we don't always
+ write a null byte into byte 1 of the buffer (closes issue #15905)
+ Reported by: ebroad Patches: freadfix.patch uploaded by ebroad
+ (license 878) Tested by: ebroad ........
+
+2009-09-16 19:25 +0000 [r218934] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 |
+ file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On
+ TCP and TLS connections do not attempt to stop retransmission of
+ the packet internally. This was preventing responses from being
+ properly processed because the packet was not being found causing
+ handle_response to return prematurely. ........
+
+2009-09-16 13:38 +0000 [r218802] Russell Bryant <russell@digium.com>
+
+ * contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged
+ revisions 218799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009)
+ | 16 lines Merged revisions 218798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009)
+ | 9 lines Remove the IAXy firmware from Asterisk. The firmware
+ can now be found on downloads.digium.com, where the rest of our
+ binary downloads live. This was the last part of our Asterisk
+ tarballs that was considered non-free by Debian. :-) (closes
+ issue #15838) Reported by: paravoid ........ ................
+
+2009-09-15 22:46 +0000 [r218733] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 218731 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500
+ (Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009)
+ | 6 lines If the user enters the same password as before, don't
+ signal an error when the change does nothing. (closes issue
+ #15492) Reported by: cbbs70a Patches:
+ 20090713__issue15492.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+2009-09-15 19:24 +0000 [r218688] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218687 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 |
+ dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines
+ upward bound checking for port string to int conversion ........
+
+2009-09-15 16:18 +0000 [r218590] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218586 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep
+ 2009) | 15 lines Merged revisions 218578 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep
+ 2009) | 8 lines Send request contact header field with response
+ to registrer queries instead of the address of record. (closes
+ issue #14438) Reported by: ravindrad Patches: regquerypatch
+ uploaded by ravindrad (license 684) Tested by: ravindrad ........
+ ................
+
+2009-09-15 16:06 +0000 [r218582] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_followme.c, /: Merged revisions 218579 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009)
+ | 16 lines Merged revisions 218577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009)
+ | 9 lines Ensure FollowMe sets language in channels it creates.
+ Also, not in the original bug report, but related fields are
+ accountcode and musicclass, and the inheritance of datastores.
+ (closes issue #15372) Reported by: Romik Patches:
+ 20090828__issue15372.diff.txt uploaded by tilghman (license 14)
+ Tested by: cervajs ........ ................
+
+2009-09-15 15:59 +0000 [r218576] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 218430 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218430 | jpeeler | 2009-09-14 17:38:25 -0500
+ (Mon, 14 Sep 2009) | 18 lines Merged revisions 218401 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009)
+ | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent
+ crash in do_monitor. After talking to rmudgett about some of his
+ recent iflist locking changes, it was determined that the only
+ place that would destroy a channel without being explicitly to do
+ so was in handle_init_event. The loop to walk the interface list
+ has been modified to wait to destroy the channel until the
+ dahdi_pvt of the channel to be destroyed is no longer needed.
+ (closes issue #15378) Reported by: samy ........ ................
+
+2009-09-15 15:42 +0000 [r218507-218575] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 |
+ mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4
+ lines Use a better method of ensuring null-termination of the
+ buffer while reading the SDP when using TCP. ........
+
+ * /, channels/chan_sip.c: Merged revisions 218499,218504 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue,
+ 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent
+ over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53
+ -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP
+ socket is null-terminated. ........
+
+2009-09-15 15:05 +0000 [r218503] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 218500 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep
+ 2009) | 9 lines Merged revisions 218497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep
+ 2009) | 1 line Use proper hostname for downloading sound files.
+ ........ ................
+
+2009-09-14 19:49 +0000 [r218364] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/Makefile, apps/app_voicemail.c, /,
+ configs/voicemail.conf.sample: Merged revisions 218361 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218361 | tilghman | 2009-09-14 14:29:48 -0500
+ (Mon, 14 Sep 2009) | 11 lines Recorded merge of revisions 218331
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009)
+ | 4 lines Don't say "Please try again" if we don't give the user
+ another chance to try again. (issue #15055, SWP-129) Reported by:
+ jthurman ........ ................
+
+2009-09-14 18:18 +0000 [r218300] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 218295 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218295 |
+ file | 2009-09-14 13:16:39 -0500 (Mon, 14 Sep 2009) | 2 lines Do
+ not attempt to add a parking extension if an error occurred while
+ reading the configuration. ........
+
+2009-09-14 15:20 +0000 [r218238] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 218224 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500
+ (Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep
+ 2009) | 8 lines Ensure we don't pickup ourselves when doing
+ pickup by exten. (closes issue #15100) Reported by: lmsteffan
+ Patches: (modified) pickup.patch uploaded by lmsteffan (license
+ 779) ........ ................
+
+2009-09-13 22:12 +0000 [r218219] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_phone.c, /: gcc 4.4: Remove a nop memset size 0
+ that annoys gcc This memset doesn't write beyond the end of the
+ buffer. (tmpbuf has size of 4). Merged revisions 218184 via
+ svnmerge from http://svn.digium.com/svn/asterisk/trunk
+
+2009-09-13 05:59 +0000 [r218151] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 218150 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r218150 | moy | 2009-09-13 01:51:46 -0400 (Sun, 13 Sep 2009) | 1
+ line get rid of mfcr2 monitor thread condition, is problematic
+ ........
+
+2009-09-11 06:00 +0000 [r217926-218055] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 218050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218050 |
+ tilghman | 2009-09-11 00:58:11 -0500 (Fri, 11 Sep 2009) | 3 lines
+ Check the origination priority for more matches, not the current
+ priority. Found by Pavel Troller on the -dev list. ........
+
+ * apps/app_queue.c, /: Merged revisions 217990 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009)
+ | 10 lines Merged revisions 217989 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009)
+ | 3 lines Don't ring another channel, if there's not enough time
+ for a queue member to answer. (Fixes AST-228) ........
+ ................
+
+ * channels/chan_iax2.c, contrib/scripts/iax-friends.sql, /,
+ channels/chan_sip.c: Merged revisions 217916 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 |
+ tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines
+ Make calltoken support work with realtime users and peers.
+ ........
+
+2009-09-10 21:21 +0000 [r217821] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 217807 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500
+ (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009)
+ | 22 lines IAX2 encryption regression The IAX2 Call Token
+ security patch inadvertently broke the use of encryption due to
+ the reorganization of code in the socket_process() function. When
+ encryption is used, an incoming full frame must first be
+ decrypted before the information elements can be parsed. The
+ security release mistakenly moved IE parsing before decryption in
+ order to process the new Call Token IE. To resolve this,
+ decryption of full frames is once again done before looking into
+ the frame. This involves searching for an existing callno,
+ checking the pvt to see if encryption is turned on, and
+ decrypting the packet before the internal fields of the full
+ frame are accessed. (closes issue #15834) Reported by: karesmakro
+ Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel
+ (license 671) Tested by: dvossel, karesmakro Review:
+ https://reviewboard.asterisk.org/r/355/ ........ ................
+
+2009-09-10 19:56 +0000 [r217739] mnick <mnick@localhost>:
+
+ * res/res_musiconhold.c, /: Merged revisions 217730 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r217730 | mnick | 2009-09-10 14:39:41 -0500 (Thu, 10 Sep 2009) |
+ 17 lines Sets the correct musicclass after an announcement
+ (closes issue #15279) Reported by: mbeckwell Patches: patch.txt
+ uploaded by mnick (license ) Tested by: mnick (closes issue
+ #15832) Reported by: mbeckwell Patches: patch.txt uploaded by
+ mnick (license 874) Tested by: mnick ........
+
+2009-09-10 18:40 +0000 [r217665] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 216805 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216805 |
+ oej | 2009-09-07 18:08:08 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
+ Since it's possible to have more than 999 calls, I'm changing the
+ call counter roof to something higher. ........
+
+2009-09-10 18:19 +0000 [r217647] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_odbc.c, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 217638 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217638 |
+ tilghman | 2009-09-10 13:17:14 -0500 (Thu, 10 Sep 2009) | 4 lines
+ Verify support for wide ODBC character types before using them.
+ (closes issue #15870) Reported by: nic_bellamy ........
+
+2009-09-10 15:14 +0000 [r217632] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 217524 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r217524 | moy | 2009-09-09 17:48:04 -0400 (Wed, 09 Sep 2009) | 1
+ line ast_log replaced for ast_verbose in MFCR2 event
+ notifications ........
+
+2009-09-10 12:09 +0000 [r217594] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 217593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 |
+ oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines
+ Include ActionID in all events that are responsed to AMI Action
+ SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy
+ Patches: manager_SIPshowregistry_actionid.patch uploaded by nic
+ bellamy (license 299) ........
+
+2009-09-09 20:37 +0000 [r217519] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc
+ 4.4 has more strict rules for aliasing. It doesn't like a struct
+ sockaddr_in pointer pointing to a struct sockaddr. So we make it
+ a union. Merged revisions 217445 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk
+
+2009-09-09 10:58 +0000 [r217369] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 217368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 |
+ oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not
+ having any TLS session to write to is a serious XMIT_ERROR.
+ ........
+
+2009-09-08 22:20 +0000 [r217299] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_meetme.c: Merged revisions 217286 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217286 |
+ seanbright | 2009-09-08 18:17:08 -0400 (Tue, 08 Sep 2009) | 4
+ lines Fix compilation of app_meetme. Reported by ebroad in
+ #asterisk-bugs ........
+
+2009-09-08 20:33 +0000 [r217217] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 217199 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009)
+ | 14 lines Merged revisions 217156 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009)
+ | 7 lines When MOH is playing on the channel, announcements sent
+ through the conference are not heard. (closes issue #14588)
+ Reported by: voipas Patches: 20090716__issue14588__2.diff.txt
+ uploaded by tilghman (license 14) Tested by: lmadsen, twisted,
+ tilghman ........ ................
+
+2009-09-08 16:39 +0000 [r217077] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 217074 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 |
+ kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9
+ lines Ensure that the default autoconf CFLAGS are not used. A
+ recent change to the configure script that allows the user to
+ specify CFLAGS and/or LDFLAGS to the script had the unfortunate
+ side effect of letting autoconf's default CFLAGS (-g -O2) feed in
+ to the rest of the build system, thereby overriding the
+ DONT_OPTIMIZE setting in menuselect. That problem is now
+ corrected. ........
+
+2009-09-08 15:36 +0000 [r217036] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_limit.c: Merged revisions 217033 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217033 |
+ tilghman | 2009-09-08 10:30:18 -0500 (Tue, 08 Sep 2009) | 4 lines
+ Remove what appears to be an unnecessary define. (closes issue
+ #15851) Reported by: tzafrir ........
+
+2009-09-08 14:27 +0000 [r216994] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 216993 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 |
+ dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines
+ caller id number empty parse_uri was not being given the correct
+ scheme's, as a result, uri parsing did not parse the username
+ correctly. One of the side effects of this is an empty caller id.
+ (closes issue #15839) Reported by: ebroad Patches:
+ blank_cidv2.patch uploaded by ebroad (license 878)
+ parse_uri_fix.diff uploaded by dvossel (license 671) Tested by:
+ ebroad, dvossel ........
+
+2009-09-07 16:43 +0000 [r216647-216845] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 216842 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 |
+ oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
+ Make sure we reset global_exclude_static at channel reload
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 216695 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 |
+ oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If
+ there is no session timer in the INVITE, set it to default value
+ (not unset minimum = -1) Patch by oej closes issue #15621
+ Reported by: fnordian Tested by: atis ........
+
+ * CHANGES, UPGRADE.txt: Add docs
+
+ * configs/sip.conf.sample, apps/app_playback.c, main/pbx.c, /,
+ channels/chan_sip.c, apps/app_disa.c: Merged revisions 216438 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre,
+ 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27
+ lines Make apps send PROGRESS control frame for early media and
+ fix too early media issue in SIP The issue at hand is that some
+ legacy (dying) PBX systems send empty media frames on PRI links
+ *before* any call progress. The SIP channel receives these frames
+ and by default signals 183 Session progress and starts sending
+ media. This will cause phones to play silence and ignore the
+ later 180 ringing message. A bad user experience. The fix is
+ twofold: - We discovered that asterisk apps that support early
+ media ("noanswer") did not send any PROGRESS frame to indicate
+ early media. Fixed. - We introduce a setting in chan_sip so that
+ users can disable any relay of media frames before the outbound
+ channel actually indicates any sort of call progress. In 1.4,
+ 1.6.0 and 1.6.1, this will be disabled for backward
+ compatibility. In later versions of Asterisk, this will be
+ enabled. We don't assume that it will change your Asterisk phone
+ experience - only for the better. We encourage third-party
+ application developers to make sure that if they have
+ applications that wants to send early media, add a PROGRESS
+ control frame transmission to make sure that all channel drivers
+ actually will start sending early media. This has not been the
+ default in Asterisk previous to this patch, so if you got
+ inspiration from our code, you need to update accordingly. Sorry
+ for the trouble and thanks for your support. This code has been
+ running for a few months in a large scale installation (over 250
+ servers with PRI and/or BRI links to old PBX systems). That's no
+ proof that this is an excellent patch, but, well, it's tested :-)
+ ........ ................
+
+2009-09-04 19:42 +0000 [r216598] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 216594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216594 |
+ dvossel | 2009-09-04 14:32:07 -0500 (Fri, 04 Sep 2009) | 7 lines
+ sip peer matching by address only with TCP/TLS This patch removes
+ the contact header matching logic and adds logic to match all
+ tcp/tls connections by ip only Review:
+ https://reviewboard.asterisk.org/r/354/ ........
+
+2009-09-04 19:32 +0000 [r216597] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 216593 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep
+ 2009) | 1 line Use ast_free() instead of free(). ........
+
+2009-09-04 17:53 +0000 [r216550-216553] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 216551 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r216551 | tilghman | 2009-09-04 12:50:21 -0500 (Fri, 04 Sep 2009)
+ | 2 lines Fix trunk breakage. ........
+
+ * UPGRADE-1.6.txt, main/pbx.c, /: Merged revisions 216547 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04
+ Sep 2009) | 3 lines Enable turning off the application delimiter
+ warning with the 'dontwarn' option. Suggested on the -dev list,
+ and implemented in an alternate way by me. ........
+
+2009-09-04 15:11 +0000 [r216469-216509] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, main/utils.c: Merged revisions 216506 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009)
+ | 9 lines Merged revisions 216435 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009)
+ | 2 lines make asterisk compile under devmode with DEBUG_THREADS
+ enabled on OpenBSD ........ ................
+
+ * /, include/asterisk/lock.h: Merged revisions 216437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r216437 | mvanbaak | 2009-09-04 16:00:38 +0200 (Fri, 04 Sep 2009)
+ | 2 lines make sure canlog is set so we can compile with
+ DEBUG_THREADS enabled on OpenBSD ........
+
+2009-09-04 13:57 +0000 [r216267-216436] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 216368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216368 |
+ russell | 2009-09-04 08:14:25 -0500 (Fri, 04 Sep 2009) | 12 lines
+ Do not treat every SIP peer as if they were configured with
+ insecure=port. There was a problem in the function responsible
+ for doing peer matching by IP address and port number such that
+ during the second pass for checking for a peer configured with
+ insecure=port, it would end up treating every peer as if it had
+ been configured that way. These changes fix the logic in the peer
+ IP and port comparison callback to handle insecure=port checking
+ properly. This problem was introduced when SIP peers were
+ converted to astobj2. Many thanks to dvossel for noticing this
+ while working on another peer matching issue. ........
+
+ * doc/IAX2-security.txt (added), /: Merged revisions 216264 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216264 | russell | 2009-09-04 05:48:44 -0500
+ (Fri, 04 Sep 2009) | 16 lines Merged revisions 216263 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216263 | russell | 2009-09-04 05:48:00 -0500
+ (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04
+ Sep 2009) | 2 lines Add a plain text version of the IAX2 security
+ document. ........ ................ ................
+
+2009-09-04 06:14 +0000 [r216225] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/astobj2.c, /: Merged revisions 216222 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216222 |
+ mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines
+ make sure 'start' is always initialized. Makes asterisk compile
+ with --enable-dev-mode ........
+
+2009-09-03 19:44 +0000 [r216014-216099] Russell Bryant <russell@digium.com>
+
+ * /, UPGRADE.txt: Merged revisions 216092 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009)
+ | 16 lines Merged revisions 216085 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216085 | russell | 2009-09-03 14:36:46 -0500
+ (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03
+ Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt.
+ ........ ................ ................
+
+ * /, doc/IAX2-security.pdf (added): Merged revisions 216009 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216009 | russell | 2009-09-03 13:45:54 -0500
+ (Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216008 | russell | 2009-09-03 13:44:58 -0500
+ (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03
+ Sep 2009) | 2 lines Add IAX2 security document related to
+ AST-2009-006. ........ ................ ................
+
+2009-09-03 18:42 +0000 [r216007] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, channels/iax2-parser.c, main/astobj2.c,
+ configs/iax.conf.sample, include/asterisk/acl.h,
+ channels/iax2-parser.h, /, include/asterisk/astobj2.h,
+ channels/iax2.h, main/acl.c: Merged revisions 215955 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009)
+ | 6 lines Merge code associated with AST-2009-006 (closes issue
+ #12912) Reported by: rathaus Tested by: tilghman, russell,
+ dvossel, dbrooks ........
+
+2009-09-03 14:21 +0000 [r215887-215929] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 215891 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215891 |
+ oej | 2009-09-03 15:02:41 +0200 (Tor, 03 Sep 2009) | 10 lines Add
+ known internal IP address when autodomain=yes (closes issue
+ #14573) Reported by: pj Patches: sip-internip-autodomain1.diff
+ uploaded by mnicholson (license 96) modified by oej Tested by: pj
+ ........
+
+ * main/rtp.c, channels/chan_sip.c: Fix bad reports in "sip show
+ channelstats". Not directly mergeable in svn trunk, needs more
+ tests, therefore committed directly to 1.6.2. (closes issue
+ #15819) Reported by: klaus3000 Patches:
+ asterisk-1.6.2-beta4-sipshowchannelstats-patch-0.2.txt uploaded
+ by klaus3000 (license 65) Tested by: klaus3000, oej
+
+2009-09-03 06:02 +0000 [r215841] Michiel van Baak <michiel@vanbaak.info>
+
+ * doc/manager_1_1.txt, /: Merged revisions 215838 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215838 |
+ mvanbaak | 2009-09-03 07:57:23 +0200 (Thu, 03 Sep 2009) | 5 lines
+ Document that SIPshowpeer and SKINNYshowline now include the
+ configured parkinglot in their response. Prodded by snuff-work on
+ #asterisk-dev IRC channel ........
+
+2009-09-03 03:44 +0000 [r215802] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215801 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215801 |
+ tilghman | 2009-09-02 22:43:51 -0500 (Wed, 02 Sep 2009) | 5 lines
+ Default the callback extension to "s". This is a regression.
+ (closes issue #15764) Reported by: elguero Change-type: bugfix
+ ........
+
+2009-09-03 00:34 +0000 [r215795] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215758 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009)
+ | 25 lines Merged revisions 215682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009)
+ | 18 lines Re-send non-100 provisional responses to prevent
+ cancellation From section 13.3.1.1 of RFC 3261: If the UAS
+ desires an extended period of time to answer the INVITE, it will
+ need to ask for an "extension" in order to prevent proxies from
+ canceling the transaction. A proxy has the option of canceling a
+ transaction when there is a gap of 3 minutes between responses in
+ a transaction. To prevent cancellation, the UAS MUST send a
+ non-100 provisional response at every minute, to handle the
+ possibility of lost provisional responses. (closes issue #11157)
+ Reported by: rjain Tested by: twilson Review:
+ https://reviewboard.asterisk.org/r/315/ ........ ................
+
+2009-09-02 21:46 +0000 [r215683] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215681 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215681 |
+ dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines
+ port string to int conversion using sscanf There are several
+ instances where a port is parsed from a uri or some other source
+ and converted to an int value using atoi(), if for some reason
+ the port string is empty, then a standard port is used. This
+ logic is used over and over, so I created a function to handle it
+ in a safer way using sscanf(). ........
+
+2009-09-02 21:37 +0000 [r215647-215680] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, channels/chan_sip.c, channels/chan_skinny.c: Merged revisions
+ 215665 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215665 |
+ mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines
+ add Parkinglot info to sip show peer <foo> and skinny show line
+ <foo> If we had this from the start, debugging the 'parking not
+ using configured parkinglot' bug would have been easier. ........
+
+ * /, main/features.c: Merged revisions 215622 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215622 |
+ mvanbaak | 2009-09-02 22:21:51 +0200 (Wed, 02 Sep 2009) | 4 lines
+ - lock channel before looking for a channel variable - Init the
+ parkings list member of struct parkinglot. Thanks Sean for the
+ explanation why this should be here. ........
+
+2009-09-02 18:52 +0000 [r215569-215570] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/Makefile, main/app.c: Merged revisions 215567 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r215567 | tilghman | 2009-09-02 13:37:25 -0500 (Wed, 02
+ Sep 2009) | 9 lines Close up to the soft open file limit (same on
+ Linux, but varies drastically on OS X). Also, a Makefile fix for
+ Darwin (OS X). (closes issue #14542) Reported by: jtodd Patches:
+ 20090901__issue14542.diff.txt uploaded by tilghman (license 14)
+ Tested by: jtodd, tilghman Change-type: bugfix ........
+
+ * /, channels/chan_sip.c: Merged revisions 215222 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215222 |
+ tilghman | 2009-09-01 16:19:40 -0500 (Tue, 01 Sep 2009) | 3 lines
+ Fix register such that lines with a transport string, but without
+ an authuser, parse correctly. (AST-228) ........
+
+2009-09-02 17:35 +0000 [r215523] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215522 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215522 |
+ dvossel | 2009-09-02 12:26:40 -0500 (Wed, 02 Sep 2009) | 11 lines
+ SIP uri parsing cleanup Now, the scheme passed to parse_uri can
+ either be a single scheme, or a list of schemes ',' delimited.
+ This gets rid of the whole problem of having to create two
+ buffers and calling parse_uri twice to check for separate
+ schemes. Review: https://reviewboard.asterisk.org/r/343/ ........
+
+2009-09-02 16:35 +0000 [r215512] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, channels/chan_skinny.c: Merged revisions 215479 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r215479 | mvanbaak | 2009-09-02 18:20:23 +0200 (Wed, 02 Sep 2009)
+ | 3 lines like in chan_sip's sip_new skinny should copy the
+ configured parkinglot from a line to the newly created channel.
+ This makes callparking honor the configured parkinglot for skinny
+ lines as well. ........
+
+2009-09-02 16:09 +0000 [r215467] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215466 |
+ dvossel | 2009-09-02 11:08:00 -0500 (Wed, 02 Sep 2009) | 16 lines
+ SIP support for keep-alive event keep-alive events are used by
+ Sipura/Linksys for NAT keepalive. There currently don't appear to
+ be any problems with NAT, but everytime a keep-alive event is
+ received, Asterisk responds with a "489 Bad event". This error
+ may indicate to a user that NAT problems exist just because this
+ even is not supported. Now, rather than respond with an error,
+ the packet is consumed and a "200 ok" is sent just to indicate we
+ received the packet. (issue #15084) Patches:
+ chan_sip.keepalive.v1.diff uploaded by IgorG (license 20)
+ ........
+
+2009-09-02 16:07 +0000 [r215465] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, channels/chan_sip.c: Merged revisions 215462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215462 |
+ mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12
+ lines Honor configured parkinglot when parking and retrieving
+ parked calls Thank oej for pointing out the fact that sip_new did
+ not copy parkinglot from the peer into the newly created channel.
+ (closes issue #15538) Reported by: gracedman Patches:
+ 2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak
+ (license 7) With mod by me to also fix callparking as well (this
+ uploaded patch only fixed retrieving a parked call) Tested by:
+ gracedman, mvanbaak ........
+
+2009-09-02 01:49 +0000 [r215376] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * /, apps/app_softhangup.c: Merged revisions 215338 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r215338 | dhubbard | 2009-09-01 20:16:59 -0500
+ (Tue, 01 Sep 2009) | 18 lines Merged revisions 215270 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009)
+ | 12 lines Use strrchr() so SoftHangup will correctly truncate
+ multi-hyphen channel names In general channel names are in the
+ form Foo/Bar-Z, but the channel name could have multiple hyphens
+ and look like Foo/B-a-r-Z. Use strrchr to truncate the channel
+ name at the last hyphen. (closes issue #15810) Reported by:
+ dhubbard Patches: dw-softhangup-1.4.patch uploaded by dhubbard
+ (license 733) ........ ................
+
+2009-09-01 20:00 +0000 [r215165] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/frame.c, /: Merged revisions 215161 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215161 |
+ kpfleming | 2009-09-01 14:50:48 -0500 (Tue, 01 Sep 2009) | 3
+ lines Ensure that frame dumps of AST_CONTROL_T38_PARAMETERS
+ frames are properly decoded. ........
+
+2009-08-31 16:22 +0000 [r214822-214960] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 214945 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r214945 | tilghman | 2009-08-31 11:18:33 -0500
+ (Mon, 31 Aug 2009) | 14 lines Merged revisions 214940 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 Aug 2009)
+ | 7 lines Also unlock the "other" channel, when returning, due to
+ glare. (closes issue #15787) Reported by: tim_ringenbach Patches:
+ chan_local.diff uploaded by tim ringenbach (license 540) Tested
+ by: tim_ringenbach ........ ................
+
+ * Makefile, /: Merged revisions 214898 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214898 |
+ tilghman | 2009-08-30 17:10:35 -0500 (Sun, 30 Aug 2009) | 2 lines
+ Force Darwin on ppc platforms to compile with a target level that
+ supports aliasing. ........
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/pbx_lua.c: Merged revisions 214819 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214819 |
+ tilghman | 2009-08-30 01:43:04 -0500 (Sun, 30 Aug 2009) | 4 lines
+ If lua is detected with the lua5.1 prefix (or not), adjust the
+ include path accordingly. Based upon feedback to a release
+ announcement on the -users list. See
+ http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html
+ ........
+
+2009-08-29 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.0-rc1 released.
+
+2009-08-28 20:17 +0000 [r214707] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 214702 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214702 | tilghman | 2009-08-28 15:14:39 -0500 (Fri, 28 Aug 2009)
+ | 15 lines Merged revisions 214701 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009)
+ | 8 lines Modify comment to be a bit more accurate. We have kept
+ this comment around long enough, that it's pretty clear that
+ we're keeping the code, because changing the code would require a
+ pretty fundamental architectural shift. We've also taken
+ criticism in some quarters, because it was believed that it was
+ referring to the code being nasty. No, the code isn't nasty, just
+ the operation itself is rather odd. Fixed for eternity (probably
+ not). ........ ................
+
+2009-08-28 20:05 +0000 [r214700] Kevin P. Fleming <kpfleming@digium.com>
+
+ * makeopts.in, Makefile, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 214696 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214696 |
+ kpfleming | 2009-08-28 15:01:21 -0500 (Fri, 28 Aug 2009) | 9
+ lines Ensure that CFLAGS and/or LDFLAGS provided to configure
+ script are preserved. Cross-compilation environments want to
+ provide 'defaults' for compiler and linker options, and
+ frequently do this by specifying CFLAGS and LDFLAGS in the
+ environment or as command-line arguments to the configure script.
+ This patch modifies the configure script and Makefile to preserve
+ these settings and ensure they are used in the build process.
+ ........
+
+2009-08-28 18:43 +0000 [r214653] Mark Michelson <mmichelson@digium.com>
+
+ * /, include/asterisk/sched.h: Merged revisions 214650 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r214650 | mmichelson | 2009-08-28 13:41:23 -0500 (Fri, 28 Aug
+ 2009) | 3 lines Fix some incorrect documentation of sched_thread
+ functions. ........
+
+2009-08-27 21:49 +0000 [r214202-214521] Tilghman Lesher <tlesher@digium.com>
+
+ * autoconf/libcurl.m4 (added), /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 214518 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214518 | tilghman | 2009-08-27 16:46:46 -0500 (Thu, 27 Aug 2009)
+ | 14 lines Merged revisions 214517 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009)
+ | 7 lines Use autoconf to detect libcurl, as this enables
+ cross-compilation checks, something we didn't allow before.
+ (closes issue #15714) Reported by: pprindeville Patches:
+ 20090813__issue15714.diff.txt uploaded by tilghman (license 14)
+ Tested by: pprindeville ........ ................
+
+ * main/manager.c, /: Merged revisions 214514 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214514 |
+ tilghman | 2009-08-27 16:26:37 -0500 (Thu, 27 Aug 2009) | 7 lines
+ Ensure that we check for the special value
+ CONFIG_STATUS_FILEINVALID. (closes issue #15786) Reported by:
+ a_villacis Patches:
+ asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch
+ uploaded by a villacis (license 660) (Plus a few of my own, to
+ catch the remaining places within manager.c where it could have
+ been a problem) ........
+
+ * autoconf/ast_ext_lib.m4, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 214466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214466 | tilghman | 2009-08-27 12:28:01 -0500 (Thu, 27 Aug 2009)
+ | 9 lines Merged revisions 214436 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009)
+ | 2 lines One more build system change, to make the descriptions
+ look better, if we have better information. ........
+ ................
+
+ * autoconf/ast_ext_lib.m4, /, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 214360 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r214360 | tilghman | 2009-08-27 11:12:03 -0500
+ (Thu, 27 Aug 2009) | 10 lines Merged revisions 214357 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009)
+ | 3 lines Make autoheader descriptions render correctly in our
+ autoconfig.h file. (Figured out while working with issue #14906)
+ ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 214199 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214199 |
+ tilghman | 2009-08-26 11:53:03 -0500 (Wed, 26 Aug 2009) | 6 lines
+ Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) (closes issue
+ #15362) Reported by: klaus3000 Patches:
+ chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license
+ 65) ........
+
+2009-08-26 16:39 +0000 [r214196] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 214195 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214195 | dvossel | 2009-08-26 11:38:53 -0500 (Wed, 26 Aug 2009)
+ | 25 lines Merged revisions 214194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009)
+ | 19 lines ast_write() ignores ast_audiohook_write() results In
+ ast_write(), if a channel has a list of audiohooks, those lists
+ are written to and the resulting frame is what ast_write() should
+ continue with. The problem was the returned audiohook frame was
+ not being handled at all, and the original frame passed into it
+ did not contain the mixed audio, so essentially audio was being
+ lost. One result of this was chan_spy's whisper mode no longer
+ worked. To complicate the issue, frames passed into ast_write may
+ either be a single frame, or a list of frames. So, as the list of
+ frames is processed in the audiohook_write, the returned frames
+ had to be added to a new list. (closes issue #15660) Reported by:
+ corruptor Tested by: dvossel ........ ................
+
+2009-08-25 22:43 +0000 [r213903-214155] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 214152 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214152 |
+ tilghman | 2009-08-25 17:39:51 -0500 (Tue, 25 Aug 2009) | 4 lines
+ Not all versions of gnu-linux use glibc, which contains iconv.
+ Some (especially embedded systems) don't have iconv at all.
+ (closes issue #15169) Reported by: pprindeville ........
+
+ * /, main/say.c: Merged revisions 214071 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214071 | tilghman | 2009-08-25 14:32:48 -0500 (Tue, 25 Aug 2009)
+ | 17 lines Merged revisions 214068-214069 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009)
+ | 6 lines Fix pronunciation of German dates. (closes issue
+ #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded
+ by Benjamin Kluck (license 803) ........ r214069 | tilghman |
+ 2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should
+ always compile before committing... ........ ................
+
+ * /, pbx/pbx_dundi.c: Merged revisions 213975 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213975 |
+ tilghman | 2009-08-25 01:51:12 -0500 (Tue, 25 Aug 2009) | 6 lines
+ DUNDILOOKUP function in 1.6 should use comma delimiters. (closes
+ issue #15322) Reported by: chappell Patches:
+ dundilookup-0015322.patch uploaded by chappell (license 8)
+ ........
+
+ * main/pbx.c, /: Merged revisions 213971 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r213971 | tilghman | 2009-08-25 01:35:37 -0500 (Tue, 25 Aug 2009)
+ | 14 lines Merged revisions 213970 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009)
+ | 7 lines Improve error message by informing user exactly which
+ function is missing a parethesis. (closes issue #15242) Reported
+ by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by
+ dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by
+ loloski (license 68) ........ ................
+
+ * Makefile, /: Merged revisions 213904 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213904 |
+ tilghman | 2009-08-24 21:54:07 -0500 (Mon, 24 Aug 2009) | 6 lines
+ The DTD should be installed in the same path as the rest of the
+ XML documentation. (closes issue #15344) Reported by: tzafrir
+ Patches: makefile_appdocs_dtd.diff uploaded by tzafrir (license
+ 46) ........
+
+ * Makefile, /: Merged revisions 213900 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r213900 | tilghman | 2009-08-24 21:41:17 -0500 (Mon, 24 Aug 2009)
+ | 11 lines Merged revisions 213899 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213899 | tilghman | 2009-08-24 21:40:22 -0500 (Mon, 24 Aug 2009)
+ | 4 lines Use the default runlevels for Debian derivatives,
+ instead of making up our own. (closes issue #14730) Reported by:
+ pkempgen ........ ................
+
+2009-08-24 16:49 +0000 [r213836] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 213833 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213833 | jpeeler | 2009-08-24 11:43:57 -0500 (Mon, 24 Aug 2009)
+ | 14 lines Fix storage of greetings when using IMAP_STORAGE The
+ store macro was not getting called preventing storage of IMAP
+ greetings at all. This has been corrected along with fixing
+ checking if the imapgreetings option is turned on to store the
+ greeting in IMAP. Lastly, the attachment filename was incorrectly
+ using the full path instead of just the basename, which was
+ causing problems with retrieval of the greeting. (closes issue
+ #14950) Reported by: noahisaac (closes issue #15729) Reported by:
+ lmadsen ........
+
+2009-08-24 04:54 +0000 [r213791] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 213790 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213790 | moy | 2009-08-24 00:46:28 -0400 (Mon, 24 Aug 2009) | 1
+ line improve handling of openr2_chan_disconnect_call API failure,
+ unlikely, but happened on openr2 library bug ........
+
+2009-08-21 22:54 +0000 [r213739] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 213738 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213738 |
+ tilghman | 2009-08-21 17:36:39 -0500 (Fri, 21 Aug 2009) | 2 lines
+ Clarifying comments in sip_register, and removing a dead section
+ ........
+
+2009-08-21 22:23 +0000 [r213721] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 213716 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213716 |
+ dvossel | 2009-08-21 17:22:11 -0500 (Fri, 21 Aug 2009) | 10 lines
+ Register request line contains wrong address when user domain and
+ register host differ (closes issue #15539) Reported by:
+ Nick_Lewis Patches: chan_sip.c-registraraddr.patch uploaded by
+ Nick (license 657) register_domain_fix_1.6.2 uploaded by dvossel
+ (license 671) Tested by: Nick_Lewis, dvossel ........
+
+2009-08-21 21:44 +0000 [r213698] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 213697 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213697 | kpfleming | 2009-08-21 16:39:51 -0500 (Fri, 21 Aug
+ 2009) | 12 lines Ensure that realtime mailboxes properly report
+ status on subscription. This patch modifies app_voicemail's
+ response to mailbox status subscriptions (via the internal event
+ system) to ensure that a subscription triggers an explicit poll
+ of the mailbox, so the subscriber can get an immediate cached
+ event with that status. Previously, the cache was only populated
+ with the status of non-realtime mailboxes. (closes issue #15717)
+ Reported by: natmlt ........
+
+2009-08-21 21:12 +0000 [r213636] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 213635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213635 |
+ dvossel | 2009-08-21 16:02:50 -0500 (Fri, 21 Aug 2009) | 5 lines
+ fixes sip register parsing when user@domain is used (issue
+ #15008) (issue #15672) ........
+
+2009-08-21 16:55 +0000 [r213563] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk.h, /: Merged revisions 213560 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r213560 | tilghman | 2009-08-21 11:53:52 -0500 (Fri, 21 Aug 2009)
+ | 14 lines Merged revisions 213559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009)
+ | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files.
+ (closes issue #15698) Reported by: slavon Patches:
+ 20090817__issue15698.diff.txt uploaded by tilghman (license 14)
+ Tested by: slavon, tilghman ........ ................
+
+2009-08-21 16:06 +0000 [r213497] Jason Parker <jparker@digium.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 213494 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r213494 | qwell | 2009-08-21 11:04:21 -0500
+ (Fri, 21 Aug 2009) | 12 lines Merged revisions 213493 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) |
+ 5 lines Clarify queues.conf comments to specify that variables
+ should be set in the dialplan. (closes issue #15755) Reported by:
+ trendboy ........ ................
+
+2009-08-21 04:25 +0000 [r213475-213481] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 213454 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213454 | moy | 2009-08-21 00:09:26 -0400 (Fri, 21 Aug 2009) | 1
+ line increment the mfcr2 monitor count when clearing the call
+ request ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 213216 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213216 | moy | 2009-08-19 23:26:59 -0400 (Wed, 19 Aug 2009) | 1
+ line fixed bug caused by calling ast_request without calling
+ ast_call on an R2 channel, ie, CHANISAVAIL ........
+
+2009-08-21 03:53 +0000 [r213453] Terry Wilson <twilson@digium.com>
+
+ * main/loader.c, /: Merged revisions 213450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213450 |
+ twilson | 2009-08-20 22:48:54 -0500 (Thu, 20 Aug 2009) | 2 lines
+ Make LOAD_ORDER actually work ........
+
+2009-08-20 21:50 +0000 [r213413] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 213404 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213404 | jpeeler | 2009-08-20 16:33:11 -0500 (Thu, 20 Aug 2009)
+ | 12 lines Fix greeting retrieval from IMAP Properly check for
+ the current voicemail state and if it doesn't exist, create it.
+ (closes issue #14597) Reported by: wtca Patches: 14597_v2.patch
+ uploaded by mmichelson (license 60) Tested by: jpeeler ........
+
+2009-08-20 20:37 +0000 [r213350] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 213327 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213327 |
+ mnicholson | 2009-08-20 15:29:32 -0500 (Thu, 20 Aug 2009) | 7
+ lines Fix a crash by checking the proper pointer for validity
+ before deferencing it. (closes issue #15751) Reported by: atis
+ Patches: ast_bridge_call_peer_cdr.patch uploaded by atis (license
+ 242) ........
+
+2009-08-19 22:41 +0000 [r213182] Jason Parker <jparker@digium.com>
+
+ * main/alaw.c, main/ulaw.c, /: Merged revisions 213179 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213179 | qwell | 2009-08-19 17:38:46 -0500 (Wed, 19 Aug 2009) |
+ 5 lines Fix compile when certain G711 menuselect options are
+ enabled. (closes issue #15697) Reported by: slavon ........
+
+2009-08-19 21:25 +0000 [r213128] David Vossel <dvossel@digium.com>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 213113 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r213113 | dvossel | 2009-08-19 16:21:00 -0500
+ (Wed, 19 Aug 2009) | 14 lines Merged revisions 213103 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009)
+ | 8 lines Fixes memory leak caused by incorrectly freeing
+ mixmonitor (closes issue #15699) Reported by: edantie Patches:
+ mixmonitor.patch uploaded by edantie (license 862) ........
+ ................
+
+2009-08-19 21:22 +0000 [r213095-213117] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 213098 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213098 |
+ tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines
+ Better parsing for the "register" line Allows characters that are
+ otherwise used as delimiters to be used within certain fields
+ (like the secret). (closes issue #15008, closes issue #15672)
+ Reported by: tilghman Patches: 20090818__issue15008.diff.txt
+ uploaded by tilghman (license 14) Tested by: lmadsen, tilghman
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 213093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213093 |
+ tilghman | 2009-08-19 15:29:41 -0500 (Wed, 19 Aug 2009) | 7 lines
+ If we have realtime caching enabled, 'sip reload' must purge
+ users/peers, even if the config files haven't changed. (closes
+ issue #12869) Reported by: bcnit Patches:
+ 20090819__issue12869__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: lasko ........
+
+2009-08-19 15:35 +0000 [r213047] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 213046 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213046 |
+ russell | 2009-08-19 10:32:18 -0500 (Wed, 19 Aug 2009) | 4 lines
+ Don't blow up on a NULL cdr. Reported in #asterisk-dev. ........
+
+2009-08-18 20:34 +0000 [r212931-212944] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /: Merged revisions 212939 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212939 |
+ kpfleming | 2009-08-18 15:33:34 -0500 (Tue, 18 Aug 2009) | 1 line
+ Remove some accidentally-committed properties. ........
+
+ * sounds/Makefile, doc/tex/asterisk.tex, CREDITS, /,
+ UPGRADE-1.4.txt, sounds/sounds.xml, build_tools/prep_tarball:
+ Merged revisions 212922 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212922 |
+ kpfleming | 2009-08-18 15:29:37 -0500 (Tue, 18 Aug 2009) | 6
+ lines Convert this branch to Opsound music-on-hold. For more
+ details:
+ http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
+ ........
+
+2009-08-18 19:28 +0000 [r212866] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/extconfig.conf.sample: Merged revisions 212857 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r212857 | tilghman | 2009-08-18 14:25:09 -0500 (Tue, 18
+ Aug 2009) | 4 lines Make the default extconfig.conf match entries
+ with the sample res_mysql.conf. This eliminates a future source
+ of possible confusion with the configuration of 1.6.1 and higher.
+ ........
+
+2009-08-18 16:56 +0000 [r212769] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, /: Merged revisions 212758 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r212758 | rmudgett | 2009-08-18 11:29:47 -0500
+ (Tue, 18 Aug 2009) | 9 lines Merged revisions 212727 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18
+ Aug 2009) | 1 line Removed some deadwood and added some doxygen
+ comments. ........ ................
+
+2009-08-18 16:41 +0000 [r212767] Sean Bright <sean@malleable.com>
+
+ * main/manager.c, /: Merged revisions 212764 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r212764 | seanbright | 2009-08-18 12:38:36 -0400 (Tue, 18 Aug
+ 2009) | 18 lines Merged revisions 212763 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug
+ 2009) | 11 lines Delay the creation of temporary files until we
+ have a valid manager command to handle. Without this patch,
+ asterisk creates a temporary file before determining if the
+ specified command is valid. If invalid, we weren't properly
+ cleaning up the file. (closes issue #15730) Reported by: zmehmood
+ Patches: M15730.diff uploaded by junky (license 177) Tested by:
+ zmehmood ........ ................
+
+2009-08-17 20:01 +0000 [r212631] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 212627 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r212627 | tilghman | 2009-08-17 14:57:42 -0500 (Mon, 17 Aug 2009)
+ | 4 lines Check the return value of opendir(3), or we may crash.
+ (closes issue #15720) Reported by: tobias_e ........
+
+2009-08-17 18:56 +0000 [r212580-212584] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_agent.c: Merged revisions 212581 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r212581 | seanbright | 2009-08-17 14:50:24 -0400 (Mon, 17 Aug
+ 2009) | 5 lines Correct spelling of AGENTACCEPTDTMF in
+ chan_agent. (closes issue #15668) Reported by: davidw ........
+
+ * main/logger.c: Merged revisions 212574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212574 |
+ seanbright | 2009-08-17 14:18:16 -0400 (Mon, 17 Aug 2009) | 8
+ lines Correct the return value check for ast_safe_system. The
+ logic here was reversed as ast_safe_system returns -1 on error
+ and not on success. Fix suggested by reporter. (closes issue
+ #15667) Reported by: loic ........
+
+2009-08-17 16:52 +0000 [r212509] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/misdn_config.c, /: Merged revisions 212506 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r212506 | jpeeler | 2009-08-17 11:50:45 -0500
+ (Mon, 17 Aug 2009) | 19 lines Merged revisions 212498 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009)
+ | 12 lines Fix segfault when reloading chan_misdn. If more ports
+ were specified than configured in misdn.conf a reload would crash
+ asterisk. The problem was the unconfigured port was using data
+ from the previously configured port. When the data for an
+ unconfigured port was freed a crash would result from the double
+ free. (closes issue #12113) Reported by: agupta Patches:
+ bug12113.patch uploaded by jpeeler (license 325) ........
+ ................
+
+2009-08-17 15:51 +0000 [r212434] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 212431 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r212431 | rmudgett | 2009-08-17 10:42:51 -0500
+ (Mon, 17 Aug 2009) | 16 lines Merged revisions 212430 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix
+ uninitialized variable causing random MWI indications. (closes
+ issue #15727) Reported by: doda Patches: dahdi_changes.patch
+ uploaded by doda (license 853) ........ r212430 | rmudgett |
+ 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix
+ uninitialized variable. ........ ................
+
+2009-08-14 17:37 +0000 [r212250] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_curl.c, /: Merged revisions 212249 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212249 |
+ tilghman | 2009-08-14 12:36:40 -0500 (Fri, 14 Aug 2009) | 2 lines
+ Add SSL_VERIFYPEER, as requested on the -users list ........
+
+2009-08-13 15:47 +0000 [r212116] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 212113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212113 |
+ kpfleming | 2009-08-13 10:46:25 -0500 (Thu, 13 Aug 2009) | 3
+ lines Ensure that T38FaxVersion is put into outgoing SDP in the
+ proper case. ........
+
+2009-08-13 13:56 +0000 [r212070] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 212067 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212067 |
+ file | 2009-08-13 10:51:04 -0300 (Thu, 13 Aug 2009) | 2 lines
+ Check an actual populated variable when seeing if we need to do
+ video or not. ........
+
+2009-08-13 11:47 +0000 [r212030] Gavin Henry <ghenry@suretecsystems.com>
+
+ * contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif, /: Merged revisions 212027 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r212027 | ghenry | 2009-08-13 12:37:12 +0100 (Thu, 13
+ Aug 2009) | 6 lines Fixed typo (closes issue #15710) Reported by:
+ suretec ........
+
+2009-08-12 23:16 +0000 [r211951-211959] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 211957 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211957 | mnicholson | 2009-08-12 18:14:36 -0500 (Wed, 12 Aug
+ 2009) | 17 lines Merged revisions 211953 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug
+ 2009) | 10 lines This patch adds additional checking when
+ generating queue log TRANSFER events. The additional checks
+ prevent generation of false TRANSFER events in certain
+ situations. (closes issue #14536) Reported by: aragon Patches:
+ queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: aragon, mnicholson ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 211876 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211876 |
+ mnicholson | 2009-08-12 14:53:14 -0500 (Wed, 12 Aug 2009) | 11
+ lines Make asterisk handle 423 Interval Too Short messages
+ better. This change uses separate values for the acceptable
+ minimum expiry provided by the 423 error and the expiry value
+ stored in the configuration file. Previously, the value pulled
+ from the configuration file would be overwritten. (closes issue
+ #14366) Reported by: Nick_Lewis Patches: sip-expiry-fix1.diff
+ uploaded by mnicholson (license 96) chan_sip.c-reqexpiry.patch
+ uploaded by Nick (license 657) Tested by: mnicholson ........
+
+2009-08-12 16:21 +0000 [r211785] Gavin Henry <ghenry@suretecsystems.com>
+
+ * res/res_config_ldap.c, contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif, /: Merged revisions 211767 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r211767 | ghenry | 2009-08-12 17:00:46 +0100 (Wed, 12
+ Aug 2009) | 33 lines Added three new attributes and applied a
+ patch to res_config_ldap.c attributetype (
+ AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC
+ 'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR
+ caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
+ attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC
+ 'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR
+ caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
+ attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent'
+ DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch
+ SUBSTR caseIgnoreSubstringsMatch SYNTAX
+ 1.3.6.1.4.1.1466.115.121.1.15) and patch
+ fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725)
+ Reported by: macogeek Patches:
+ fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license
+ 863) Tested by: suretec ........
+
+2009-08-10 19:51 +0000 [r211580-211585] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500
+ (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10
+ Aug 2009) | 1 line Conversion specifiers, not format specifiers
+ ........ ................
+
+ * apps/app_queue.c, apps/app_talkdetect.c, agi/eagi-sphinx-test.c,
+ res/res_config_curl.c, channels/chan_usbradio.c,
+ channels/chan_misdn.c, res/snmp/agent.c, apps/app_sms.c,
+ apps/app_verbose.c, apps/app_stack.c, apps/app_mixmonitor.c,
+ main/asterisk.c, main/dsp.c, main/timing.c,
+ doc/CODING-GUIDELINES, funcs/func_speex.c, main/frame.c,
+ utils/muted.c, apps/app_meetme.c, apps/app_alarmreceiver.c,
+ cdr/cdr_pgsql.c, res/res_musiconhold.c, channels/chan_iax2.c,
+ apps/app_followme.c, main/enum.c, main/indications.c,
+ res/res_config_sqlite.c, channels/misdn_config.c, utils/frame.c,
+ main/cli.c, pbx/pbx_loopback.c, channels/chan_phone.c,
+ funcs/func_enum.c, res/res_smdi.c, channels/chan_skinny.c,
+ funcs/func_odbc.c, apps/app_minivm.c, res/res_agi.c,
+ res/res_config_ldap.c, apps/app_adsiprog.c,
+ funcs/func_dialplan.c, main/pbx.c, main/dnsmgr.c,
+ funcs/func_sprintf.c, funcs/func_timeout.c, channels/chan_sip.c,
+ apps/app_privacy.c, res/res_limit.c, apps/app_waitforsilence.c,
+ codecs/codec_speex.c, agi/eagi-test.c, apps/app_morsecode.c,
+ funcs/func_cut.c, channels/chan_oss.c, main/netsock.c,
+ apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c,
+ pbx/pbx_dundi.c, utils/extconf.c, pbx/pbx_config.c,
+ apps/app_chanspy.c, res/res_odbc.c, apps/app_voicemail.c,
+ apps/app_dahdibarge.c, funcs/func_rand.c, apps/app_readfile.c, /,
+ apps/app_record.c, main/utils.c, cdr/cdr_adaptive_odbc.c,
+ res/res_http_post.c, main/config.c, res/ael/pval.c, main/cdr.c,
+ main/channel.c, channels/chan_dahdi.c, pbx/pbx_spool.c,
+ main/manager.c, apps/app_setcallerid.c, apps/app_osplookup.c,
+ main/features.c, main/http.c, channels/xpmr/xpmr.c,
+ apps/app_rpt.c, channels/chan_mgcp.c, res/res_config_pgsql.c,
+ channels/chan_agent.c, funcs/func_math.c, apps/app_waituntil.c,
+ apps/app_disa.c, main/acl.c, apps/app_originate.c,
+ channels/iax2-provision.c: AST-2009-005
+
+2009-08-10 14:15 +0000 [r211350] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 |
+ file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix
+ retrieval of the port used for the video stream when adding SDP
+ to a SIP message. (closes issue #15121) Reported by: jsmith
+ ........
+
+2009-08-09 15:43 +0000 [r211235-211278] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/astfd.c: Merged revisions 211275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009)
+ | 9 lines Merged revisions 211274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009)
+ | 2 lines Small oops. Clear the flags which have been checked.
+ ........ ................
+
+ * apps/app_stack.c, /: Merged revisions 211232 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 |
+ tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines
+ Check for NULL frame, before dereferencing pointer. (closes issue
+ #15617) Reported by: rain ........
+
+2009-08-07 20:18 +0000 [r211122] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009)
+ | 11 lines Recorded merge of revisions 211112 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009)
+ | 4 lines Resolve a deadlock involving app_chanspy and
+ masquerades. (ABE-1936) ........ ................
+
+2009-08-07 18:20 +0000 [r211051] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 211040 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009)
+ | 21 lines Merged revisions 211038 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009)
+ | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name,
+ not the membername. This is a partial revert of revision 82590,
+ which was an attempted cleanup, but in reality, it broke
+ QUEUE_MEMBER_LIST, which has always been intended as a method by
+ which component interfaces could be queried from the queue.
+ Membername isn't useful here, because that field cannot be used
+ to obtain further information about the member. See the
+ documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
+ QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
+ member argument for further justification. (closes issue #15664)
+ Reported by: rain Patches: app_queue-queue_member_list.diff
+ uploaded by rain (license 327) ........ ................
+
+2009-08-07 13:10 +0000 [r210995] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /: Merged revisions 210992 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 |
+ kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13
+ lines Workaround broken T.38 endpoints that offer tiny
+ MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as
+ the maximum IFP size that should be sent to them, rather than the
+ maximum packet payload size. If such an endpoint also requests
+ UDPRedundancy as the error correction mode, we'll end up
+ calculating a tiny maximum IFP size, so small as to be unusable.
+ This patch sets a lower bound on what we'll consider the remote's
+ maximum IFP size to be, assuming that endpoints that do this
+ really can accept larger packets than they've offered to accept.
+ (closes issue #15649) Reported by: dazza76 ........
+
+2009-08-06 21:47 +0000 [r210911-210917] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 210914 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009)
+ | 14 lines Merged revisions 210913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009)
+ | 7 lines Because channel information can be accessed outside of
+ the channel thread, we must lock the channel prior to modifying
+ it. (closes issue #15397) Reported by: caspy Patches:
+ 20090714__issue15397.diff.txt uploaded by tilghman (license 14)
+ Tested by: caspy ........ ................
+
+ * apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged
+ revisions 210908 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 |
+ tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines
+ Allow Gosub to recognize quote delimiters without consuming them.
+ (closes issue #15557) Reported by: rain Patches:
+ 20090723__issue15557.diff.txt uploaded by tilghman (license 14)
+ Tested by: rain Review: https://reviewboard.asterisk.org/r/316/
+ ........
+
+2009-08-06 17:49 +0000 [r210820] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 |
+ file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines
+ Accept additional T.38 reinvites after an initial one has been
+ handled. Discussion of this subject has yielded that it is not
+ actually acceptable to change T.38 parameters after the initial
+ reinvite but declining is harsh and can cause the fax to fail
+ when it may be possible to allow it to continue. This patch
+ changes things so that additional T.38 reinvites are accepted but
+ parameter changes ignored. This gives the fax a fighting chance.
+ (closes issue #15610) Reported by: huangtx2009 ........
+
+2009-08-05 20:43 +0000 [r210686] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500
+ (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009)
+ | 14 lines Dialplan starts execution before the channel setup is
+ complete. * Issue 15655: For the case where dialing is complete
+ for an incoming call, dahdi_new() was asked to start the PBX and
+ then the code set more channel variables. If the dialplan hungup
+ before these channel variables got set, asterisk would likely
+ crash. * Fixed potential for overlap incoming call to erroneously
+ set channel variables as global dialplan variables if the
+ ast_channel structure failed to get allocated. * Added missing
+ set of CALLINGSUBADDR in the dialing is complete case. (closes
+ issue #15655) Reported by: alecdavis ........ ................
+
+2009-08-05 18:56 +0000 [r210565-210566] Leif Madsen <lmadsen@digium.com>
+
+ * /: Merged revisions 210564 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210564 | lmadsen | 2009-08-05 13:49:58 -0500 (Wed, 05 Aug 2009)
+ | 19 lines Merged revisions 210563 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009)
+ | 11 lines Update imapstorage.txt documentation. Updated the
+ imapstorage.txt documentation to reflect that issues with
+ c-client versions older than 2007 seem to cause crashing issues
+ that are not seen with more recent versions. Documentation has
+ been updated to reflect this. (closes issue #14496) Reported by:
+ vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+ uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+ dbrooks ........ ................
+
+ * doc/tex/imapstorage.tex: Merged revisions 210564 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500
+ (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009)
+ | 11 lines Update imapstorage.txt documentation. Updated the
+ imapstorage.txt documentation to reflect that issues with
+ c-client versions older than 2007 seem to cause crashing issues
+ that are not seen with more recent versions. Documentation has
+ been updated to reflect this. (closes issue #14496) Reported by:
+ vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+ uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+ dbrooks ........ ................
+
+2009-08-04 14:55 +0000 [r210191-210241] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 210238 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug
+ 2009) | 16 lines Merged revisions 210237 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug
+ 2009) | 10 lines Eliminate spurious compiler warnings from system
+ headers on *BSD platforms. Ensure that system headers located in
+ /usr/local/include are actually treated as system headers by the
+ compiler, and not as local headers which are subject to warnings
+ from the -Wundef compiler option and others. (closes issue
+ #15606) Reported by: mvanbaak ........ ................
+
+ * configs/sip.conf.sample, configs/skinny.conf.sample, main/rtp.c,
+ channels/chan_mgcp.c, doc/chan_sip-perf-testing.txt,
+ contrib/scripts/realtime_pgsql.sql, /, channels/chan_sip.c,
+ channels/chan_skinny.c, configs/mgcp.conf.sample,
+ doc/res_config_sqlite.txt, doc/tex/phoneprov.tex, UPGRADE.txt,
+ configs/res_ldap.conf.sample: Merged revisions 210190 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03
+ Aug 2009) | 11 lines Rename 'canreinvite' option to
+ 'directmedia', with backwards compatibility. It is clear from
+ multiple mailing list, forum, wiki and other sorts of posts that
+ users don't really understand the effects that the 'canreinvite'
+ config option actually has, and that in some cases they think
+ that setting it to 'no' will actually cause various other
+ features (T.38, MOH, etc.) to not work properly, when in fact
+ this is not the case. This patch changes the proper name of the
+ option to what it should have been from the beginning
+ ('directmedia'), but preserves backwards compatibility for
+ existing configurations. ........
+
+2009-08-01 11:33 +0000 [r209837-209906] Russell Bryant <russell@digium.com>
+
+ * main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r209887 | russell | 2009-08-01 06:29:25 -0500
+ (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009)
+ | 5 lines Resolve a valgrind warning about a read from
+ uninitialized memory. (issue #15396) Reported by: aragon ........
+ ................
+
+ * apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r209839 | russell | 2009-08-01 06:02:07 -0500
+ (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009)
+ | 13 lines Modify how Playtones() is used in Milliwatt() to
+ resolve gain issue. When Milliwatt() was changed internally to
+ use Playtones() so that the proper tone was used, it introduced a
+ drop in gain in the output signal. So, use the playtones API
+ directly and specify a volume argument such that the output
+ matches the gain of the original Milliwatt() code. (closes issue
+ #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff
+ uploaded by russell (license 2) Tested by: rue_mohr ........
+ ................
+
+ * /, main/event.c: Merged revisions 209835 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209835 |
+ russell | 2009-08-01 05:43:40 -0500 (Sat, 01 Aug 2009) | 6 lines
+ Fix ast_event_queue_and_cache() to actually do the cache() part.
+ (closes issue #15624) Reported by: ffossard Tested by: russell
+ ........
+
+2009-08-01 01:34 +0000 [r209816] Kevin P. Fleming <kpfleming@digium.com>
+
+ * pbx/pbx_config.c, channels/misdn/isdn_lib.c, utils/frame.c,
+ main/pbx.c, /, main/Makefile, channels/misdn/ie.c: Merged
+ revisions 209760-209761 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209760 | kpfleming | 2009-07-31 20:03:07 -0500 (Fri, 31 Jul
+ 2009) | 13 lines Merged revisions 209759 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul
+ 2009) | 7 lines Minor changes inspired by testing with latest
+ GCC. The latest GCC (what will become 4.5.x) has a few new
+ warnings, that in these cases found some either downright buggy
+ code, or at least seriously poorly designed code that could be
+ improved. ........ ................ r209761 | kpfleming |
+ 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert
+ accidental Makefile change. ................
+
+2009-07-31 22:01 +0000 [r209715] Russell Bryant <russell@digium.com>
+
+ * /, main/event.c: Merged revisions 209711 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 |
+ russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines
+ Fix some places where ast_event_type was used instead of
+ ast_event_ie_type. ........
+
+2009-07-30 18:51 +0000 [r209594] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_console.c, include/asterisk/abstract_jb.h,
+ apps/app_forkcdr.c, channels/chan_dahdi.c,
+ contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c,
+ codecs/lpc10/pitsyn.c: Merged revisions 209554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 |
+ dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
+ Fixes numerous spelling errors. Patch submitted by alecdavis.
+ (closes issue #15595) Reported by: alecdavis ........
+
+2009-07-30 14:40 +0000 [r209518] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 209516 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 |
+ mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8
+ lines Fix a crash that can result if text codecs are allowed but
+ textsupport is disabled. (closes issue #15596) Reported by:
+ fabled Patches: sip-red.patch uploaded by fabled (license 448)
+ ........
+
+2009-07-28 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-beta4
+
+2009-07-28 00:19 +0000 [r209328] Tilghman Lesher <tlesher@digium.com>
+
+ * /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009)
+ | 9 lines Merged revisions 209315 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009)
+ | 2 lines Publish French extra sounds ........ ................
+
+2009-07-27 21:44 +0000 [r209265-209282] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 209279 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 |
+ kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7
+ lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE
+ messages about T.38 negotiation in debug level 1 messages, clean
+ up some looping logic, and correct an improper use of ast_free()
+ for freeing an ast_frame. ........
+
+ * /, apps/app_fax.c: Merged revisions 209256 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 |
+ kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10
+ lines Make T.38 switchover in ReceiveFAX synchronous. In receive
+ mode, if the channel that ReceiveFAX is running on supports T.38,
+ we should *always* attempt to switch T.38, rather than listening
+ for an incoming CNG tone and only triggering on that. The channel
+ may be using a low-bitrate codec that distorts the CNG tone, the
+ sending FAX endpoint may not send CNG at all, or there could be a
+ variety of other reasons that we don't detect it, but in all
+ those cases if T.38 is available we certainly want to use it.
+ ........
+
+2009-07-27 20:58 +0000 [r209238] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, /: Merged revisions 209235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 |
+ mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5
+ lines Gracefully handle malformed RTP text packets. AST-2009-004
+ ........
+
+2009-07-27 20:33 +0000 [r209234] David Brooks <dbrooks@digium.com>
+
+ * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c,
+ channels/chan_vpb.cc, res/res_smdi.c, /,
+ include/asterisk/module.h, main/features.c, res/res_agi.c: Merged
+ revisions 209098 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 |
+ dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines
+ Fixing typos. Replaces "recieved" with "received" and "initilize"
+ with "initialize" (closes issue #15571) Reported by: alecdavis
+ ........
+
+2009-07-27 20:23 +0000 [r209135-209222] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 209197 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul
+ 2009) | 9 lines Honor channel's music class when using realtime
+ music on hold. (closes issue #15051) Reported by: alexh Patches:
+ 15051.patch uploaded by mmichelson (license 60) Tested by: alexh
+ ........
+
+ * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
+ 209132 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul
+ 2009) | 24 lines Merged revisions 209131 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
+ 2009) | 18 lines Allow for UDPTL to use only even-numbered ports
+ if desired. There are some VoIP providers out there that will not
+ accept SDP offers with odd numbered UDPTL ports. While it is my
+ personal opinion that these VoIP providers are misinterpreting
+ RFC 2327, it really is not a big deal to play along with their
+ silly little games. Of course, since restricting UDPTL ports to
+ only even numbers reduces the range of available ports by half,
+ so the option to use only even port numbers is off by default. A
+ user can enable the behavior by setting use_even_ports=yes in
+ udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
+ 15182.patch uploaded by mmichelson (license 60) Tested by:
+ CGMChris ........ ................
+
+2009-07-27 16:07 +0000 [r209063] David Brooks <dbrooks@digium.com>
+
+ * apps/app_rpt.c, res/res_smdi.c, pbx/pbx_dundi.c: Just replacing
+ typos "recieved" with "received". From issue #15360, forgot to
+ apply to trunk and other branches.
+
+2009-07-27 15:40 +0000 [r209059] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 209056 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 |
+ kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10
+ lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and
+ underscore-variants to sub-makes. During the recent Makefile
+ improvements I made, it seemed the 'make' was automatically
+ carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so
+ I removed the explict export of them. However, there are some
+ circumstances where make does this, and some where it does not,
+ so I've brought them back to ensure they are always exported. I
+ also removed an extraneous double setting of _ASTLDFLAGS on *BSD
+ platforms. ........
+
+2009-07-27 01:23 +0000 [r208927] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_iax2.c, /, main/translate.c: Merged revisions
+ 208924 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009)
+ | 9 lines Merged revisions 208923 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009)
+ | 2 lines Fix logic errors from 208746 ........ ................
+
+2009-07-26 14:07 +0000 [r208889] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq, /: Merged revisions 208886 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r208886 | mvanbaak | 2009-07-26 16:00:52 +0200 (Sun, 26
+ Jul 2009) | 2 lines add OpenBSD to the install_prereq script
+ ........
+
+2009-07-25 12:31 +0000 [r208816-208853] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq, /: Merged revisions 208848 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r208848 | mvanbaak | 2009-07-25 14:28:38 +0200 (Sat, 25
+ Jul 2009) | 2 lines libxml2-dev is needed as well by default.
+ ........
+
+ * main/cli.c, /, configs/cli_aliases.conf.sample: Merged revisions
+ 208813 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208813 |
+ mvanbaak | 2009-07-25 14:03:25 +0200 (Sat, 25 Jul 2009) | 10
+ lines add default alias reload to run module reload. Requiring
+ 'module reload' to reload everything, including core etc makes
+ russell very unhappy. The default configuration already loads the
+ 'friendly' aliases template. Added 'reload=module reload' to that
+ template. Also removed the comment in main/cli.c that reload
+ should come back. ........
+
+2009-07-25 06:26 +0000 [r208755] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_iax2.c, /, channels/chan_skinny.c,
+ main/translate.c: Merged revisions 208749 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009)
+ | 13 lines Merged revisions 208746 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009)
+ | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly
+ trivial changes, but I did not know of any other way to fix the
+ "dereferencing type-punned pointer will break strict-aliasing
+ rules" error without creating a tmp variable in chan_skinny.
+ ........ ................
+
+2009-07-24 21:13 +0000 [r208695-208710] Russell Bryant <russell@digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 208709 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208709 |
+ russell | 2009-07-24 16:12:43 -0500 (Fri, 24 Jul 2009) | 2 lines
+ Remove trailing whitespace. ........
+
+ * main/cli.c, /: Merged revisions 208706 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208706 |
+ russell | 2009-07-24 15:54:37 -0500 (Fri, 24 Jul 2009) | 6 lines
+ Note that "reload" needs to be added back. I keep getting annoyed
+ at having to type "module reload" to reload everything, so I'm
+ adding a note that we need to add "reload" back. "module reload"
+ doesn't really make sense as the command to reload everything,
+ including the core. ........
+
+ * main/cli.c, /: Merged revisions 208693 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208693 |
+ russell | 2009-07-24 15:25:23 -0500 (Fri, 24 Jul 2009) | 2 lines
+ Don't log a warning for something that does not affect operation.
+ ........
+
+2009-07-24 19:42 +0000 [r208664] Mark Michelson <mmichelson@digium.com>
+
+ * /: Fixing trunk-blocked property.
+
+2009-07-24 18:56 +0000 [r208596] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 208593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009)
+ | 14 lines Merged revisions 208592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009)
+ | 7 lines Do not log an ERROR if autoservice_stop() returns -1.
+ This does not indicate an error. A return of -1 just means that
+ the channel has been hung up. (reported in #asterisk-dev)
+ ........ ................
+
+2009-07-24 18:32 +0000 [r208591] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul
+ 2009) | 16 lines Merged revisions 208587 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul
+ 2009) | 10 lines Only send a BYE when hanging up a channel that
+ is up. For cases where Asterisk sends an INVITE and receives a
+ non 2XX final response, Asterisk would follow the INVITE
+ transaction by immediately sending a BYE, which was unnecessary.
+ (closes issue #14575) Reported by: chris-mac ........
+ ................
+
+2009-07-24 15:06 +0000 [r208551] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
+ Merged revisions 208548 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 |
+ kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8
+ lines Resolve a T.38 negotiation issue left over from the
+ udptl-updates merge. The udptl-updates branch that was merged
+ yesterday failed to properly send back T.38 SDP responses with
+ the correct error correction mode, if the incoming SDP from the
+ other end caused us to change error correction modes. This patch
+ corrects that situation. ........
+
+2009-07-24 14:39 +0000 [r208545] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq, /: Merged revisions 208542 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r208542 | mvanbaak | 2009-07-24 16:35:49 +0200 (Fri, 24
+ Jul 2009) | 13 lines use aptitude for debian based systems The
+ function to check wether we need to install packages was using
+ dpkg-query which was gives wrong output on Debian 5 Also, the
+ apt-get has been replaced with aptitude because aptitude is now
+ the preferred way to handle packages on Debian (closes issue
+ #15570) Reported by: mvanbaak Patches:
+ 2009072400_installprereq-aptitude.diff uploaded by mvanbaak
+ (license 7) ........
+
+2009-07-23 22:31 +0000 [r208501] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/frame.h, main/rtp.c, main/channel.c,
+ main/udptl.c, main/frame.c, /, channels/chan_sip.c,
+ apps/app_fax.c, UPGRADE.txt, include/asterisk/udptl.h: Merged
+ revisions 208464 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 |
+ kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46
+ lines Rework of T.38 negotiation and UDPTL API to address
+ interoperability problems Over the past couple of months, a
+ number of issues with Asterisk negotiating (and successfully
+ completing) T.38 sessions with various endpoints have been found.
+ This patch attempts to address many of them, primarily focused
+ around ensuring that the endpoints' MaxDatagram size is honored,
+ and in addition by ensuring that T.38 session parameter
+ negotiation is performed correctly according to the ITU T.38
+ Recommendation. The major changes here are: 1) T.38 applications
+ in Asterisk (app_fax) only generate/receive IFP packets, they do
+ not ever work with UDPTL packets. As a result of this, they
+ cannot be allowed to generate packets that would overflow the
+ other endpoints' MaxDatagram size after the UDPTL stack adds any
+ error correction information. With this patch, the application is
+ told the maximum *IFP* size it can generate, based on a
+ calculation using the far end MaxDatagram size and the active
+ error correction mode on the T.38 session. The same is true for
+ sending *our* MaxDatagram size to the remote endpoint; it is
+ computed from the value that the application says it can accept
+ (for a single IFP packet) combined with the active error
+ correction mode. 2) All treatment of T.38 session parameters as
+ 'capabilities' in chan_sip has been removed; these parameters are
+ not at all like audio/video stream capabilities. There are strict
+ rules to follow for computing an answer to a T.38 offer, and
+ chan_sip now follows those rules, using the desired parameters
+ from the application (or channel) that wants to accept the T.38
+ negotiation. 3) chan_sip now stores and forwards
+ ast_control_t38_parameters structures for tracking 'our' and
+ 'their' T.38 session parameters; this greatly simplifies
+ negotiation, especially for pass-through calls. 4) Since T.38
+ negotiation without specifying parameters or receiving the final
+ negotiated parameters is not very worthwhile, the AST_CONTROL_T38
+ control frame has been removed. A note has been added to
+ UPGRADE.txt about this removal, since any out-of-tree
+ applications that use it will no longer function properly until
+ they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review:
+ https://reviewboard.asterisk.org/r/310/ ........
+
+2009-07-23 19:36 +0000 [r208391] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208388 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul
+ 2009) | 24 lines Merged revisions 208386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul
+ 2009) | 17 lines Fix a problem where a 491 response could be sent
+ out of dialog. This generalizes the fix for issue 13849. The
+ initial fix corrected the problem that Asterisk would reply with
+ a 491 if a reinvite were received from an endpoint and we had not
+ yet received an ACK from that endpoint for the initial INVITE it
+ had sent us. This expansion also allows Asterisk to appropriately
+ handle an INVITE with authorization credentials if Asterisk had
+ not received an ACK from the previous transaction in which
+ Asterisk had responded to an unauthorized INVITE with a 407.
+ (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
+ uploaded by mmichelson (license 60) Tested by: klaus3000 ........
+ ................
+
+2009-07-23 19:25 +0000 [r208387] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 208383 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r208383 | jpeeler | 2009-07-23 14:21:50 -0500
+ (Thu, 23 Jul 2009) | 12 lines Merged revisions 208380 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009)
+ | 6 lines Only set the priindication setting when not performing
+ a reload (closes issue #14696) Reported by: fdecher ........
+ ................
+
+2009-07-23 16:30 +0000 [r208266-208320] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208314 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul
+ 2009) | 9 lines Merged revisions 208312 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul
+ 2009) | 3 lines Remove inaccurate XXX comment. ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 208263 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul
+ 2009) | 15 lines Merged revisions 208262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul
+ 2009) | 8 lines Properly handle 183 responses which do not
+ contain an SDP. (closes issue #15442) Reported by: ffloimair
+ Patches: 15442.patch uploaded by mmichelson (license 60) Tested
+ by: tkarl, ffloimair ........ ................
+
+2009-07-22 21:46 +0000 [r208116] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_festival.c: Merged revisions 208113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208113 |
+ qwell | 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) | 9 lines
+ Restore an int declaration on PPC platforms. This x is one crafty
+ little bugger... It was used for 2 different things (one of which
+ was only done on PPC) in 1.4. One of the uses were removed in
+ trunk, and with it went the declaration. (closes issue #14038)
+ Reported by: ffloimair ........
+
+2009-07-22 16:52 +0000 [r207949-208053] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_realtime.c: Merged revisions 208052 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208052 |
+ tilghman | 2009-07-22 11:49:42 -0500 (Wed, 22 Jul 2009) | 7 lines
+ Clarify documentation on 'realtime update2' to show more than one
+ condition. (closes issue #15357) Reported by: snuffy Patches:
+ bug_fix_doc_update2.diff uploaded by snuffy (license 35)
+ (slightly modified by me) ........
+
+ * /, funcs/func_strings.c: Merged revisions 207946 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207946 | tilghman | 2009-07-21 17:45:32 -0500
+ (Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009)
+ | 8 lines Force an error if a blank is passed to QUOTE (because
+ the documentation states the argument is not optional). This
+ change makes URIENCODE and QUOTE behave similarly, since the
+ documentation states that the argument is not optional, for both.
+ (closes issue #15439) Reported by: pkempgen Patches:
+ 20090706__issue15439.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+2009-07-21 22:23 +0000 [r207930] Russell Bryant <russell@digium.com>
+
+ * doc/CODING-GUIDELINES, /: Merged revisions 207925 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r207925 | russell | 2009-07-21 17:22:18 -0500 (Tue, 21 Jul 2009)
+ | 4 lines Note that we use tabs instead of spaces for
+ indentation. I'm surprised this was never actually in here...
+ ........
+
+2009-07-21 20:30 +0000 [r207785-207862] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 207854 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500
+ (Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009)
+ | 9 lines Wait for wink before dialing when using E&M wink
+ signaling There was already code for other signaling types in
+ dahdi_handle_event to handle dialing if a dial operation dial
+ string was present. Simply add SIG_EMWINK to the list. (closes
+ issue #14434) Reported by: araasch ........ ................
+
+ * channels/chan_dahdi.c: Revert r207638, this approach could
+ potentially block for an unacceptable amount of time.
+
+2009-07-21 14:32 +0000 [r207727] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, /: Merged revisions 207723 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul
+ 2009) | 11 lines Merged revisions 207714 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul
+ 2009) | 5 lines Document default timeout for AMI originations.
+ AST-224 ........ ................
+
+2009-07-21 13:56 +0000 [r207685] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/Makefile, doc/video_console.txt, Makefile, agi/Makefile,
+ codecs/Makefile, utils/Makefile, funcs/Makefile,
+ codecs/lpc10/Makefile, main/db1-ast/Makefile, /, main/Makefile,
+ codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules,
+ pbx/Makefile, res/Makefile: Merged revisions 207680 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207680 | kpfleming | 2009-07-21 08:28:04 -0500
+ (Tue, 21 Jul 2009) | 18 lines Merged revisions 207647 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul
+ 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are
+ honored. This commit changes the build system so that
+ user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to
+ the compiler/linker *after* all flags provided by the build
+ system itself, so that the user can effectively override the
+ build system's flags if desired. In addition, ASTCFLAGS and
+ ASTLDFLAGS can now be provided *either* in the environment before
+ running 'make', or as variable assignments on the 'make' command
+ line. As a result, the use of COPTS and LDOPTS is no longer
+ necessary, so they are no longer documented, but are still
+ supported so as not to break existing build systems that supply
+ them when building Asterisk. ........ ................
+
+2009-07-21 04:51 +0000 [r207638] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Wait for wink before dialing when using
+ E&M wink signaling This patch adds a new dahdi_wait function to
+ specifically wait for the wink event. If the wink is not
+ eventually received the channel is hung up. (closes issue #14434)
+ Reported by: araasch Patches: emwinkmod uploaded by araasch
+ (license 693)
+
+2009-07-20 22:14 +0000 [r207523] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 207424 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul
+ 2009) | 39 lines Merged revisions 207423 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul
+ 2009) | 33 lines Answer video SDP offers properly when
+ videosupport is not enabled. Copied from Review board: In issue
+ 12434, the reporter describes a situation in which audio and
+ video is offered on the call, but because videosupport is
+ disabled in sip.conf, Asterisk gives no response at all to the
+ video offer. According to RFC 3264, all media offers should have
+ a corresponding answer. For offers we do not intend to actually
+ reply to with meaningful values, we should still reply with the
+ port for the media stream set to 0. In this patch, we take note
+ of what types of media have been offered and save the information
+ on the sip_pvt. The SDP in the response will take into account
+ whether media was offered. If we are not otherwise going to
+ answer a media offer, we will insert an appropriate m= line with
+ the port set to 0. It is important to note that this patch is
+ pretty much a bandage being applied to a broken bone. The patch
+ *only* helps for situations where video is offered but
+ videosupport is disabled and when udptl_pt is disabled but T.38
+ is offered. Asterisk is not guaranteed to respond to every media
+ offer. Notable cases are when multiple streams of the same type
+ are offered. The 2 media stream limit is still present with this
+ patch, too. In trunk and the 1.6.X branches, things will be a bit
+ different since Asterisk also supports text in SDPs as well.
+ (closes issue #12434) Reported by: mnnojd Review:
+ https://reviewboard.asterisk.org/r/311 Review:
+ https://reviewboard.asterisk.org/r/313 ........ ................
+
+2009-07-20 16:41 +0000 [r207364] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 207361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009)
+ | 16 lines Merged revisions 207360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009)
+ | 9 lines Only do the chan->fdno check in ast_read() in a
+ developer build. I changed this check to only happen in a
+ dev-mode build. I also added a comment explaining what is going
+ on. I also made it so that detection of this situation does not
+ affect ast_read() operation. (closes issue #14723) Reported by:
+ seadweller ........ ................
+
+2009-07-18 04:19 +0000 [r207327] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 207317 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r207317 | tilghman | 2009-07-17 23:16:44 -0500 (Fri, 17 Jul 2009)
+ | 3 lines Flag field in wrong position. Reported by "Hoggins!" on
+ asterisk-dev list. ........
+
+2009-07-18 03:50 +0000 [r207315] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c: Merged
+ revisions 145293,158010 from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 to make
+ merging easier. These changes are already on trunk.
+ ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500
+ (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c
+ channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk
+ to make merging easier later. ........ r145200 | rmudgett |
+ 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines *
+ Miscellaneous formatting changes to make v1.4 and trunk more
+ merge compatible in the mISDN area. channels/chan_misdn.c *
+ Eliminated redundant code in cb_events() EVENT_SETUP ........
+ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008)
+ | 9 lines improved helptext of misdn_set_opt. ........ r142181 |
+ rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
+ Cleaned up comment ........ r138738 | rmudgett | 2008-08-18
+ 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
+ channels/chan_misdn.c * Made bearer2str() use
+ allowed_bearers_array[] * Made use the causes.h defines instead
+ of hardcoded numbers. * Made use Asterisk presentation indicator
+ values if either of the mISDN presentation or screen options are
+ negative. * Updated the misdn_set_opt application option
+ descriptions. * Renamed the awkward Caller ID presentation
+ misdn_set_opt application option value not_screened to
+ restricted. Deprecated the not_screened option value.
+ channels/misdn/isdn_lib.c * Made use the causes.h defines instead
+ of hardcoded numbers. * Fixed some spelling errors and typos. *
+ Added all defined facility code strings to fac2str().
+ channels/misdn/isdn_lib.h * Added doxygen comments to struct
+ misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen
+ comments to struct misdn_stack. channels/misdn_config.c
+ configs/misdn.conf.sample * Updated the mISDN presentation and
+ screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex)
+ * Updated the misdn_set_opt application option descriptions. *
+ Fixed some spelling errors and typos. ................ r158010 |
+ rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines
+ Merged revision 157977 from
+ https://origsvn.digium.com/svn/asterisk/team/group/issue8824
+ ........ Fixes JIRA ABE-1726 The dial extension could be empty if
+ you are using MISDN_KEYPAD to control ISDN provider features.
+ ................
+
+2009-07-17 22:31 +0000 [r207226-207257] Tilghman Lesher <tlesher@digium.com>
+
+ * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 207255 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r207255 | tilghman | 2009-07-17 17:29:50 -0500 (Fri, 17
+ Jul 2009) | 2 lines Add flag here, too (as requested by jsmith)
+ ........
+
+ * /, doc/tex/odbcstorage.tex, UPGRADE.txt: Merged revisions 207224
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r207224 | tilghman | 2009-07-17 17:04:43 -0500 (Fri, 17
+ Jul 2009) | 2 lines Document the "flag" field in the
+ voicemessages table. ........
+
+2009-07-17 19:40 +0000 [r207104-207159] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 207156 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207156 | jpeeler | 2009-07-17 14:37:38 -0500
+ (Fri, 17 Jul 2009) | 14 lines Merged revisions 207155 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009)
+ | 7 lines Fix format specifier to print out an unsigned long
+ long. Yep, it's even ifdefed out code. But it made it to the RR
+ list... (closes issue #14726) Reported by: lmadsen ........
+ ................
+
+ * configs/chan_dahdi.conf.sample, /: Merged revisions 207095 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17
+ Jul 2009) | 2 lines Update some missing allowed options for
+ overlapdial ........
+
+2009-07-17 17:52 +0000 [r206869-207030] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 207029 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 |
+ dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines
+ sip option flags handled incorrectly (closes issue #15376)
+ Reported by: Takehiko Ooshima Tested by: dvossel,
+ Takehiko_Ooshima ........
+
+ * /, channels/chan_sip.c: Merged revisions 206939 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009)
+ | 20 lines Merged revisions 206938 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009)
+ | 14 lines SIP incorrect From: header information when callpres
+ is prohib Some ITSP make use of the "Anonymous" display name to
+ detect a requirement to withhold caller id across the PSTN. This
+ does not work if the display name is "Unknown". (closes issue
+ #14465) Reported by: Nick_Lewis Patches:
+ chan_sip.c-callerpres.patch uploaded by Nick (license 657)
+ chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license
+ 671) Tested by: Nick_Lewis, dvossel ........ ................
+
+ * /, funcs/func_timeout.c: Merged revisions 206877 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r206877 | dvossel | 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009)
+ | 6 lines TIMEOUT(absolute) returned negative value. (closes
+ issue #15513) Reported by: ys ........
+
+ * configs/iax.conf.sample, /: Merged revisions 206873 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206873 | dvossel | 2009-07-16 16:33:51 -0500
+ (Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009)
+ | 6 lines error in iax.conf related IP-based access control
+ (closes issue #15518) Reported by: pkempgen ........
+ ................
+
+ * /, main/callerid.c: Merged revisions 206868 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009)
+ | 14 lines Merged revisions 206867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009)
+ | 8 lines avoid segfault caused by user error If the CALLERPRES()
+ dialplan function is set to nothing, a segfault occurs. This is
+ user error to begin with, but I'd rather see a cli warning
+ message than have Asterisk crash on me. ........ ................
+
+2009-07-16 16:53 +0000 [r206811] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_realtime.c, /: Merged revisions 206808 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206808 | tilghman | 2009-07-16 11:51:05 -0500
+ (Thu, 16 Jul 2009) | 13 lines Merged revisions 206807 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009)
+ | 6 lines Fix a memory leak. (closes issue #15517) Reported by:
+ adomjan Patches: func_realtime.c-ast_variable_destroy.diff
+ uploaded by adomjan (license 487) ........ ................
+
+2009-07-15 22:04 +0000 [r206770] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 206768 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 |
+ dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines
+ Session timer were not activated if Supported header field in
+ INVITE had both "timer" and other options. (closes issue #15403)
+ Reported by: makoto Patches: sip-session-timer.patch uploaded by
+ makoto (license ........
+
+2009-07-15 21:50 +0000 [r206765] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /:
+ Merged revisions 206707 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009)
+ | 33 lines Merged revisions 206706 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500
+ (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
+ .......... Fixed chan_misdn crash because mISDNuser library is
+ not thread safe. With Asterisk the mISDNuser library is driven by
+ two threads concurrently: 1.
+ channels/misdn/isdn_lib.c::manager_event_handler() 2.
+ channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls
+ into the library are done concurrently and recursively from
+ isdn_lib.c. Both threads can fiddle with the master/child
+ layer3_proc_t lists. One thread may traverse the list when the
+ other interrupts it and then removes the list element which the
+ first thread was currently handling. This is exactly what caused
+ the crash. About 60 calls were needed to a Gigaset CX475 before
+ it occurred once. This patch adds locking when calling into the
+ mISDNuser library. This also fixes some cb_log calls with wrong
+ port parameter. JIRA ABE-1913 Patches: misdn-locking.patch
+ (Modified with mostly cosmetic changes) ..........
+ ................ ................
+
+2009-07-15 20:20 +0000 [r206703] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 206702 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 |
+ dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines
+ callerid(num) is wrong when username is missing A domain only sip
+ uri <sip:123.123.123.123> would return 123.123.123.123 as callid
+ num. Now, if the username is missing from a uri, the callerid num
+ field is left empty. (closes issue #15476) Reported by: viraptor
+ ........
+
+2009-07-15 16:04 +0000 [r206639] Sean Bright <sean@malleable.com>
+
+ * codecs/codec_dahdi.c, /: Merged revisions 206636 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206636 | seanbright | 2009-07-15 12:00:24 -0400
+ (Wed, 15 Jul 2009) | 9 lines Merged revisions 206635 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed,
+ 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we
+ are asking for it. ........ ................
+
+2009-07-14 20:26 +0000 [r206598] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c, contrib/scripts/meetme.sql: Merged
+ revisions 206567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206567 |
+ tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14 Jul 2009) | 6 lines
+ Document all meetme realtime fields, and in the process, make
+ some field lengths more consistent. (closes issue #15493)
+ Reported by: lasko Patches: meetme.diff uploaded by lasko
+ (license 833) ........
+
+2009-07-14 19:49 +0000 [r206565] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, /: Merged revisions 206489 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206489 | rmudgett | 2009-07-14 12:01:48 -0500
+ (Tue, 14 Jul 2009) | 35 lines Merged revisions 206487 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009)
+ | 28 lines Fixes several call transfer issues with chan_misdn. *
+ issue #14355 - Crash if attempt to transfer a call to an
+ application. Masquerade the other pair of the four asterisk
+ channels involved in the two calls. The held call already must be
+ a bridged call (not an applicaton) or it would have been
+ rejected. * issue #14692 - Held calls are not automatically
+ cleared after transfer. Allow the core to initate disconnect of
+ held calls to the ISDN port. This also fixes a similar case where
+ the party on hold hangs up before being transferred or taken off
+ hold. * JIRA ABE-1903 - Orphaned held calls left in
+ music-on-hold. Do not simply block passing the hangup event on
+ held calls to asterisk core. * Fixed to allow held calls to be
+ transferred to ringing calls. Previously, held calls could only
+ be transferred to connected calls. * Eliminated unused call
+ states to simplify hangup code. * Eliminated most uses of
+ "holded" because it is not a word. (closes issue #14355) (closes
+ issue #14692) Reported by: sodom Patches:
+ misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
+ Tested by: rmudgett ........ ................
+
+2009-07-14 14:59 +0000 [r206389] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 206386 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206386 | russell | 2009-07-14 09:51:44 -0500
+ (Tue, 14 Jul 2009) | 20 lines Merged revisions 206385 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r206385 | russell | 2009-07-14 09:48:00 -0500
+ (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009)
+ | 6 lines Ensure apathetic replies are sent out on the proper
+ socket. chan_iax2 supports multiple address bindings. The
+ send_apathetic_reply() function did not attempt to send its
+ response on the same socket that the incoming message came in on.
+ ........ ................ ................
+
+2009-07-14 01:59 +0000 [r206373] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
+ revisions 206341 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009)
+ | 11 lines Merged revisions 206284 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009)
+ | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911
+ ........ ................
+
+2009-07-13 23:27 +0000 [r206281] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 206280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206280 |
+ dvossel | 2009-07-13 18:26:51 -0500 (Mon, 13 Jul 2009) | 9 lines
+ dns lookup of peername rather than peer's host in
+ transmit_register() (closes issue #15052) Reported by: fsantulli
+ Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by
+ fsantulli (license 818) Tested by: fsantulli ........
+
+2009-07-13 16:24 +0000 [r206187] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 206185 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r206185 | tilghman | 2009-07-13 11:23:07 -0500 (Mon, 13 Jul 2009)
+ | 2 lines Remove reference to non-existent help file ........
+
+2009-07-10 21:46 +0000 [r205986] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 |
+ dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines
+ SIP register not using peer's outbound proxy If callbackextension
+ is defined for a peer it successfully causes a registration to
+ occur, but the registration ignores the outboundproxy settings
+ for the peer. This patch allows the peer to be passed to
+ obproxy_get() in transmit_register(). (closes issue #14344)
+ Reported by: Nick_Lewis Patches:
+ callbackextension_peer_trunk.diff uploaded by dvossel (license
+ 671) Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/294/ ........
+
+2009-07-10 18:45 +0000 [r205942] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /: Merged revisions 205939 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 |
+ kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line
+ Update comments about the level of T.38 support in Asterisk.
+ ........
+
+2009-07-10 17:54 +0000 [r205882] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205878 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul
+ 2009) | 30 lines Merged revisions 205877 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500
+ (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500
+ (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
+ 2009) | 10 lines Ensure that outbound NOTIFY requests are
+ properly routed through stateful proxies. With this change, we
+ make note of Record-Route headers present in any SUBSCRIBE
+ request that we receive so that our outbound NOTIFY requests will
+ have the proper Route headers in them. (closes issue #14725)
+ Reported by: ibc ........ ................ ................
+ ................
+
+2009-07-10 16:47 +0000 [r205841] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009)
+ | 37 lines Merged revisions 205804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009)
+ | 31 lines SIP registration auth loop caused by stale nonce If an
+ endpoint sends two registration requests in a very short period
+ of time with the same nonce, both receive 401 responses from
+ Asterisk, each with a different nonce (the second 401 containing
+ the current nonce and the first one being stale). If the endpoint
+ responds to the first 401, it does not match the current nonce so
+ Asterisk sends a third 401 with a newly generated nonce (which
+ updates the current nonce)... Now if the endpoint responds to the
+ second 401, it does not match the current nonce either and
+ Asterisk sends a fourth 401 with a newly generated nonce... This
+ loop goes on and on. There appears to be a simple fix for this.
+ If the nonce from the request does not match our nonce, but is a
+ good response to a previous nonce, instead of sending a 401 with
+ a newly generated nonce, use the current one instead. This breaks
+ the loop as the nonce is not updated until a response is
+ received. Additional logic has been added to make sure no nonce
+ can be responded to twice though. (closes issue #15102) Reported
+ by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license
+ 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by:
+ Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........
+ ................
+
+2009-07-10 16:01 +0000 [r205781] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 205780 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205780 |
+ kpfleming | 2009-07-10 11:00:44 -0500 (Fri, 10 Jul 2009) | 11
+ lines Eliminate extraneous LOG_DEBUG messages generated by
+ app_fax. The transmit_audio() and transmit_t38() functions in
+ app_fax have processing loops that are supposed to wait for
+ frames to arrive on the channel and then handle them, but they
+ also have short timeouts so that the loops can have watchdog
+ timers and do other required processing. This commit changes the
+ loops to not actually call ast_read() and attempt to process the
+ returned frame unless a frame actually arrived, eliminating
+ hundreds of LOG_DEBUG messages and slightly improving
+ performance. ........
+
+2009-07-10 15:58 +0000 [r205779] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul
+ 2009) | 16 lines Merged revisions 205775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
+ 2009) | 10 lines Ensure that outbound NOTIFY requests are
+ properly routed through stateful proxies. With this change, we
+ make note of Record-Route headers present in any SUBSCRIBE
+ request that we receive so that our outbound NOTIFY requests will
+ have the proper Route headers in them. (closes issue #14725)
+ Reported by: ibc ........ ................
+
+2009-07-10 15:36 +0000 [r205773] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 205770 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205770 |
+ kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12
+ lines Fix some remaining T.38 negotiation problems in app_fax.
+ Revision 205696 did not quite fix all the issues with the T.38
+ negotiation changes and app_fax; this patch corrects them, along
+ with a couple of other minor issues. (closes issue #15480)
+ Reported by: dimas Patches: test2-15480.patch uploaded by dimas
+ (license 88) ........
+
+2009-07-09 23:56 +0000 [r205731] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Merged revisions 205728 via svn merge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009)
+ | 21 lines No audio on calls from Asterisk to various ISDN
+ devices until DTMF sent by caller. Add missing clearing of the
+ dialing flag when the ISDN call is CONNECTED. (i.e. When libpri
+ generates the event PRI_EVENT_ANSWER.) (closes issue #15420)
+ Reported by: scottbmilne Patches: bug15420-1.4.25.1-diff2.txt
+ uploaded by alecdavis (license 585) Tested by: scottbmilne,
+ alecdavis (closes issue #15416) Reported by: avinoash (closes
+ issue #15389) Reported by: alecdavis This patch should also fix
+ the following issue: (issue #15205) Reported by: vinsik ........
+
+2009-07-09 21:27 +0000 [r205699] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/frame.h, /, channels/chan_sip.c, apps/app_fax.c:
+ Merged revisions 205696 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 |
+ kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16
+ lines Repair ability of SendFAX/ReceiveFAX to respond to T.38
+ switchover. Recent changes in T.38 negotiation in Asterisk caused
+ these applications to not respond when the other endpoint
+ initiated a switchover to T.38; this resulted in the T.38
+ switchover failing, and the FAX attempt to be made using an audio
+ connection, instead of T.38 (which would usually cause the FAX to
+ fail completely). This patch corrects this problem, and the
+ applications will now correctly respond to the T.38 switchover
+ request. In addition, the response will include the appopriate
+ T.38 session parameters based on what the other end offered and
+ what our end is capable of. (closes issue #14849) Reported by:
+ afosorio ........
+
+2009-07-09 16:19 +0000 [r205595-205603] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/time.h, /: Merged revisions 205600 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500
+ (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09
+ Jul 2009) | 2 lines Changing ast_samp2tv to not use floating
+ point. ........ ................
+
+ * channels/chan_iax2.c, include/asterisk/frame.h, main/rtp.c, /:
+ Merged revisions 205479 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009)
+ | 16 lines Merged revisions 205471 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009)
+ | 10 lines Fixes 8khz assumptions Many calculations assume 8khz
+ is the codec rate. This is not always the case. This patch only
+ addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there
+ are other areas that make this assumption as well. Review:
+ https://reviewboard.asterisk.org/r/306/ ........ ................
+
+2009-07-09 08:34 +0000 [r205535] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, main/ssl.c: Merged revisions 205532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205532 |
+ mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines
+ pthread_self returns a pthread_t which is not an unsigned int on
+ all pthread implementations. Casting it to an unsigned int fixes
+ compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit
+ ........
+
+2009-07-08 22:15 +0000 [r205411-205413] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/pbx.h, include/asterisk/devicestate.h,
+ main/pbx.c, /, main/devicestate.c: Merged revisions 205412 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205412 | dvossel | 2009-07-08 17:15:06 -0500
+ (Wed, 08 Jul 2009) | 12 lines Merged revisions 205409 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009)
+ | 6 lines moving ast_devstate_to_extenstate to pbx.c from
+ devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This
+ change fixes a compile time error with chan_vpb as well. ........
+ ................
+
+ * /, main/devicestate.c: Merged revisions 205410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205410 |
+ dvossel | 2009-07-08 17:02:54 -0500 (Wed, 08 Jul 2009) | 3 lines
+ missing comma in devstatestring array ........
+
+2009-07-08 19:28 +0000 [r205353] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 205350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul
+ 2009) | 20 lines Merged revisions 205349 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul
+ 2009) | 14 lines Prevent phantom calls to queue members. If a
+ caller were to hang up while a periodic announcement or position
+ were being said, the return value for those functions would
+ incorrectly indicate that the caller was still in the queue. With
+ these changes, the problem does not occur. (closes issue #14631)
+ Reported by: latinsud Patches: queue_announce_ghost_call2.diff
+ uploaded by latinsud (license 745) (with small modification from
+ me) ........ ................
+
+2009-07-08 18:22 +0000 [r205302] Jason Parker <jparker@digium.com>
+
+ * config.guess, config.sub, /: Merged revisions 205291 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205291 | qwell | 2009-07-08 13:19:46 -0500
+ (Wed, 08 Jul 2009) | 9 lines Merged revisions 205288 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul
+ 2009) | 1 line Update config.guess and config.sub from the
+ savannah.gnu.org git repo. ........ ................
+
+2009-07-08 18:18 +0000 [r205287] David Brooks <dbrooks@digium.com>
+
+ * /, main/features.c: Merged revisions 205254 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205254 |
+ dbrooks | 2009-07-08 12:26:26 -0500 (Wed, 08 Jul 2009) | 8 lines
+ Fixes Park() argument handling Park() was not respecting the
+ arguments passed to it. Any extension/context/priority given to
+ it was being ignored. This patch remedies this. (closes issue
+ #15380) Reported by: DLNoah ........
+
+2009-07-08 17:00 +0000 [r205223] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c: oops, fixing build
+
+2009-07-08 16:55 +0000 [r205217] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/time.h, /: Merged revisions 205216 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500
+ (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009)
+ | 10 lines ast_samp2tv needs floating point for 16khz audio In
+ ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The
+ .5 is currently stripped off because we don't calculate using
+ floating points. This causes madness with 16khz audio. (issue
+ ABE-1899) Review: https://reviewboard.asterisk.org/r/305/
+ ........ ................
+
+2009-07-08 16:30 +0000 [r205207] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c: Merged revisions 205196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009)
+ | 9 lines Merged revisions 205188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009)
+ | 2 lines Add redirection warnings for the invalid language codes
+ previously removed. ........ ................
+
+2009-07-08 15:57 +0000 [r205148-205154] Russell Bryant <russell@digium.com>
+
+ * /, main/ssl.c: Merged revisions 205151 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205151 |
+ russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines
+ Use tabs instead of spaces for indentation. ........
+
+ * include/asterisk/_private.h, res/res_jabber.c, main/asterisk.c,
+ /, main/Makefile, res/res_crypto.c, main/ssl.c (added): Merged
+ revisions 205120 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 |
+ russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines
+ Move OpenSSL initialization to a single place, make library usage
+ thread-safe. While doing some reading about OpenSSL, I noticed a
+ couple of things that needed to be improved with our usage of
+ OpenSSL. 1) We had initialization of the library done in multiple
+ modules. This has now been moved to a core function that gets
+ executed during Asterisk startup. We already link OpenSSL into
+ the core for TCP/TLS functionality, so this was the most logical
+ place to do it. 2) OpenSSL is not thread-safe by default.
+ However, making it thread safe is very easy. We just have to
+ provide a couple of callbacks. One callback returns a thread ID.
+ The other handles locking. For more information, start with the
+ "Is OpenSSL thread-safe?" question on the FAQ page of
+ openssl.org. ........
+
+2009-07-06 13:41 +0000 [r204951] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c, /: Merged revisions 204948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r204948 |
+ kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7
+ lines Improve handling of AST_CONTROL_T38 and
+ AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This
+ change allows applications that request T.38 negotiation on a
+ channel that does not support it to get the proper indication
+ that it is not supported, rather than thinking that negotiation
+ was started when it was not. ........
+
+2009-07-02 22:06 +0000 [r204838] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 204835 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r204835 | rmudgett | 2009-07-02 17:01:28 -0500
+ (Thu, 02 Jul 2009) | 17 lines Merged revisions 204834 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009)
+ | 10 lines Removed confusing warning message "Got Busy in
+ Connected State" If an incoming mISDN call is answered with the
+ Answer application and a subsequent Dial gets a busy endpoint
+ then it is valid for that already connected channel to get the
+ busy indication. Asterisk will play the busy tones until the
+ dialplan plays something else or hangs up the call. (closes issue
+ #11974) Reported by: fvdb ........ ................
+
+2009-07-02 16:12 +0000 [r204711] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, /,
+ main/devicestate.c: Merged revisions 204710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009)
+ | 21 lines Merged revisions 204681 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009)
+ | 14 lines Improved mapping of extension states from combined
+ device states. This fixes a few issues with incorrect extension
+ states and adds a cli command, core show device2extenstate, to
+ display all possible state mappings. (closes issue #15413)
+ Reported by: legart Patches: exten_helper.diff uploaded by
+ dvossel (license 671) Tested by: dvossel, legart, amilcar Review:
+ https://reviewboard.asterisk.org/r/301/ ........ ................
+
+2009-06-30 21:30 +0000 [r204611] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c, UPGRADE.txt: Merged revisions 204563 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r204563 | tilghman | 2009-06-30 15:41:04 -0500
+ (Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009)
+ | 6 lines More incorrect language codes, plus ensuring that
+ regionalizations use the specified language, and not English for
+ grammar. (closes issue #15022) Reported by: greenfieldtech
+ Patches: 20090519__issue15022.diff.txt uploaded by tilghman
+ (license 14) ........ ................
+
+2009-06-30 18:55 +0000 [r204478] Jason Parker <jparker@digium.com>
+
+ * /, main/say.c: Merged revisions 204475 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) |
+ 9 lines Merged revisions 204474 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) |
+ 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a
+ comment typo in passing. ........ ................
+
+2009-06-30 18:44 +0000 [r204473] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /, main/say.c, UPGRADE.txt: Recorded merge
+ of revisions 204470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009)
+ | 18 lines Recorded merge of revisions 204469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009)
+ | 11 lines "tw" is the language specification for Twi (from
+ Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier
+ Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman
+ (license 14) 20090617__issue15346__trunk.diff.txt uploaded by
+ tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt
+ uploaded by tilghman (license 14)
+ 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman
+ (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by
+ tilghman (license 14) Tested by: volivier ........
+ ................
+
+2009-06-30 17:22 +0000 [r204442] Russell Bryant <russell@digium.com>
+
+ * configs/res_config_sqlite.conf (removed),
+ configs/res_config_sqlite.conf.sample (added), /: Merged
+ revisions 204440 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r204440 |
+ russell | 2009-06-30 12:22:16 -0500 (Tue, 30 Jun 2009) | 2 lines
+ Rename res_config_sqlite.conf to res_config_sqlite.conf.sample
+ (missing .sample). ........
+
+2009-06-29 22:53 +0000 [r204250-204304] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 204301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun
+ 2009) | 15 lines Merged revisions 204300 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun
+ 2009) | 9 lines Add error message so that it is clear why a SIP
+ peer was not processed when a DNS lookup fails on a host or
+ outboundproxy. (closes issue #13432) Reported by: p_lindheimer
+ Patches: outboundproxy.patch uploaded by p (license 558) ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 204247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun
+ 2009) | 32 lines Merged revisions 204243,204246 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun
+ 2009) | 22 lines Fix a problem where chan_sip would ignore "old"
+ but valid responses. chan_sip has had a problem for quite a long
+ time that would manifest when Asterisk would send multiple SIP
+ responses on the same dialog before receiving a response. The
+ problem occurred because chan_sip only kept track of the highest
+ outgoing sequence number used on the dialog. If Asterisk sent two
+ requests out, and a response arrived for the first request sent,
+ then Asterisk would ignore the response. The result was that
+ Asterisk would continue retransmitting the requests and ignoring
+ the responses until the maximum number of retransmissions had
+ been reached. The fix here is to rearrange the code a bit so that
+ instead of simply comparing the sequence number of the response
+ to our latest outgoing sequence number, we walk our list of
+ outstanding packets and determine if there is a match. If there
+ is, we continue. If not, then we ignore the response. In doing
+ this, I found a few completely useless variables that I have now
+ removed. (closes issue #11231) Reported by: flefoll Review:
+ https://reviewboard.asterisk.org/r/298 ........ r204246 |
+ mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3
+ lines Fix build oops. ........ ................
+
+2009-06-27 09:55 +0000 [r203961] Russell Bryant <russell@digium.com>
+
+ * CHANGES, /: Merged revisions 203960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r203960 |
+ russell | 2009-06-27 04:51:45 -0500 (Sat, 27 Jun 2009) | 2 lines
+ Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt.
+ ........
+
+2009-06-27 01:24 +0000 [r203941] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203909 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r203909 | rmudgett | 2009-06-26 20:07:52 -0500
+ (Fri, 26 Jun 2009) | 23 lines Merged revisions 203908 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009)
+ | 16 lines The ISDN CPE side should not exclusively pick B
+ channels normally. Before this patch, Asterisk unconditionally
+ picked B channels exclusively on the CPE side and normally
+ allowed alternative B channels on the network side. Now Asterisk
+ does the opposite. Reasons for the CPE side to normally not pick
+ B channels exclusively: * For CPE point-to-multipoint mode (i.e.
+ phone side), the CPE side does not have enough information to
+ exclusively pick B channels. (There may be other devices on the
+ line.) * Q.931 gives preference to the network side picking B
+ channels. * Some telcos require the CPE side to not pick B
+ channels exclusively. (closes issue #14383) Reported by:
+ mbrancaleoni ........ ................
+
+2009-06-26 22:14 +0000 [r203857] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203853 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r203853 | jpeeler | 2009-06-26 17:11:31 -0500
+ (Fri, 26 Jun 2009) | 12 lines Merged revisions 203848 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009)
+ | 5 lines Make sure to recreate the dahdi pseudo channel after
+ dahdi restart (closes issue #14477) Reported by: timking ........
+ ................
+
+2009-06-26 21:27 +0000 [r203782-203828] Russell Bryant <russell@digium.com>
+
+ * /, main/file.c: Merged revisions 203802 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009)
+ | 22 lines Merged revisions 203785 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009)
+ | 15 lines Don't fast forward past the end of a message. This is
+ nice change for users of the voicemail application. If someone
+ gets a little carried away with fast forwarding through a
+ message, they can easily get to the end and accidentally exit the
+ voicemail application by hitting the fast forward key during the
+ following prompt. This adds some safety by not allowing a fast
+ forward past the end of a message. (closes issue #14554) Reported
+ by: lacoursj Patches: 21761.patch uploaded by lacoursj (license
+ 707) Tested by: lacoursj ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 203779 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 |
+ russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines
+ Ensure the TCP read buffer is fully initialized before handling
+ each packet. (closes issue #14452) Reported by: umberto71
+ ........
+
+2009-06-26 20:18 +0000 [r203731] David Brooks <dbrooks@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 203721 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009)
+ | 16 lines Fixing voicemail's error in checking max silence vs
+ min message length Max silence was represented in milliseconds,
+ yet vmminsecs (minmessage) was represented as seconds. Also, the
+ inequality was reversed. The warning, if triggered, was "Max
+ silence should be less than minmessage or you may get empty
+ messages", which should have been logged if max silence was
+ greater than minmessage, but the check was for less than. Also,
+ conforming if statement to coding guidelines. closes issue
+ #15331) Reported by: markd Review:
+ https://reviewboard.asterisk.org/r/293/ ........
+
+2009-06-26 19:49 +0000 [r203715] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, /,
+ main/devicestate.c: Merged revisions 203702 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r203702 |
+ russell | 2009-06-26 14:31:14 -0500 (Fri, 26 Jun 2009) | 5 lines
+ Make invalid hints report Unavailable instead of Idle. (closes
+ issue #14413) Reported by: pj ........
+
+2009-06-26 19:48 +0000 [r203712] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 203710 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009)
+ | 7 lines moving debug message from level 0 to 1. (closes issue
+ #15404) Reported by: leobrown Patches: iax_codec_debug.patch
+ uploaded by leobrown (license 541) ........
+
+2009-06-26 19:42 +0000 [r203709] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009)
+ | 16 lines Check if polarityonanswerdelay has elapsed before
+ setting a channel as answered after a polarity reversal.
+ Previously on a polarity switch event chan_dahdi would set the
+ channel immediately as answered. This would cause problems if a
+ polarity reversal occurred when the line was picked up as the
+ dial would not have yet occurred. Now if the polarity reversal
+ occurs before delay has elapsed after coming off hook or an
+ answer, it is ignored. Also, some refactoring was done in
+ _handle_event. (closes issue #13917) Reported by: alecdavis
+ Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by
+ alecdavis (license 585) Tested by: alecdavis ........
+
+2009-06-26 19:38 +0000 [r203705] Joshua Colp <jcolp@digium.com>
+
+ * configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c,
+ main/channel.c, main/frame.c, /, channels/chan_sip.c,
+ apps/app_fax.c: Merged revisions 203699 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 |
+ file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines
+ Improve T.38 negotiation by exchanging session parameters between
+ application and channel. ........
+
+2009-06-25 21:46 +0000 [r203445] David Vossel <dvossel@digium.com>
+
+ * main/ast_expr2.fl, main/ast_expr2.c, /: Merged revisions 203444
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25
+ Jun 2009) | 4 lines fixes a few redundant conditions (issue
+ #15269) ........
+
+2009-06-25 21:21 +0000 [r203400] Terry Wilson <twilson@digium.com>
+
+ * main/cli.c, /: Merged revisions 203381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009)
+ | 11 lines Merged revisions 203380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009)
+ | 4 lines I didn't see that Mark already fixed the underlying
+ issue! Yay for removing useless code. ........ ................
+
+2009-06-25 21:08 +0000 [r203379] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 203376 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009)
+ | 16 lines Merged revisions 203375 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009)
+ | 9 lines Fix a case where CDR answer time could be before the
+ start time involving parking. (closes issue #13794) Reported by:
+ davidw Patches: 13794.patch uploaded by murf (license 17)
+ 13794.patch.160 uploaded by murf (license 17) Tested by: murf,
+ dbrooks ........ ................
+
+2009-06-25 19:27 +0000 [r203276] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203258 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) |
+ 10 lines Unmute when we get a dtmfup (we muted on dtmfdown)
+ event. This would occasionally cause one-way audio when using
+ hardware DTMF detection. (closes issue #14761) Reported by:
+ tzafrir Patches: v1-14761.patch uploaded by dimas (license 88)
+ Tested by: tzafrir, dimas ........
+
+2009-06-25 16:08 +0000 [r203119] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 203116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009)
+ | 18 lines Merged revisions 203115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009)
+ | 11 lines Resolve a crash related to a T.38 reinvite race
+ condition. This change resolves a crash observed locally during
+ some T.38 testing. A call was set up using a call file, and when
+ the T.38 reinvite came in, the channel state was still
+ AST_STATE_DOWN. The reason is explained by a comment in the code
+ that previously lived in the handling of AST_STATE_RINGING. This
+ change modifies the logic to handle the same race condition for
+ any channel state that is not UP. (closes ABE-1895) ........
+ ................
+
+2009-06-24 21:27 +0000 [r203077] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203037 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r203037 | rmudgett | 2009-06-24 16:08:55 -0500
+ (Wed, 24 Jun 2009) | 15 lines Merged revisions 203036 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009)
+ | 8 lines Improved chan_dahdi.conf pritimer error checking. Valid
+ format is: pritimer=timer_name,timer_value * Fixed segfault if
+ the ',' is missing. * Completely check the range returned by
+ pri_timer2idx() to prevent possible access outside array bounds.
+ ........ ................
+
+2009-06-24 18:30 +0000 [r202970] Mark Miche